From 5163c1eede8e9073e5b6bf1a988ed07d35820343 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 9 Feb 2015 13:26:01 +0200 Subject: [PATCH 001/162] ASoC: omap: Kconfig: Support for omap5-uevm analog audio The analog audio is supported via omap-abe-twl6040 machine driver on omap5-uevm. Update the Kconfig file to reflect this and select the Palmas clock driver which is providing the 32K clock for audio on omap5-uevm. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index a2cd3486ac55..e7c78b0406b5 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -100,17 +100,19 @@ config SND_OMAP_SOC_OMAP_TWL4030 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST) + depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST) select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC + select COMMON_CLK_PALMAS if SOC_OMAP5 help Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports: - SDP4430/Blaze boards - PandaBoard (4430) - PandaBoardES (4460) + - omap5-uevm (5432) config SND_OMAP_SOC_OMAP3_PANDORA tristate "SoC Audio support for OMAP3 Pandora" From b6d6c6e95ff0e78f9b8393e6b9f25d5a4341ae1a Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 9 Feb 2015 16:08:25 +0100 Subject: [PATCH 002/162] ASoC: atmel_ssc_dai: Allow more rates When the SSC acts as BCK master, use a ratnum rule to limit the rate instead of only doing the standard rates. When the SSC acts as BCK slave, allow any BCK frequency up to the SSC master clock, divided by either of 2, 3 or 6. Put a cap at 384kHz. Who's /ever/ going to need more than that? The divider of 2, 3 or 6 is selected based on the Serial Clock Ratio Considerations section from the SSC documentation: The Transmitter and the Receiver can be programmed to operate with the clock signals provided on either the TK or RK pins. This allows the SSC to support many slave-mode data transfers. In this case, the maximum clock speed allowed on the RK pin is: - Peripheral clock divided by 2 if Receiver Frame Synchro is input - Peripheral clock divided by 3 if Receiver Frame Synchro is output In addition, the maximum clock speed allowed on the TK pin is: - Peripheral clock divided by 6 if Transmit Frame Synchro is input - Peripheral clock divided by 2 if Transmit Frame Synchro is output Signed-off-by: Peter Rosin Acked-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 111 ++++++++++++++++++++++++++++++-- sound/soc/atmel/atmel_ssc_dai.h | 1 + 2 files changed, 108 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 379ac2a6ab16..6b8e64899205 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,94 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +/* + * When the bit clock is input, limit the maximum rate according to the + * Serial Clock Ratio Considerations section from the SSC documentation: + * + * The Transmitter and the Receiver can be programmed to operate + * with the clock signals provided on either the TK or RK pins. + * This allows the SSC to support many slave-mode data transfers. + * In this case, the maximum clock speed allowed on the RK pin is: + * - Peripheral clock divided by 2 if Receiver Frame Synchro is input + * - Peripheral clock divided by 3 if Receiver Frame Synchro is output + * In addition, the maximum clock speed allowed on the TK pin is: + * - Peripheral clock divided by 6 if Transmit Frame Synchro is input + * - Peripheral clock divided by 2 if Transmit Frame Synchro is output + * + * When the bit clock is output, limit the rate according to the + * SSC divider restrictions. + */ +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro (i.e. capture) + * is output (format is _CFS) and the RK pin + * is used for input (format is _CBM_). + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro (i.e. playback) + * is input (format is _CFM) and the TK pin + * is used for input (format _CBM_ but not + * using the RK pin). + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,6 +288,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); @@ -207,6 +296,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); /* Reset the SSC to keep it at a clean status */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); @@ -219,6 +309,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -783,8 +884,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -804,12 +903,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d511495..80b153857a88 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); From b6a42670e074da39b5a9f990774359e0733ca9cd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2015 22:37:16 +0100 Subject: [PATCH 003/162] ALSA: seq: Move EXPORT_SYMBOL() after each function ... to follow the standard coding style. Signed-off-by: Takashi Iwai --- sound/core/seq/seq_device.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 0631bdadd12b..a752a79a8d3a 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -133,11 +133,13 @@ void snd_seq_autoload_lock(void) { atomic_inc(&snd_seq_in_init); } +EXPORT_SYMBOL(snd_seq_autoload_lock); void snd_seq_autoload_unlock(void) { atomic_dec(&snd_seq_in_init); } +EXPORT_SYMBOL(snd_seq_autoload_unlock); static void autoload_drivers(void) { @@ -195,10 +197,12 @@ void snd_seq_autoload_init(void) queue_autoload_drivers(); #endif } +EXPORT_SYMBOL(snd_seq_autoload_init); #else #define try_autoload(ops) /* NOP */ #endif + void snd_seq_device_load_drivers(void) { #ifdef CONFIG_MODULES @@ -206,6 +210,7 @@ void snd_seq_device_load_drivers(void) flush_work(&autoload_work); #endif } +EXPORT_SYMBOL(snd_seq_device_load_drivers); /* * register a sequencer device @@ -268,6 +273,7 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, return 0; } +EXPORT_SYMBOL(snd_seq_device_new); /* * free the existing device @@ -326,6 +332,7 @@ static int snd_seq_device_dev_register(struct snd_device *device) unlock_driver(ops); return 0; } +EXPORT_SYMBOL(snd_seq_device_register_driver); /* * disconnect the device @@ -344,6 +351,7 @@ static int snd_seq_device_dev_disconnect(struct snd_device *device) unlock_driver(ops); return 0; } +EXPORT_SYMBOL(snd_seq_device_unregister_driver); /* * register device driver @@ -604,13 +612,3 @@ static void __exit alsa_seq_device_exit(void) module_init(alsa_seq_device_init) module_exit(alsa_seq_device_exit) - -EXPORT_SYMBOL(snd_seq_device_load_drivers); -EXPORT_SYMBOL(snd_seq_device_new); -EXPORT_SYMBOL(snd_seq_device_register_driver); -EXPORT_SYMBOL(snd_seq_device_unregister_driver); -#ifdef CONFIG_MODULES -EXPORT_SYMBOL(snd_seq_autoload_init); -EXPORT_SYMBOL(snd_seq_autoload_lock); -EXPORT_SYMBOL(snd_seq_autoload_unlock); -#endif From 72496edcf85e048b4c5373d518e4f27938d9594e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Feb 2015 22:39:51 +0100 Subject: [PATCH 004/162] ALSA: seq: Don't compile snd_seq_device_load_drivers() for built-in Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 4 ++++ sound/core/seq/seq_device.c | 9 +++------ 2 files changed, 7 insertions(+), 6 deletions(-) diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index 2b5f24cc7548..d52433563da2 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -67,7 +67,11 @@ struct snd_seq_dev_ops { /* * prototypes */ +#ifdef CONFIG_MODULES void snd_seq_device_load_drivers(void); +#else +#define snd_seq_device_load_drivers() +#endif int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, struct snd_seq_device **result); int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, int argsize); int snd_seq_device_unregister_driver(char *id); diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index a752a79a8d3a..075a66c0cc6a 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -198,19 +198,16 @@ void snd_seq_autoload_init(void) #endif } EXPORT_SYMBOL(snd_seq_autoload_init); -#else -#define try_autoload(ops) /* NOP */ -#endif - void snd_seq_device_load_drivers(void) { -#ifdef CONFIG_MODULES queue_autoload_drivers(); flush_work(&autoload_work); -#endif } EXPORT_SYMBOL(snd_seq_device_load_drivers); +#else +#define try_autoload(ops) /* NOP */ +#endif /* * register a sequencer device From 7c37ae5c625aaa4836466cfaea829a3199dfc571 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 10:51:59 +0100 Subject: [PATCH 005/162] ALSA: seq: Rewrite sequencer device binding with standard bus We've used the old house-made code for binding the sequencer device and driver. This can be far better implemented with the standard bus nowadays. This patch refactors the whole sequencer binding code with the bus /sys/bus/snd_seq. The devices appear as id-card-device on this bus and are bound with the drivers corresponding to the given id like the former implementation. The module autoload is also kept like before. There is no change in API functions by this patch, and almost all transitions are kept inside seq_device.c. The proc file output will change slightly but kept compatible as much as possible. Further integration works will follow in later patches. Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 3 + sound/core/seq/seq_device.c | 543 +++++++++++------------------------- 2 files changed, 171 insertions(+), 375 deletions(-) diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index d52433563da2..ea256b825146 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -43,8 +43,11 @@ struct snd_seq_device { void *private_data; /* private data for the caller */ void (*private_free)(struct snd_seq_device *device); struct list_head list; /* link to next device */ + struct device dev; }; +#define to_seq_dev(_dev) \ + container_of(_dev, struct snd_seq_device, dev) /* driver operators * init_device: diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 075a66c0cc6a..d3320ffe43c2 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -36,6 +36,7 @@ * */ +#include #include #include #include @@ -51,77 +52,57 @@ MODULE_AUTHOR("Takashi Iwai "); MODULE_DESCRIPTION("ALSA sequencer device management"); MODULE_LICENSE("GPL"); -/* driver state */ -#define DRIVER_EMPTY 0 -#define DRIVER_LOADED (1<<0) -#define DRIVER_REQUESTED (1<<1) -#define DRIVER_LOCKED (1<<2) -#define DRIVER_REQUESTING (1<<3) - -struct ops_list { - char id[ID_LEN]; /* driver id */ - int driver; /* driver state */ - int used; /* reference counter */ - int argsize; /* argument size */ - - /* operators */ +struct snd_seq_driver { + struct device_driver driver; + char id[ID_LEN]; + int argsize; struct snd_seq_dev_ops ops; - - /* registered devices */ - struct list_head dev_list; /* list of devices */ - int num_devices; /* number of associated devices */ - int num_init_devices; /* number of initialized devices */ - struct mutex reg_mutex; - - struct list_head list; /* next driver */ }; +#define to_seq_drv(_drv) \ + container_of(_drv, struct snd_seq_driver, driver) -static LIST_HEAD(opslist); -static int num_ops; -static DEFINE_MUTEX(ops_mutex); +/* + * bus definition + */ +static int snd_seq_bus_match(struct device *dev, struct device_driver *drv) +{ + struct snd_seq_device *sdev = to_seq_dev(dev); + struct snd_seq_driver *sdrv = to_seq_drv(drv); + + return strcmp(sdrv->id, sdev->id) == 0 && + sdrv->argsize == sdev->argsize; +} + +static struct bus_type snd_seq_bus_type = { + .name = "snd_seq", + .match = snd_seq_bus_match, +}; + +/* + * proc interface -- just for compatibility + */ #ifdef CONFIG_PROC_FS static struct snd_info_entry *info_entry; -#endif -/* - * prototypes - */ -static int snd_seq_device_free(struct snd_seq_device *dev); -static int snd_seq_device_dev_free(struct snd_device *device); -static int snd_seq_device_dev_register(struct snd_device *device); -static int snd_seq_device_dev_disconnect(struct snd_device *device); +static int print_dev_info(struct device *dev, void *data) +{ + struct snd_seq_device *sdev = to_seq_dev(dev); + struct snd_info_buffer *buffer = data; -static int init_device(struct snd_seq_device *dev, struct ops_list *ops); -static int free_device(struct snd_seq_device *dev, struct ops_list *ops); -static struct ops_list *find_driver(char *id, int create_if_empty); -static struct ops_list *create_driver(char *id); -static void unlock_driver(struct ops_list *ops); -static void remove_drivers(void); + snd_iprintf(buffer, "snd-%s,%s,%d\n", sdev->id, + dev->driver ? "loaded" : "empty", + dev->driver ? 1 : 0); + return 0; +} -/* - * show all drivers and their status - */ - -#ifdef CONFIG_PROC_FS static void snd_seq_device_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) { - snd_iprintf(buffer, "snd-%s%s%s%s,%d\n", - ops->id, - ops->driver & DRIVER_LOADED ? ",loaded" : (ops->driver == DRIVER_EMPTY ? ",empty" : ""), - ops->driver & DRIVER_REQUESTED ? ",requested" : "", - ops->driver & DRIVER_LOCKED ? ",locked" : "", - ops->num_devices); - } - mutex_unlock(&ops_mutex); + bus_for_each_dev(&snd_seq_bus_type, NULL, buffer, print_dev_info); } #endif - + /* * load all registered drivers (called from seq_clientmgr.c) */ @@ -141,52 +122,29 @@ void snd_seq_autoload_unlock(void) } EXPORT_SYMBOL(snd_seq_autoload_unlock); -static void autoload_drivers(void) +static int request_seq_drv(struct device *dev, void *data) +{ + struct snd_seq_device *sdev = to_seq_dev(dev); + + if (!dev->driver) + request_module("snd-%s", sdev->id); + return 0; +} + +static void autoload_drivers(struct work_struct *work) { /* avoid reentrance */ - if (atomic_inc_return(&snd_seq_in_init) == 1) { - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) { - if ((ops->driver & DRIVER_REQUESTING) && - !(ops->driver & DRIVER_REQUESTED)) { - ops->used++; - mutex_unlock(&ops_mutex); - ops->driver |= DRIVER_REQUESTED; - request_module("snd-%s", ops->id); - mutex_lock(&ops_mutex); - ops->used--; - } - } - mutex_unlock(&ops_mutex); - } + if (atomic_inc_return(&snd_seq_in_init) == 1) + bus_for_each_dev(&snd_seq_bus_type, NULL, NULL, + request_seq_drv); atomic_dec(&snd_seq_in_init); } -static void call_autoload(struct work_struct *work) -{ - autoload_drivers(); -} - -static DECLARE_WORK(autoload_work, call_autoload); - -static void try_autoload(struct ops_list *ops) -{ - if (!ops->driver) { - ops->driver |= DRIVER_REQUESTING; - schedule_work(&autoload_work); - } -} +static DECLARE_WORK(autoload_work, autoload_drivers); static void queue_autoload_drivers(void) { - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) - try_autoload(ops); - mutex_unlock(&ops_mutex); + schedule_work(&autoload_work); } void snd_seq_autoload_init(void) @@ -206,9 +164,50 @@ void snd_seq_device_load_drivers(void) } EXPORT_SYMBOL(snd_seq_device_load_drivers); #else -#define try_autoload(ops) /* NOP */ +#define queue_autoload_drivers() /* NOP */ #endif +/* + * device management + */ +static int snd_seq_device_dev_free(struct snd_device *device) +{ + struct snd_seq_device *dev = device->device_data; + + put_device(&dev->dev); + return 0; +} + +static int snd_seq_device_dev_register(struct snd_device *device) +{ + struct snd_seq_device *dev = device->device_data; + int err; + + err = device_add(&dev->dev); + if (err < 0) + return err; + if (!dev->dev.driver) + queue_autoload_drivers(); + return 0; +} + +static int snd_seq_device_dev_disconnect(struct snd_device *device) +{ + struct snd_seq_device *dev = device->device_data; + + device_del(&dev->dev); + return 0; +} + +static void snd_seq_dev_release(struct device *dev) +{ + struct snd_seq_device *sdev = to_seq_dev(dev); + + if (sdev->private_free) + sdev->private_free(sdev); + kfree(sdev); +} + /* * register a sequencer device * card = card info @@ -220,7 +219,6 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, struct snd_seq_device **result) { struct snd_seq_device *dev; - struct ops_list *ops; int err; static struct snd_device_ops dops = { .dev_free = snd_seq_device_dev_free, @@ -234,16 +232,10 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, if (snd_BUG_ON(!id)) return -EINVAL; - ops = find_driver(id, 1); - if (ops == NULL) + dev = kzalloc(sizeof(*dev) + argsize, GFP_KERNEL); + if (!dev) return -ENOMEM; - dev = kzalloc(sizeof(*dev)*2 + argsize, GFP_KERNEL); - if (dev == NULL) { - unlock_driver(ops); - return -ENOMEM; - } - /* set up device info */ dev->card = card; dev->device = device; @@ -251,20 +243,19 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, dev->argsize = argsize; dev->status = SNDRV_SEQ_DEVICE_FREE; - /* add this device to the list */ - mutex_lock(&ops->reg_mutex); - list_add_tail(&dev->list, &ops->dev_list); - ops->num_devices++; - mutex_unlock(&ops->reg_mutex); + device_initialize(&dev->dev); + dev->dev.parent = &card->card_dev; + dev->dev.bus = &snd_seq_bus_type; + dev->dev.release = snd_seq_dev_release; + dev_set_name(&dev->dev, "%s-%d-%d", dev->id, card->number, device); - if ((err = snd_device_new(card, SNDRV_DEV_SEQUENCER, dev, &dops)) < 0) { - snd_seq_device_free(dev); + /* add this device to the list */ + err = snd_device_new(card, SNDRV_DEV_SEQUENCER, dev, &dops); + if (err < 0) { + put_device(&dev->dev); return err; } - try_autoload(ops); - unlock_driver(ops); - if (result) *result = dev; @@ -273,82 +264,33 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, EXPORT_SYMBOL(snd_seq_device_new); /* - * free the existing device + * driver binding - just pass to each driver callback */ -static int snd_seq_device_free(struct snd_seq_device *dev) +static int snd_seq_drv_probe(struct device *dev) { - struct ops_list *ops; - - if (snd_BUG_ON(!dev)) - return -EINVAL; - - ops = find_driver(dev->id, 0); - if (ops == NULL) - return -ENXIO; - - /* remove the device from the list */ - mutex_lock(&ops->reg_mutex); - list_del(&dev->list); - ops->num_devices--; - mutex_unlock(&ops->reg_mutex); - - free_device(dev, ops); - if (dev->private_free) - dev->private_free(dev); - kfree(dev); - - unlock_driver(ops); + struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); + struct snd_seq_device *sdev = to_seq_dev(dev); + int err; + err = sdrv->ops.init_device(sdev); + if (err < 0) + return err; + sdev->status = SNDRV_SEQ_DEVICE_REGISTERED; return 0; } -static int snd_seq_device_dev_free(struct snd_device *device) +static int snd_seq_drv_remove(struct device *dev) { - struct snd_seq_device *dev = device->device_data; - return snd_seq_device_free(dev); -} + struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); + struct snd_seq_device *sdev = to_seq_dev(dev); + int err; -/* - * register the device - */ -static int snd_seq_device_dev_register(struct snd_device *device) -{ - struct snd_seq_device *dev = device->device_data; - struct ops_list *ops; - - ops = find_driver(dev->id, 0); - if (ops == NULL) - return -ENOENT; - - /* initialize this device if the corresponding driver was - * already loaded - */ - if (ops->driver & DRIVER_LOADED) - init_device(dev, ops); - - unlock_driver(ops); + err = sdrv->ops.free_device(sdev); + if (err < 0) + return err; + sdev->status = SNDRV_SEQ_DEVICE_FREE; return 0; } -EXPORT_SYMBOL(snd_seq_device_register_driver); - -/* - * disconnect the device - */ -static int snd_seq_device_dev_disconnect(struct snd_device *device) -{ - struct snd_seq_device *dev = device->device_data; - struct ops_list *ops; - - ops = find_driver(dev->id, 0); - if (ops == NULL) - return -ENOENT; - - free_device(dev, ops); - - unlock_driver(ops); - return 0; -} -EXPORT_SYMBOL(snd_seq_device_unregister_driver); /* * register device driver @@ -358,226 +300,66 @@ EXPORT_SYMBOL(snd_seq_device_unregister_driver); int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, int argsize) { - struct ops_list *ops; - struct snd_seq_device *dev; + struct snd_seq_driver *sdrv; + int err; if (id == NULL || entry == NULL || entry->init_device == NULL || entry->free_device == NULL) return -EINVAL; - ops = find_driver(id, 1); - if (ops == NULL) + sdrv = kzalloc(sizeof(*sdrv), GFP_KERNEL); + if (!sdrv) return -ENOMEM; - if (ops->driver & DRIVER_LOADED) { - pr_warn("ALSA: seq: driver_register: driver '%s' already exists\n", id); - unlock_driver(ops); - return -EBUSY; - } - mutex_lock(&ops->reg_mutex); - /* copy driver operators */ - ops->ops = *entry; - ops->driver |= DRIVER_LOADED; - ops->argsize = argsize; + sdrv->driver.name = id; + sdrv->driver.bus = &snd_seq_bus_type; + sdrv->driver.probe = snd_seq_drv_probe; + sdrv->driver.remove = snd_seq_drv_remove; + strlcpy(sdrv->id, id, sizeof(sdrv->id)); + sdrv->argsize = argsize; + sdrv->ops = *entry; - /* initialize existing devices if necessary */ - list_for_each_entry(dev, &ops->dev_list, list) { - init_device(dev, ops); - } - mutex_unlock(&ops->reg_mutex); - - unlock_driver(ops); - - return 0; + err = driver_register(&sdrv->driver); + if (err < 0) + kfree(sdrv); + return err; } +EXPORT_SYMBOL(snd_seq_device_register_driver); - -/* - * create driver record +/* callback to find a specific driver; data is a pointer to the id string ptr. + * when the id matches, store the driver pointer in return and break the loop. */ -static struct ops_list * create_driver(char *id) +static int find_drv(struct device_driver *drv, void *data) { - struct ops_list *ops; + struct snd_seq_driver *sdrv = to_seq_drv(drv); + void **ptr = (void **)data; - ops = kzalloc(sizeof(*ops), GFP_KERNEL); - if (ops == NULL) - return ops; - - /* set up driver entry */ - strlcpy(ops->id, id, sizeof(ops->id)); - mutex_init(&ops->reg_mutex); - /* - * The ->reg_mutex locking rules are per-driver, so we create - * separate per-driver lock classes: - */ - lockdep_set_class(&ops->reg_mutex, (struct lock_class_key *)id); - - ops->driver = DRIVER_EMPTY; - INIT_LIST_HEAD(&ops->dev_list); - /* lock this instance */ - ops->used = 1; - - /* register driver entry */ - mutex_lock(&ops_mutex); - list_add_tail(&ops->list, &opslist); - num_ops++; - mutex_unlock(&ops_mutex); - - return ops; + if (strcmp(sdrv->id, *ptr)) + return 0; /* id don't match, continue the loop */ + *ptr = sdrv; + return 1; /* break the loop */ } - /* * unregister the specified driver */ int snd_seq_device_unregister_driver(char *id) { - struct ops_list *ops; - struct snd_seq_device *dev; + struct snd_seq_driver *sdrv = (struct snd_seq_driver *)id; - ops = find_driver(id, 0); - if (ops == NULL) + if (!bus_for_each_drv(&snd_seq_bus_type, NULL, &sdrv, find_drv)) return -ENXIO; - if (! (ops->driver & DRIVER_LOADED) || - (ops->driver & DRIVER_LOCKED)) { - pr_err("ALSA: seq: driver_unregister: cannot unload driver '%s': status=%x\n", - id, ops->driver); - unlock_driver(ops); - return -EBUSY; - } - - /* close and release all devices associated with this driver */ - mutex_lock(&ops->reg_mutex); - ops->driver |= DRIVER_LOCKED; /* do not remove this driver recursively */ - list_for_each_entry(dev, &ops->dev_list, list) { - free_device(dev, ops); - } - - ops->driver = 0; - if (ops->num_init_devices > 0) - pr_err("ALSA: seq: free_driver: init_devices > 0!! (%d)\n", - ops->num_init_devices); - mutex_unlock(&ops->reg_mutex); - - unlock_driver(ops); - - /* remove empty driver entries */ - remove_drivers(); - + driver_unregister(&sdrv->driver); + kfree(sdrv); return 0; } - - -/* - * remove empty driver entries - */ -static void remove_drivers(void) -{ - struct list_head *head; - - mutex_lock(&ops_mutex); - head = opslist.next; - while (head != &opslist) { - struct ops_list *ops = list_entry(head, struct ops_list, list); - if (! (ops->driver & DRIVER_LOADED) && - ops->used == 0 && ops->num_devices == 0) { - head = head->next; - list_del(&ops->list); - kfree(ops); - num_ops--; - } else - head = head->next; - } - mutex_unlock(&ops_mutex); -} - -/* - * initialize the device - call init_device operator - */ -static int init_device(struct snd_seq_device *dev, struct ops_list *ops) -{ - if (! (ops->driver & DRIVER_LOADED)) - return 0; /* driver is not loaded yet */ - if (dev->status != SNDRV_SEQ_DEVICE_FREE) - return 0; /* already initialized */ - if (ops->argsize != dev->argsize) { - pr_err("ALSA: seq: incompatible device '%s' for plug-in '%s' (%d %d)\n", - dev->name, ops->id, ops->argsize, dev->argsize); - return -EINVAL; - } - if (ops->ops.init_device(dev) >= 0) { - dev->status = SNDRV_SEQ_DEVICE_REGISTERED; - ops->num_init_devices++; - } else { - pr_err("ALSA: seq: init_device failed: %s: %s\n", - dev->name, dev->id); - } - - return 0; -} - -/* - * release the device - call free_device operator - */ -static int free_device(struct snd_seq_device *dev, struct ops_list *ops) -{ - int result; - - if (! (ops->driver & DRIVER_LOADED)) - return 0; /* driver is not loaded yet */ - if (dev->status != SNDRV_SEQ_DEVICE_REGISTERED) - return 0; /* not registered */ - if (ops->argsize != dev->argsize) { - pr_err("ALSA: seq: incompatible device '%s' for plug-in '%s' (%d %d)\n", - dev->name, ops->id, ops->argsize, dev->argsize); - return -EINVAL; - } - if ((result = ops->ops.free_device(dev)) >= 0 || result == -ENXIO) { - dev->status = SNDRV_SEQ_DEVICE_FREE; - dev->driver_data = NULL; - ops->num_init_devices--; - } else { - pr_err("ALSA: seq: free_device failed: %s: %s\n", - dev->name, dev->id); - } - - return 0; -} - -/* - * find the matching driver with given id - */ -static struct ops_list * find_driver(char *id, int create_if_empty) -{ - struct ops_list *ops; - - mutex_lock(&ops_mutex); - list_for_each_entry(ops, &opslist, list) { - if (strcmp(ops->id, id) == 0) { - ops->used++; - mutex_unlock(&ops_mutex); - return ops; - } - } - mutex_unlock(&ops_mutex); - if (create_if_empty) - return create_driver(id); - return NULL; -} - -static void unlock_driver(struct ops_list *ops) -{ - mutex_lock(&ops_mutex); - ops->used--; - mutex_unlock(&ops_mutex); -} - +EXPORT_SYMBOL(snd_seq_device_unregister_driver); /* * module part */ -static int __init alsa_seq_device_init(void) +static int __init seq_dev_proc_init(void) { #ifdef CONFIG_PROC_FS info_entry = snd_info_create_module_entry(THIS_MODULE, "drivers", @@ -594,17 +376,28 @@ static int __init alsa_seq_device_init(void) return 0; } +static int __init alsa_seq_device_init(void) +{ + int err; + + err = bus_register(&snd_seq_bus_type); + if (err < 0) + return err; + err = seq_dev_proc_init(); + if (err < 0) + bus_unregister(&snd_seq_bus_type); + return err; +} + static void __exit alsa_seq_device_exit(void) { #ifdef CONFIG_MODULES cancel_work_sync(&autoload_work); #endif - remove_drivers(); #ifdef CONFIG_PROC_FS snd_info_free_entry(info_entry); #endif - if (num_ops) - pr_err("ALSA: seq: drivers not released (%d)\n", num_ops); + bus_unregister(&snd_seq_bus_type); } module_init(alsa_seq_device_init) From af03c243a1f014145dae34368fe975b2f08ed964 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 13:40:50 +0100 Subject: [PATCH 006/162] ALSA: seq: Clean up device and driver structs Use const string pointer instead of copying the id string to each object. Also drop the status and list fields of snd_seq_device struct that are no longer used. Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 18 ++++++------------ sound/core/seq/seq_device.c | 31 ++++++++++--------------------- 2 files changed, 16 insertions(+), 33 deletions(-) diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index ea256b825146..b13cd2930d32 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -25,24 +25,16 @@ * registered device information */ -#define ID_LEN 32 - -/* status flag */ -#define SNDRV_SEQ_DEVICE_FREE 0 -#define SNDRV_SEQ_DEVICE_REGISTERED 1 - struct snd_seq_device { /* device info */ struct snd_card *card; /* sound card */ int device; /* device number */ - char id[ID_LEN]; /* driver id */ + const char *id; /* driver id */ char name[80]; /* device name */ int argsize; /* size of the argument */ void *driver_data; /* private data for driver */ - int status; /* flag - read only */ void *private_data; /* private data for the caller */ void (*private_free)(struct snd_seq_device *device); - struct list_head list; /* link to next device */ struct device dev; }; @@ -75,9 +67,11 @@ void snd_seq_device_load_drivers(void); #else #define snd_seq_device_load_drivers() #endif -int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, struct snd_seq_device **result); -int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, int argsize); -int snd_seq_device_unregister_driver(char *id); +int snd_seq_device_new(struct snd_card *card, int device, const char *id, + int argsize, struct snd_seq_device **result); +int snd_seq_device_register_driver(const char *id, + struct snd_seq_dev_ops *entry, int argsize); +int snd_seq_device_unregister_driver(const char *id); #define SNDRV_SEQ_DEVICE_ARGPTR(dev) (void *)((char *)(dev) + sizeof(struct snd_seq_device)) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index d3320ffe43c2..49daf6e3a387 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -54,7 +54,7 @@ MODULE_LICENSE("GPL"); struct snd_seq_driver { struct device_driver driver; - char id[ID_LEN]; + const char *id; int argsize; struct snd_seq_dev_ops ops; }; @@ -215,8 +215,8 @@ static void snd_seq_dev_release(struct device *dev) * id = id of driver * result = return pointer (NULL allowed if unnecessary) */ -int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, - struct snd_seq_device **result) +int snd_seq_device_new(struct snd_card *card, int device, const char *id, + int argsize, struct snd_seq_device **result) { struct snd_seq_device *dev; int err; @@ -239,9 +239,8 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, /* set up device info */ dev->card = card; dev->device = device; - strlcpy(dev->id, id, sizeof(dev->id)); + dev->id = id; dev->argsize = argsize; - dev->status = SNDRV_SEQ_DEVICE_FREE; device_initialize(&dev->dev); dev->dev.parent = &card->card_dev; @@ -270,26 +269,16 @@ static int snd_seq_drv_probe(struct device *dev) { struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); struct snd_seq_device *sdev = to_seq_dev(dev); - int err; - err = sdrv->ops.init_device(sdev); - if (err < 0) - return err; - sdev->status = SNDRV_SEQ_DEVICE_REGISTERED; - return 0; + return sdrv->ops.init_device(sdev); } static int snd_seq_drv_remove(struct device *dev) { struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); struct snd_seq_device *sdev = to_seq_dev(dev); - int err; - err = sdrv->ops.free_device(sdev); - if (err < 0) - return err; - sdev->status = SNDRV_SEQ_DEVICE_FREE; - return 0; + return sdrv->ops.free_device(sdev); } /* @@ -297,8 +286,8 @@ static int snd_seq_drv_remove(struct device *dev) * id = driver id * entry = driver operators - duplicated to each instance */ -int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, - int argsize) +int snd_seq_device_register_driver(const char *id, + struct snd_seq_dev_ops *entry, int argsize) { struct snd_seq_driver *sdrv; int err; @@ -315,7 +304,7 @@ int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, sdrv->driver.bus = &snd_seq_bus_type; sdrv->driver.probe = snd_seq_drv_probe; sdrv->driver.remove = snd_seq_drv_remove; - strlcpy(sdrv->id, id, sizeof(sdrv->id)); + sdrv->id = id; sdrv->argsize = argsize; sdrv->ops = *entry; @@ -343,7 +332,7 @@ static int find_drv(struct device_driver *drv, void *data) /* * unregister the specified driver */ -int snd_seq_device_unregister_driver(char *id) +int snd_seq_device_unregister_driver(const char *id) { struct snd_seq_driver *sdrv = (struct snd_seq_driver *)id; From 056622053b8ae02978678ac1321b5bd956e7c812 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 13:43:22 +0100 Subject: [PATCH 007/162] ALSA: seq: Define driver object in each driver This patch moves the driver object initialization and allocation to each driver's module init/exit code like other normal drivers. The snd_seq_driver struct is now published in seq_device.h, and each driver is responsible to define it with proper driver attributes (name, probe and remove) with snd_seq_driver specific attributes as id and argsize fields. The helper functions snd_seq_driver_register(), snd_seq_driver_unregister() and module_snd_seq_driver() are used for simplifying codes. Signed-off-by: Takashi Iwai --- include/sound/seq_device.h | 27 ++++++--- sound/core/seq/oss/seq_oss.c | 20 ++++--- sound/core/seq/oss/seq_oss_synth.c | 6 +- sound/core/seq/oss/seq_oss_synth.h | 4 +- sound/core/seq/seq_device.c | 93 ++++-------------------------- sound/core/seq/seq_midi.c | 28 +++++---- sound/drivers/opl3/opl3_seq.c | 34 +++++------ sound/drivers/opl4/opl4_seq.c | 33 +++++------ sound/isa/sb/emu8000_synth.c | 33 +++++------ sound/pci/emu10k1/emu10k1_synth.c | 33 +++++------ 10 files changed, 122 insertions(+), 189 deletions(-) diff --git a/include/sound/seq_device.h b/include/sound/seq_device.h index b13cd2930d32..ddc0d504cf39 100644 --- a/include/sound/seq_device.h +++ b/include/sound/seq_device.h @@ -41,8 +41,10 @@ struct snd_seq_device { #define to_seq_dev(_dev) \ container_of(_dev, struct snd_seq_device, dev) +/* sequencer driver */ + /* driver operators - * init_device: + * probe: * Initialize the device with given parameters. * Typically, * 1. call snd_hwdep_new @@ -50,15 +52,19 @@ struct snd_seq_device { * 3. call snd_hwdep_register * 4. store the instance to dev->driver_data pointer. * - * free_device: + * remove: * Release the private data. * Typically, call snd_device_free(dev->card, dev->driver_data) */ -struct snd_seq_dev_ops { - int (*init_device)(struct snd_seq_device *dev); - int (*free_device)(struct snd_seq_device *dev); +struct snd_seq_driver { + struct device_driver driver; + char *id; + int argsize; }; +#define to_seq_drv(_drv) \ + container_of(_drv, struct snd_seq_driver, driver) + /* * prototypes */ @@ -69,12 +75,17 @@ void snd_seq_device_load_drivers(void); #endif int snd_seq_device_new(struct snd_card *card, int device, const char *id, int argsize, struct snd_seq_device **result); -int snd_seq_device_register_driver(const char *id, - struct snd_seq_dev_ops *entry, int argsize); -int snd_seq_device_unregister_driver(const char *id); #define SNDRV_SEQ_DEVICE_ARGPTR(dev) (void *)((char *)(dev) + sizeof(struct snd_seq_device)) +int __must_check __snd_seq_driver_register(struct snd_seq_driver *drv, + struct module *mod); +#define snd_seq_driver_register(drv) \ + __snd_seq_driver_register(drv, THIS_MODULE) +void snd_seq_driver_unregister(struct snd_seq_driver *drv); + +#define module_snd_seq_driver(drv) \ + module_driver(drv, snd_seq_driver_register, snd_seq_driver_unregister) /* * id strings for generic devices diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 16d42679e43f..ae1814aa767e 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -65,13 +65,19 @@ static unsigned int odev_poll(struct file *file, poll_table * wait); * module interface */ +static struct snd_seq_driver seq_oss_synth_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_seq_oss_synth_probe, + .remove = snd_seq_oss_synth_remove, + }, + .id = SNDRV_SEQ_DEV_ID_OSS, + .argsize = sizeof(struct snd_seq_oss_reg), +}; + static int __init alsa_seq_oss_init(void) { int rc; - static struct snd_seq_dev_ops ops = { - snd_seq_oss_synth_register, - snd_seq_oss_synth_unregister, - }; snd_seq_autoload_lock(); if ((rc = register_device()) < 0) @@ -86,8 +92,8 @@ static int __init alsa_seq_oss_init(void) goto error; } - if ((rc = snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OSS, &ops, - sizeof(struct snd_seq_oss_reg))) < 0) { + rc = snd_seq_driver_register(&seq_oss_synth_driver); + if (rc < 0) { snd_seq_oss_delete_client(); unregister_proc(); unregister_device(); @@ -104,7 +110,7 @@ static int __init alsa_seq_oss_init(void) static void __exit alsa_seq_oss_exit(void) { - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OSS); + snd_seq_driver_unregister(&seq_oss_synth_driver); snd_seq_oss_delete_client(); unregister_proc(); unregister_device(); diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 701feb71b700..835edc80f918 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -98,8 +98,9 @@ snd_seq_oss_synth_init(void) * registration of the synth device */ int -snd_seq_oss_synth_register(struct snd_seq_device *dev) +snd_seq_oss_synth_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); int i; struct seq_oss_synth *rec; struct snd_seq_oss_reg *reg = SNDRV_SEQ_DEVICE_ARGPTR(dev); @@ -149,8 +150,9 @@ snd_seq_oss_synth_register(struct snd_seq_device *dev) int -snd_seq_oss_synth_unregister(struct snd_seq_device *dev) +snd_seq_oss_synth_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); int index; struct seq_oss_synth *rec = dev->driver_data; unsigned long flags; diff --git a/sound/core/seq/oss/seq_oss_synth.h b/sound/core/seq/oss/seq_oss_synth.h index dbdfcbb80eaa..74ac55f166b6 100644 --- a/sound/core/seq/oss/seq_oss_synth.h +++ b/sound/core/seq/oss/seq_oss_synth.h @@ -28,8 +28,8 @@ #include void snd_seq_oss_synth_init(void); -int snd_seq_oss_synth_register(struct snd_seq_device *dev); -int snd_seq_oss_synth_unregister(struct snd_seq_device *dev); +int snd_seq_oss_synth_probe(struct device *dev); +int snd_seq_oss_synth_remove(struct device *dev); void snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp); void snd_seq_oss_synth_setup_midi(struct seq_oss_devinfo *dp); void snd_seq_oss_synth_cleanup(struct seq_oss_devinfo *dp); diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 49daf6e3a387..48b20f009598 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -52,16 +52,6 @@ MODULE_AUTHOR("Takashi Iwai "); MODULE_DESCRIPTION("ALSA sequencer device management"); MODULE_LICENSE("GPL"); -struct snd_seq_driver { - struct device_driver driver; - const char *id; - int argsize; - struct snd_seq_dev_ops ops; -}; - -#define to_seq_drv(_drv) \ - container_of(_drv, struct snd_seq_driver, driver) - /* * bus definition */ @@ -263,86 +253,23 @@ int snd_seq_device_new(struct snd_card *card, int device, const char *id, EXPORT_SYMBOL(snd_seq_device_new); /* - * driver binding - just pass to each driver callback + * driver registration */ -static int snd_seq_drv_probe(struct device *dev) +int __snd_seq_driver_register(struct snd_seq_driver *drv, struct module *mod) { - struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); - struct snd_seq_device *sdev = to_seq_dev(dev); - - return sdrv->ops.init_device(sdev); -} - -static int snd_seq_drv_remove(struct device *dev) -{ - struct snd_seq_driver *sdrv = to_seq_drv(dev->driver); - struct snd_seq_device *sdev = to_seq_dev(dev); - - return sdrv->ops.free_device(sdev); -} - -/* - * register device driver - * id = driver id - * entry = driver operators - duplicated to each instance - */ -int snd_seq_device_register_driver(const char *id, - struct snd_seq_dev_ops *entry, int argsize) -{ - struct snd_seq_driver *sdrv; - int err; - - if (id == NULL || entry == NULL || - entry->init_device == NULL || entry->free_device == NULL) + if (WARN_ON(!drv->driver.name || !drv->id)) return -EINVAL; - - sdrv = kzalloc(sizeof(*sdrv), GFP_KERNEL); - if (!sdrv) - return -ENOMEM; - - sdrv->driver.name = id; - sdrv->driver.bus = &snd_seq_bus_type; - sdrv->driver.probe = snd_seq_drv_probe; - sdrv->driver.remove = snd_seq_drv_remove; - sdrv->id = id; - sdrv->argsize = argsize; - sdrv->ops = *entry; - - err = driver_register(&sdrv->driver); - if (err < 0) - kfree(sdrv); - return err; + drv->driver.bus = &snd_seq_bus_type; + drv->driver.owner = mod; + return driver_register(&drv->driver); } -EXPORT_SYMBOL(snd_seq_device_register_driver); +EXPORT_SYMBOL_GPL(__snd_seq_driver_register); -/* callback to find a specific driver; data is a pointer to the id string ptr. - * when the id matches, store the driver pointer in return and break the loop. - */ -static int find_drv(struct device_driver *drv, void *data) +void snd_seq_driver_unregister(struct snd_seq_driver *drv) { - struct snd_seq_driver *sdrv = to_seq_drv(drv); - void **ptr = (void **)data; - - if (strcmp(sdrv->id, *ptr)) - return 0; /* id don't match, continue the loop */ - *ptr = sdrv; - return 1; /* break the loop */ + driver_unregister(&drv->driver); } - -/* - * unregister the specified driver - */ -int snd_seq_device_unregister_driver(const char *id) -{ - struct snd_seq_driver *sdrv = (struct snd_seq_driver *)id; - - if (!bus_for_each_drv(&snd_seq_bus_type, NULL, &sdrv, find_drv)) - return -ENXIO; - driver_unregister(&sdrv->driver); - kfree(sdrv); - return 0; -} -EXPORT_SYMBOL(snd_seq_device_unregister_driver); +EXPORT_SYMBOL_GPL(snd_seq_driver_unregister); /* * module part diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 68fec776da26..79c73119cedc 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -273,8 +273,9 @@ static void snd_seq_midisynth_delete(struct seq_midisynth *msynth) /* register new midi synth port */ static int -snd_seq_midisynth_register_port(struct snd_seq_device *dev) +snd_seq_midisynth_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct seq_midisynth_client *client; struct seq_midisynth *msynth, *ms; struct snd_seq_port_info *port; @@ -427,8 +428,9 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev) /* release midi synth port */ static int -snd_seq_midisynth_unregister_port(struct snd_seq_device *dev) +snd_seq_midisynth_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct seq_midisynth_client *client; struct seq_midisynth *msynth; struct snd_card *card = dev->card; @@ -457,23 +459,29 @@ snd_seq_midisynth_unregister_port(struct snd_seq_device *dev) return 0; } +static struct snd_seq_driver seq_midisynth_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_seq_midisynth_probe, + .remove = snd_seq_midisynth_remove, + }, + .id = SNDRV_SEQ_DEV_ID_MIDISYNTH, + .argsize = 0, +}; static int __init alsa_seq_midi_init(void) { - static struct snd_seq_dev_ops ops = { - snd_seq_midisynth_register_port, - snd_seq_midisynth_unregister_port, - }; - memset(&synths, 0, sizeof(synths)); + int err; + snd_seq_autoload_lock(); - snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_MIDISYNTH, &ops, 0); + err = snd_seq_driver_register(&seq_midisynth_driver); snd_seq_autoload_unlock(); - return 0; + return err; } static void __exit alsa_seq_midi_exit(void) { - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_MIDISYNTH); + snd_seq_driver_unregister(&seq_midisynth_driver); } module_init(alsa_seq_midi_init) diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index a9f618e06a22..fdae5d7f421f 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -216,8 +216,9 @@ static int snd_opl3_synth_create_port(struct snd_opl3 * opl3) /* ------------------------------ */ -static int snd_opl3_seq_new_device(struct snd_seq_device *dev) +static int snd_opl3_seq_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl3 *opl3; int client, err; char name[32]; @@ -257,8 +258,9 @@ static int snd_opl3_seq_new_device(struct snd_seq_device *dev) return 0; } -static int snd_opl3_seq_delete_device(struct snd_seq_device *dev) +static int snd_opl3_seq_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl3 *opl3; opl3 = *(struct snd_opl3 **)SNDRV_SEQ_DEVICE_ARGPTR(dev); @@ -275,22 +277,14 @@ static int snd_opl3_seq_delete_device(struct snd_seq_device *dev) return 0; } -static int __init alsa_opl3_seq_init(void) -{ - static struct snd_seq_dev_ops ops = - { - snd_opl3_seq_new_device, - snd_opl3_seq_delete_device - }; +static struct snd_seq_driver opl3_seq_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_opl3_seq_probe, + .remove = snd_opl3_seq_remove, + }, + .id = SNDRV_SEQ_DEV_ID_OPL3, + .argsize = sizeof(struct snd_opl3 *), +}; - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OPL3, &ops, - sizeof(struct snd_opl3 *)); -} - -static void __exit alsa_opl3_seq_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OPL3); -} - -module_init(alsa_opl3_seq_init) -module_exit(alsa_opl3_seq_exit) +module_snd_seq_driver(opl3_seq_driver); diff --git a/sound/drivers/opl4/opl4_seq.c b/sound/drivers/opl4/opl4_seq.c index 99197699c55a..03d6202f4829 100644 --- a/sound/drivers/opl4/opl4_seq.c +++ b/sound/drivers/opl4/opl4_seq.c @@ -124,8 +124,9 @@ static void snd_opl4_seq_free_port(void *private_data) snd_midi_channel_free_set(opl4->chset); } -static int snd_opl4_seq_new_device(struct snd_seq_device *dev) +static int snd_opl4_seq_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl4 *opl4; int client; struct snd_seq_port_callback pcallbacks; @@ -180,8 +181,9 @@ static int snd_opl4_seq_new_device(struct snd_seq_device *dev) return 0; } -static int snd_opl4_seq_delete_device(struct snd_seq_device *dev) +static int snd_opl4_seq_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_opl4 *opl4; opl4 = *(struct snd_opl4 **)SNDRV_SEQ_DEVICE_ARGPTR(dev); @@ -195,21 +197,14 @@ static int snd_opl4_seq_delete_device(struct snd_seq_device *dev) return 0; } -static int __init alsa_opl4_synth_init(void) -{ - static struct snd_seq_dev_ops ops = { - snd_opl4_seq_new_device, - snd_opl4_seq_delete_device - }; +static struct snd_seq_driver opl4_seq_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_opl4_seq_probe, + .remove = snd_opl4_seq_remove, + }, + .id = SNDRV_SEQ_DEV_ID_OPL4, + .argsize = sizeof(struct snd_opl4 *), +}; - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_OPL4, &ops, - sizeof(struct snd_opl4 *)); -} - -static void __exit alsa_opl4_synth_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_OPL4); -} - -module_init(alsa_opl4_synth_init) -module_exit(alsa_opl4_synth_exit) +module_snd_seq_driver(opl4_seq_driver); diff --git a/sound/isa/sb/emu8000_synth.c b/sound/isa/sb/emu8000_synth.c index 72332dfada9a..4aa719cad331 100644 --- a/sound/isa/sb/emu8000_synth.c +++ b/sound/isa/sb/emu8000_synth.c @@ -34,8 +34,9 @@ MODULE_LICENSE("GPL"); /* * create a new hardware dependent device for Emu8000 */ -static int snd_emu8000_new_device(struct snd_seq_device *dev) +static int snd_emu8000_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emu8000 *hw; struct snd_emux *emu; @@ -93,8 +94,9 @@ static int snd_emu8000_new_device(struct snd_seq_device *dev) /* * free all resources */ -static int snd_emu8000_delete_device(struct snd_seq_device *dev) +static int snd_emu8000_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emu8000 *hw; if (dev->driver_data == NULL) @@ -114,21 +116,14 @@ static int snd_emu8000_delete_device(struct snd_seq_device *dev) * INIT part */ -static int __init alsa_emu8000_init(void) -{ - - static struct snd_seq_dev_ops ops = { - snd_emu8000_new_device, - snd_emu8000_delete_device, - }; - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_EMU8000, &ops, - sizeof(struct snd_emu8000*)); -} +static struct snd_seq_driver emu8000_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_emu8000_probe, + .remove = snd_emu8000_remove, + }, + .id = SNDRV_SEQ_DEV_ID_EMU8000, + .argsize = sizeof(struct snd_emu8000 *), +}; -static void __exit alsa_emu8000_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_EMU8000); -} - -module_init(alsa_emu8000_init) -module_exit(alsa_emu8000_exit) +module_snd_seq_driver(emu8000_driver); diff --git a/sound/pci/emu10k1/emu10k1_synth.c b/sound/pci/emu10k1/emu10k1_synth.c index 4c41c903a840..5457d5613f6b 100644 --- a/sound/pci/emu10k1/emu10k1_synth.c +++ b/sound/pci/emu10k1/emu10k1_synth.c @@ -29,8 +29,9 @@ MODULE_LICENSE("GPL"); /* * create a new hardware dependent device for Emu10k1 */ -static int snd_emu10k1_synth_new_device(struct snd_seq_device *dev) +static int snd_emu10k1_synth_probe(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emux *emux; struct snd_emu10k1 *hw; struct snd_emu10k1_synth_arg *arg; @@ -79,8 +80,9 @@ static int snd_emu10k1_synth_new_device(struct snd_seq_device *dev) return 0; } -static int snd_emu10k1_synth_delete_device(struct snd_seq_device *dev) +static int snd_emu10k1_synth_remove(struct device *_dev) { + struct snd_seq_device *dev = to_seq_dev(_dev); struct snd_emux *emux; struct snd_emu10k1 *hw; unsigned long flags; @@ -104,21 +106,14 @@ static int snd_emu10k1_synth_delete_device(struct snd_seq_device *dev) * INIT part */ -static int __init alsa_emu10k1_synth_init(void) -{ - - static struct snd_seq_dev_ops ops = { - snd_emu10k1_synth_new_device, - snd_emu10k1_synth_delete_device, - }; - return snd_seq_device_register_driver(SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH, &ops, - sizeof(struct snd_emu10k1_synth_arg)); -} +static struct snd_seq_driver emu10k1_synth_driver = { + .driver = { + .name = KBUILD_MODNAME, + .probe = snd_emu10k1_synth_probe, + .remove = snd_emu10k1_synth_remove, + }, + .id = SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH, + .argsize = sizeof(struct snd_emu10k1_synth_arg), +}; -static void __exit alsa_emu10k1_synth_exit(void) -{ - snd_seq_device_unregister_driver(SNDRV_SEQ_DEV_ID_EMU10K1_SYNTH); -} - -module_init(alsa_emu10k1_synth_init) -module_exit(alsa_emu10k1_synth_exit) +module_snd_seq_driver(emu10k1_synth_driver); From 54a721abd7953a58e5479065c0cfdd8679d785c9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Feb 2015 14:20:24 +0100 Subject: [PATCH 008/162] ALSA: seq: Drop snd_seq_autoload_lock() and _unlock() The autoload lock became already superfluous due to the recent rework of autoload code. Let's drop them now. This allows us to simplify a few codes nicely. Signed-off-by: Takashi Iwai --- include/sound/seq_kernel.h | 6 +----- sound/core/seq/oss/seq_oss.c | 2 -- sound/core/seq/seq_device.c | 19 +++++++------------ sound/core/seq/seq_dummy.c | 6 +----- sound/core/seq/seq_midi.c | 18 +----------------- 5 files changed, 10 insertions(+), 41 deletions(-) diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h index 18a2ac58b88f..feb58d455560 100644 --- a/include/sound/seq_kernel.h +++ b/include/sound/seq_kernel.h @@ -99,13 +99,9 @@ int snd_seq_event_port_attach(int client, struct snd_seq_port_callback *pcbp, int snd_seq_event_port_detach(int client, int port); #ifdef CONFIG_MODULES -void snd_seq_autoload_lock(void); -void snd_seq_autoload_unlock(void); void snd_seq_autoload_init(void); -#define snd_seq_autoload_exit() snd_seq_autoload_lock() +void snd_seq_autoload_exit(void); #else -#define snd_seq_autoload_lock() -#define snd_seq_autoload_unlock() #define snd_seq_autoload_init() #define snd_seq_autoload_exit() #endif diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index ae1814aa767e..72873a46afeb 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -79,7 +79,6 @@ static int __init alsa_seq_oss_init(void) { int rc; - snd_seq_autoload_lock(); if ((rc = register_device()) < 0) goto error; if ((rc = register_proc()) < 0) { @@ -104,7 +103,6 @@ static int __init alsa_seq_oss_init(void) snd_seq_oss_synth_init(); error: - snd_seq_autoload_unlock(); return rc; } diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 48b20f009598..355b34269bd1 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -98,19 +98,8 @@ static void snd_seq_device_info(struct snd_info_entry *entry, */ #ifdef CONFIG_MODULES -/* avoid auto-loading during module_init() */ +/* flag to block auto-loading */ static atomic_t snd_seq_in_init = ATOMIC_INIT(1); /* blocked as default */ -void snd_seq_autoload_lock(void) -{ - atomic_inc(&snd_seq_in_init); -} -EXPORT_SYMBOL(snd_seq_autoload_lock); - -void snd_seq_autoload_unlock(void) -{ - atomic_dec(&snd_seq_in_init); -} -EXPORT_SYMBOL(snd_seq_autoload_unlock); static int request_seq_drv(struct device *dev, void *data) { @@ -147,6 +136,12 @@ void snd_seq_autoload_init(void) } EXPORT_SYMBOL(snd_seq_autoload_init); +void snd_seq_autoload_exit(void) +{ + atomic_inc(&snd_seq_in_init); +} +EXPORT_SYMBOL(snd_seq_autoload_exit); + void snd_seq_device_load_drivers(void) { queue_autoload_drivers(); diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index 5d905d90d504..d3a2ec4f0561 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -214,11 +214,7 @@ delete_client(void) static int __init alsa_seq_dummy_init(void) { - int err; - snd_seq_autoload_lock(); - err = register_client(); - snd_seq_autoload_unlock(); - return err; + return register_client(); } static void __exit alsa_seq_dummy_exit(void) diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index 79c73119cedc..5dd0ee258359 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -469,20 +469,4 @@ static struct snd_seq_driver seq_midisynth_driver = { .argsize = 0, }; -static int __init alsa_seq_midi_init(void) -{ - int err; - - snd_seq_autoload_lock(); - err = snd_seq_driver_register(&seq_midisynth_driver); - snd_seq_autoload_unlock(); - return err; -} - -static void __exit alsa_seq_midi_exit(void) -{ - snd_seq_driver_unregister(&seq_midisynth_driver); -} - -module_init(alsa_seq_midi_init) -module_exit(alsa_seq_midi_exit) +module_snd_seq_driver(seq_midisynth_driver); From 226e2f1b0bb4a5f724dd119c1eeb8b8e89e87fab Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 12 Feb 2015 16:41:26 +0200 Subject: [PATCH 009/162] ASoC: davinci-mcasp: Add support for CBS_CFM mode Support for setups where codec is bitclock slave and frame master. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de3b155a5011..031c1fb44ae7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -441,6 +441,18 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); mcasp->bclk_master = 1; break; + case SND_SOC_DAIFMT_CBS_CFM: + /* codec is clock slave and frame master */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); + mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + mcasp->bclk_master = 1; + break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); From 38ebb7034970efe5c7419267e499295e5893b565 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 6 Feb 2015 14:45:33 +0100 Subject: [PATCH 010/162] ALSA: Consolidate snd_find_free_minor() A really small cleanup to consolidate snd_find_free_minor() and snd_kernel_minor() so that we can get rid of one more ifdef. Signed-off-by: Takashi Iwai --- sound/core/sound.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/core/sound.c b/sound/core/sound.c index 185cec01ee25..5fc93d00572a 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -186,7 +186,7 @@ static const struct file_operations snd_fops = }; #ifdef CONFIG_SND_DYNAMIC_MINORS -static int snd_find_free_minor(int type) +static int snd_find_free_minor(int type, struct snd_card *card, int dev) { int minor; @@ -209,7 +209,7 @@ static int snd_find_free_minor(int type) return -EBUSY; } #else -static int snd_kernel_minor(int type, struct snd_card *card, int dev) +static int snd_find_free_minor(int type, struct snd_card *card, int dev) { int minor; @@ -237,6 +237,8 @@ static int snd_kernel_minor(int type, struct snd_card *card, int dev) } if (snd_BUG_ON(minor < 0 || minor >= SNDRV_OS_MINORS)) return -EINVAL; + if (snd_minors[minor]) + return -EBUSY; return minor; } #endif @@ -276,13 +278,7 @@ int snd_register_device(int type, struct snd_card *card, int dev, preg->private_data = private_data; preg->card_ptr = card; mutex_lock(&sound_mutex); -#ifdef CONFIG_SND_DYNAMIC_MINORS - minor = snd_find_free_minor(type); -#else - minor = snd_kernel_minor(type, card, dev); - if (minor >= 0 && snd_minors[minor]) - minor = -EBUSY; -#endif + minor = snd_find_free_minor(type, card, dev); if (minor < 0) { err = minor; goto error; From 5ecc5dc720307d3fb0167e2b14f50e97dd9a2233 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Tue, 17 Feb 2015 00:05:04 +0100 Subject: [PATCH 011/162] ALSA: hdspm - DRY cleanup in .open callbacks This commit removes code duplication between snd_hdspm_{capture,playback}_open. No semantic changes intended, this is purely cosmetic. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 100 +++++++++----------------------------- 1 file changed, 24 insertions(+), 76 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ca67f896d117..51e984170b3d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6043,23 +6043,30 @@ hdspm_hw_constraints_aes32_sample_rates = { .mask = 0 }; -static int snd_hdspm_playback_open(struct snd_pcm_substream *substream) +static int snd_hdspm_open(struct snd_pcm_substream *substream) { struct hdspm *hdspm = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); spin_lock_irq(&hdspm->lock); - snd_pcm_set_sync(substream); + runtime->hw = (playback) ? snd_hdspm_playback_subinfo : + snd_hdspm_capture_subinfo; + if (playback) { + if (hdspm->capture_substream == NULL) + hdspm_stop_audio(hdspm); - runtime->hw = snd_hdspm_playback_subinfo; + hdspm->playback_pid = current->pid; + hdspm->playback_substream = substream; + } else { + if (hdspm->playback_substream == NULL) + hdspm_stop_audio(hdspm); - if (hdspm->capture_substream == NULL) - hdspm_stop_audio(hdspm); - - hdspm->playback_pid = current->pid; - hdspm->playback_substream = substream; + hdspm->capture_pid = current->pid; + hdspm->capture_substream = substream; + } spin_unlock_irq(&hdspm->lock); @@ -6094,16 +6101,20 @@ static int snd_hdspm_playback_open(struct snd_pcm_substream *substream) &hdspm_hw_constraints_aes32_sample_rates); } else { snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - snd_hdspm_hw_rule_rate_out_channels, hdspm, + (playback ? + snd_hdspm_hw_rule_rate_out_channels : + snd_hdspm_hw_rule_rate_in_channels), hdspm, SNDRV_PCM_HW_PARAM_CHANNELS, -1); } snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_out_channels, hdspm, + (playback ? snd_hdspm_hw_rule_out_channels : + snd_hdspm_hw_rule_in_channels), hdspm, SNDRV_PCM_HW_PARAM_CHANNELS, -1); snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_out_channels_rate, hdspm, + (playback ? snd_hdspm_hw_rule_out_channels_rate : + snd_hdspm_hw_rule_in_channels_rate), hdspm, SNDRV_PCM_HW_PARAM_RATE, -1); return 0; @@ -6123,69 +6134,6 @@ static int snd_hdspm_playback_release(struct snd_pcm_substream *substream) return 0; } - -static int snd_hdspm_capture_open(struct snd_pcm_substream *substream) -{ - struct hdspm *hdspm = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - spin_lock_irq(&hdspm->lock); - snd_pcm_set_sync(substream); - runtime->hw = snd_hdspm_capture_subinfo; - - if (hdspm->playback_substream == NULL) - hdspm_stop_audio(hdspm); - - hdspm->capture_pid = current->pid; - hdspm->capture_substream = substream; - - spin_unlock_irq(&hdspm->lock); - - snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); - snd_pcm_hw_constraint_pow2(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE); - - switch (hdspm->io_type) { - case AIO: - case RayDAT: - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - 32, 4096); - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, - 16384, 16384); - break; - - default: - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIOD_SIZE, - 64, 8192); - snd_pcm_hw_constraint_minmax(runtime, - SNDRV_PCM_HW_PARAM_PERIODS, - 2, 2); - break; - } - - if (AES32 == hdspm->io_type) { - runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - &hdspm_hw_constraints_aes32_sample_rates); - } else { - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - snd_hdspm_hw_rule_rate_in_channels, hdspm, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - } - - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_in_channels, hdspm, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - - snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - snd_hdspm_hw_rule_in_channels_rate, hdspm, - SNDRV_PCM_HW_PARAM_RATE, -1); - - return 0; -} - static int snd_hdspm_capture_release(struct snd_pcm_substream *substream) { struct hdspm *hdspm = snd_pcm_substream_chip(substream); @@ -6414,7 +6362,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, } static struct snd_pcm_ops snd_hdspm_playback_ops = { - .open = snd_hdspm_playback_open, + .open = snd_hdspm_open, .close = snd_hdspm_playback_release, .ioctl = snd_hdspm_ioctl, .hw_params = snd_hdspm_hw_params, @@ -6426,7 +6374,7 @@ static struct snd_pcm_ops snd_hdspm_playback_ops = { }; static struct snd_pcm_ops snd_hdspm_capture_ops = { - .open = snd_hdspm_capture_open, + .open = snd_hdspm_open, .close = snd_hdspm_capture_release, .ioctl = snd_hdspm_ioctl, .hw_params = snd_hdspm_hw_params, From 8b73b867294364ae3007affdf8cfd28f022b158c Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Tue, 17 Feb 2015 00:05:05 +0100 Subject: [PATCH 012/162] ALSA: hdspm - DRY cleanup in .release callback This commit removes code duplication between snd_hdspm_{capture,playback}_release. No semantic changes intended, this is purely cosmetic. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 29 +++++++++++------------------ 1 file changed, 11 insertions(+), 18 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 51e984170b3d..4e1cfb91a8d8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6120,33 +6120,26 @@ static int snd_hdspm_open(struct snd_pcm_substream *substream) return 0; } -static int snd_hdspm_playback_release(struct snd_pcm_substream *substream) +static int snd_hdspm_release(struct snd_pcm_substream *substream) { struct hdspm *hdspm = snd_pcm_substream_chip(substream); + bool playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); spin_lock_irq(&hdspm->lock); - hdspm->playback_pid = -1; - hdspm->playback_substream = NULL; + if (playback) { + hdspm->playback_pid = -1; + hdspm->playback_substream = NULL; + } else { + hdspm->capture_pid = -1; + hdspm->capture_substream = NULL; + } spin_unlock_irq(&hdspm->lock); return 0; } -static int snd_hdspm_capture_release(struct snd_pcm_substream *substream) -{ - struct hdspm *hdspm = snd_pcm_substream_chip(substream); - - spin_lock_irq(&hdspm->lock); - - hdspm->capture_pid = -1; - hdspm->capture_substream = NULL; - - spin_unlock_irq(&hdspm->lock); - return 0; -} - static int snd_hdspm_hwdep_dummy_op(struct snd_hwdep *hw, struct file *file) { /* we have nothing to initialize but the call is required */ @@ -6363,7 +6356,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, static struct snd_pcm_ops snd_hdspm_playback_ops = { .open = snd_hdspm_open, - .close = snd_hdspm_playback_release, + .close = snd_hdspm_release, .ioctl = snd_hdspm_ioctl, .hw_params = snd_hdspm_hw_params, .hw_free = snd_hdspm_hw_free, @@ -6375,7 +6368,7 @@ static struct snd_pcm_ops snd_hdspm_playback_ops = { static struct snd_pcm_ops snd_hdspm_capture_ops = { .open = snd_hdspm_open, - .close = snd_hdspm_capture_release, + .close = snd_hdspm_release, .ioctl = snd_hdspm_ioctl, .hw_params = snd_hdspm_hw_params, .hw_free = snd_hdspm_hw_free, From 0c8d948565490d2a2db9d9a5aec388342c7d38ce Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Tue, 17 Feb 2015 00:05:06 +0100 Subject: [PATCH 013/162] ALSA: hdspm - DRY cleanup of snd_pcm_ops This commit removes code duplication between snd_hdspm_{capture,playback}_ops. No semantic changes intended, this is purely cosmetic. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 18 +++--------------- 1 file changed, 3 insertions(+), 15 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 4e1cfb91a8d8..cb666c73712d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6354,19 +6354,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, return 0; } -static struct snd_pcm_ops snd_hdspm_playback_ops = { - .open = snd_hdspm_open, - .close = snd_hdspm_release, - .ioctl = snd_hdspm_ioctl, - .hw_params = snd_hdspm_hw_params, - .hw_free = snd_hdspm_hw_free, - .prepare = snd_hdspm_prepare, - .trigger = snd_hdspm_trigger, - .pointer = snd_hdspm_hw_pointer, - .page = snd_pcm_sgbuf_ops_page, -}; - -static struct snd_pcm_ops snd_hdspm_capture_ops = { +static struct snd_pcm_ops snd_hdspm_ops = { .open = snd_hdspm_open, .close = snd_hdspm_release, .ioctl = snd_hdspm_ioctl, @@ -6462,9 +6450,9 @@ static int snd_hdspm_create_pcm(struct snd_card *card, strcpy(pcm->name, hdspm->card_name); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, - &snd_hdspm_playback_ops); + &snd_hdspm_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, - &snd_hdspm_capture_ops); + &snd_hdspm_ops); pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; From 76a3aeac2f6c02ecf065fa9baa279dd54bf2d819 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:27 +0100 Subject: [PATCH 014/162] hdspm.h: include stdint.h in userspace MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes compilation error: sound/hdspm.h:43:2: error: unknown type name ‘uint32_t’ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/hdspm.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index b357f1a5e29c..5737332d38f2 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -20,6 +20,12 @@ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#ifdef __KERNEL__ +#include +#else +#include +#endif + /* Maximum channels is 64 even on 56Mode you have 64playbacks to matrix */ #define HDSPM_MAX_CHANNELS 64 From 4bebf7091aa15ec60edf0dcbc654410a87ca21fe Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:38 +0100 Subject: [PATCH 015/162] include/uapi/sound/asound.h: include stdlib.h in userspace MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes compiler errors like: error: field ‘trigger_tstamp’ has incomplete type error: invalid application of ‘sizeof’ to incomplete t ype ‘struct timespec’ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1f23cd635957..1fe3f4f3d696 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -25,6 +25,9 @@ #include +#ifndef __KERNEL__ +#include +#endif /* * protocol version From bbf91c1c5bfc00c2961f15657359ee7e87de4269 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:42 +0100 Subject: [PATCH 016/162] include/uapi/sound/asequencer.h: include sound/asound.h MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes userspace compilation error: error: unknown type name ‘snd_seq_client_type_t’ snd_seq_client_type_t type; /* client type */ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/asequencer.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 09c8a00ea503..5a5fa4956ebd 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -22,6 +22,7 @@ #ifndef _UAPI__SOUND_ASEQUENCER_H #define _UAPI__SOUND_ASEQUENCER_H +#include /** version of the sequencer */ #define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) From b9956409c281931c74ba8d0a2b61a98076a58602 Mon Sep 17 00:00:00 2001 From: Mikko Rapeli Date: Tue, 17 Feb 2015 00:05:43 +0100 Subject: [PATCH 017/162] include/uapi/sound/emu10k1.h: include sound/asound.h MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixes userspace compilation errors like: error: field ‘id’ has incomplete type struct snd_ctl_elem_id id; /* full control ID definition */ Signed-off-by: Mikko Rapeli Signed-off-by: Takashi Iwai --- include/uapi/sound/emu10k1.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/uapi/sound/emu10k1.h b/include/uapi/sound/emu10k1.h index d1bbaf78457a..ec1535bb6aed 100644 --- a/include/uapi/sound/emu10k1.h +++ b/include/uapi/sound/emu10k1.h @@ -23,8 +23,7 @@ #define _UAPI__SOUND_EMU10K1_H #include - - +#include /* * ---- FX8010 ---- From ef7449780ebb596b47d5019eb7ba7878c30a3404 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2015 13:56:29 +0100 Subject: [PATCH 018/162] ALSA: hda - Drop hda_bus_template for snd_hda_bus_new() Instead of copying from the given template, let the caller fills the fields after creation. This simplifies the code after all. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 15 +--------- sound/pci/hda/hda_codec.h | 13 +------- sound/pci/hda/hda_controller.c | 54 +++++++++++++++++++--------------- 3 files changed, 32 insertions(+), 50 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2fe86d2e1b09..215bf048c668 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -861,14 +861,12 @@ static int snd_hda_bus_dev_disconnect(struct snd_device *device) /** * snd_hda_bus_new - create a HDA bus * @card: the card entry - * @temp: the template for hda_bus information * @busp: the pointer to store the created bus instance * * Returns 0 if successful, or a negative error code. */ int snd_hda_bus_new(struct snd_card *card, - const struct hda_bus_template *temp, - struct hda_bus **busp) + struct hda_bus **busp) { struct hda_bus *bus; int err; @@ -877,11 +875,6 @@ int snd_hda_bus_new(struct snd_card *card, .dev_free = snd_hda_bus_dev_free, }; - if (snd_BUG_ON(!temp)) - return -EINVAL; - if (snd_BUG_ON(!temp->ops.command || !temp->ops.get_response)) - return -EINVAL; - if (busp) *busp = NULL; @@ -892,12 +885,6 @@ int snd_hda_bus_new(struct snd_card *card, } bus->card = card; - bus->private_data = temp->private_data; - bus->pci = temp->pci; - bus->modelname = temp->modelname; - bus->power_save = temp->power_save; - bus->ops = temp->ops; - mutex_init(&bus->cmd_mutex); mutex_init(&bus->prepare_mutex); INIT_LIST_HEAD(&bus->codec_list); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9c8820f21f94..5a6594884069 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -101,15 +101,6 @@ struct hda_bus_ops { #endif }; -/* template to pass to the bus constructor */ -struct hda_bus_template { - void *private_data; - struct pci_dev *pci; - const char *modelname; - int *power_save; - struct hda_bus_ops ops; -}; - /* * codec bus * @@ -119,7 +110,6 @@ struct hda_bus_template { struct hda_bus { struct snd_card *card; - /* copied from template */ void *private_data; struct pci_dev *pci; const char *modelname; @@ -420,8 +410,7 @@ enum { /* * constructors */ -int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, - struct hda_bus **busp); +int snd_hda_bus_new(struct snd_card *card, struct hda_bus **busp); int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, struct hda_codec **codecp); int snd_hda_codec_configure(struct hda_codec *codec); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index dfcb5e929f9f..31ff8b55f386 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1815,39 +1815,45 @@ static int get_jackpoll_interval(struct azx *chip) return j; } +static struct hda_bus_ops bus_ops = { + .command = azx_send_cmd, + .get_response = azx_get_response, + .attach_pcm = azx_attach_pcm_stream, + .bus_reset = azx_bus_reset, +#ifdef CONFIG_PM + .pm_notify = azx_power_notify, +#endif +#ifdef CONFIG_SND_HDA_DSP_LOADER + .load_dsp_prepare = azx_load_dsp_prepare, + .load_dsp_trigger = azx_load_dsp_trigger, + .load_dsp_cleanup = azx_load_dsp_cleanup, +#endif +}; + /* Codec initialization */ int azx_codec_create(struct azx *chip, const char *model, unsigned int max_slots, int *power_save_to) { - struct hda_bus_template bus_temp; + struct hda_bus *bus; int c, codecs, err; - memset(&bus_temp, 0, sizeof(bus_temp)); - bus_temp.private_data = chip; - bus_temp.modelname = model; - bus_temp.pci = chip->pci; - bus_temp.ops.command = azx_send_cmd; - bus_temp.ops.get_response = azx_get_response; - bus_temp.ops.attach_pcm = azx_attach_pcm_stream; - bus_temp.ops.bus_reset = azx_bus_reset; -#ifdef CONFIG_PM - bus_temp.power_save = power_save_to; - bus_temp.ops.pm_notify = azx_power_notify; -#endif -#ifdef CONFIG_SND_HDA_DSP_LOADER - bus_temp.ops.load_dsp_prepare = azx_load_dsp_prepare; - bus_temp.ops.load_dsp_trigger = azx_load_dsp_trigger; - bus_temp.ops.load_dsp_cleanup = azx_load_dsp_cleanup; -#endif - - err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus); + err = snd_hda_bus_new(chip->card, &bus); if (err < 0) return err; + chip->bus = bus; + bus->private_data = chip; + bus->pci = chip->pci; + bus->modelname = model; + bus->ops = bus_ops; +#ifdef CONFIG_PM + bus->power_save = power_save_to; +#endif + if (chip->driver_caps & AZX_DCAPS_RIRB_DELAY) { dev_dbg(chip->card->dev, "Enable delay in RIRB handling\n"); - chip->bus->needs_damn_long_delay = 1; + bus->needs_damn_long_delay = 1; } codecs = 0; @@ -1883,15 +1889,15 @@ int azx_codec_create(struct azx *chip, const char *model, */ if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); - chip->bus->sync_write = 1; - chip->bus->allow_bus_reset = 1; + bus->sync_write = 1; + bus->allow_bus_reset = 1; } /* Then create codec instances */ for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; - err = snd_hda_codec_new(chip->bus, c, &codec); + err = snd_hda_codec_new(bus, c, &codec); if (err < 0) continue; codec->jackpoll_interval = get_jackpoll_interval(chip); From 922c88a8368a61ee93653d4a2888a7f4ce263102 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2015 14:46:40 +0100 Subject: [PATCH 019/162] ALSA: hda - Embed struct hda_bus_unsolicited into struct hda_bus There is no big merit to handle hda_bus_unsolicited object individually, as it's tightly coupled with the hda_bus object itself. Embedding it makes the code simpler in the end. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 39 ++++----------------------------------- sound/pci/hda/hda_codec.h | 13 +++++++++++-- sound/pci/hda/hda_local.h | 17 ----------------- 3 files changed, 15 insertions(+), 54 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 215bf048c668..0734cb35887d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -762,10 +762,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) return 0; trace_hda_unsol_event(bus, res, res_ex); - unsol = bus->unsol; - if (!unsol) - return 0; - + unsol = &bus->unsol; wp = (unsol->wp + 1) % HDA_UNSOL_QUEUE_SIZE; unsol->wp = wp; @@ -784,9 +781,8 @@ EXPORT_SYMBOL_GPL(snd_hda_queue_unsol_event); */ static void process_unsol_events(struct work_struct *work) { - struct hda_bus_unsolicited *unsol = - container_of(work, struct hda_bus_unsolicited, work); - struct hda_bus *bus = unsol->bus; + struct hda_bus *bus = container_of(work, struct hda_bus, unsol.work); + struct hda_bus_unsolicited *unsol = &bus->unsol; struct hda_codec *codec; unsigned int rp, caddr, res; @@ -804,27 +800,6 @@ static void process_unsol_events(struct work_struct *work) } } -/* - * initialize unsolicited queue - */ -static int init_unsol_queue(struct hda_bus *bus) -{ - struct hda_bus_unsolicited *unsol; - - if (bus->unsol) /* already initialized */ - return 0; - - unsol = kzalloc(sizeof(*unsol), GFP_KERNEL); - if (!unsol) { - dev_err(bus->card->dev, "can't allocate unsolicited queue\n"); - return -ENOMEM; - } - INIT_WORK(&unsol->work, process_unsol_events); - unsol->bus = bus; - bus->unsol = unsol; - return 0; -} - /* * destructor */ @@ -836,7 +811,6 @@ static void snd_hda_bus_free(struct hda_bus *bus) WARN_ON(!list_empty(&bus->codec_list)); if (bus->workq) flush_workqueue(bus->workq); - kfree(bus->unsol); if (bus->ops.private_free) bus->ops.private_free(bus); if (bus->workq) @@ -888,6 +862,7 @@ int snd_hda_bus_new(struct snd_card *card, mutex_init(&bus->cmd_mutex); mutex_init(&bus->prepare_mutex); INIT_LIST_HEAD(&bus->codec_list); + INIT_WORK(&bus->unsol.work, process_unsol_events); snprintf(bus->workq_name, sizeof(bus->workq_name), "hd-audio%d", card->number); @@ -1689,12 +1664,6 @@ int snd_hda_codec_configure(struct hda_codec *codec) return err; } - if (codec->patch_ops.unsol_event) { - err = init_unsol_queue(codec->bus); - if (err < 0) - return err; - } - /* audio codec should override the mixer name */ if (codec->afg || !*codec->bus->card->mixername) snprintf(codec->bus->card->mixername, diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5a6594884069..96421a3b32cd 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -66,7 +66,6 @@ struct hda_beep; struct hda_codec; struct hda_pcm; struct hda_pcm_stream; -struct hda_bus_unsolicited; /* NID type */ typedef u16 hda_nid_t; @@ -101,6 +100,16 @@ struct hda_bus_ops { #endif }; +/* unsolicited event handler */ +#define HDA_UNSOL_QUEUE_SIZE 64 +struct hda_bus_unsolicited { + /* ring buffer */ + u32 queue[HDA_UNSOL_QUEUE_SIZE * 2]; + unsigned int rp, wp; + /* workqueue */ + struct work_struct work; +}; + /* * codec bus * @@ -126,7 +135,7 @@ struct hda_bus { struct mutex prepare_mutex; /* unsolicited event queue */ - struct hda_bus_unsolicited *unsol; + struct hda_bus_unsolicited unsol; char workq_name[16]; struct workqueue_struct *workq; /* common workqueue for codecs */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 62658f2f8c9f..49c08a7278c4 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -466,23 +466,6 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, const struct snd_hda_pin_quirk *pin_quirk, const struct hda_fixup *fixlist); - -/* - * unsolicited event handler - */ - -#define HDA_UNSOL_QUEUE_SIZE 64 - -struct hda_bus_unsolicited { - /* ring buffer */ - u32 queue[HDA_UNSOL_QUEUE_SIZE * 2]; - unsigned int rp, wp; - - /* workqueue */ - struct work_struct work; - struct hda_bus *bus; -}; - /* helper macros to retrieve pin default-config values */ #define get_defcfg_connect(cfg) \ ((cfg & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT) From 364aa716f43c991052cbb4fa05e3754bacccb95c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 16:51:17 +0100 Subject: [PATCH 020/162] ALSA: hda - Introduce azx_has_pm_runtime() macro For making the debugging of runtime PM easier, introduce azx_has_pm_runtime() and use it in all places checking the runtime pm driver capability. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 4 ++-- sound/pci/hda/hda_intel.c | 12 +++++------- sound/pci/hda/hda_priv.h | 3 +++ 3 files changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 31ff8b55f386..3589fc2165b0 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1681,7 +1681,7 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) int i; #ifdef CONFIG_PM - if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) + if (azx_has_pm_runtime(chip)) if (!pm_runtime_active(chip->card->dev)) return IRQ_NONE; #endif @@ -1784,7 +1784,7 @@ static void azx_power_notify(struct hda_bus *bus, bool power_up) { struct azx *chip = bus->private_data; - if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + if (!azx_has_pm_runtime(chip)) return; if (power_up) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 36d2f20db7a4..589883291f4e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -852,7 +852,7 @@ static int azx_runtime_suspend(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + if (!azx_has_pm_runtime(chip)) return 0; /* enable controller wake up event */ @@ -885,7 +885,7 @@ static int azx_runtime_resume(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + if (!azx_has_pm_runtime(chip)) return 0; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { @@ -928,8 +928,7 @@ static int azx_runtime_idle(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (!power_save_controller || - !(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) + if (!power_save_controller || !azx_has_pm_runtime(chip)) return -EBUSY; return 0; @@ -1071,8 +1070,7 @@ static int azx_free(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); int i; - if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) - && chip->running) + if (azx_has_pm_runtime(chip) && chip->running) pm_runtime_get_noresume(&pci->dev); azx_del_card_list(chip); @@ -1938,7 +1936,7 @@ static int azx_probe_continue(struct azx *chip) power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); - if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || hda->use_vga_switcheroo) + if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); out_free: diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h index daf458299753..a7b4a25c571c 100644 --- a/sound/pci/hda/hda_priv.h +++ b/sound/pci/hda/hda_priv.h @@ -403,4 +403,7 @@ struct azx { #define azx_sd_readb(chip, dev, reg) \ ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg)) +#define azx_has_pm_runtime(chip) \ + (!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)) + #endif /* __SOUND_HDA_PRIV_H */ From 89a93fea6182a71cedce9de1d901e4f379322cf3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 18:04:17 +0100 Subject: [PATCH 021/162] ALSA: hda - Fold hda_priv.h into hda_controller.h There is no big reason to keep them separately. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 1 - sound/pci/hda/hda_controller.h | 391 ++++++++++++++++++++++++++++++- sound/pci/hda/hda_i915.c | 2 +- sound/pci/hda/hda_intel.c | 1 - sound/pci/hda/hda_intel.h | 2 +- sound/pci/hda/hda_priv.h | 409 --------------------------------- sound/pci/hda/hda_tegra.c | 1 - 7 files changed, 392 insertions(+), 415 deletions(-) delete mode 100644 sound/pci/hda/hda_priv.h diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 3589fc2165b0..6b3254d479e1 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -30,7 +30,6 @@ #include #include #include -#include "hda_priv.h" #include "hda_controller.h" #define CREATE_TRACE_POINTS diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index c90d10fd4d8f..bd50b49cdd0d 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -15,10 +15,399 @@ #ifndef __SOUND_HDA_CONTROLLER_H #define __SOUND_HDA_CONTROLLER_H +#include +#include #include +#include #include #include "hda_codec.h" -#include "hda_priv.h" + +/* + * registers + */ +#define AZX_REG_GCAP 0x00 +#define AZX_GCAP_64OK (1 << 0) /* 64bit address support */ +#define AZX_GCAP_NSDO (3 << 1) /* # of serial data out signals */ +#define AZX_GCAP_BSS (31 << 3) /* # of bidirectional streams */ +#define AZX_GCAP_ISS (15 << 8) /* # of input streams */ +#define AZX_GCAP_OSS (15 << 12) /* # of output streams */ +#define AZX_REG_VMIN 0x02 +#define AZX_REG_VMAJ 0x03 +#define AZX_REG_OUTPAY 0x04 +#define AZX_REG_INPAY 0x06 +#define AZX_REG_GCTL 0x08 +#define AZX_GCTL_RESET (1 << 0) /* controller reset */ +#define AZX_GCTL_FCNTRL (1 << 1) /* flush control */ +#define AZX_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ +#define AZX_REG_WAKEEN 0x0c +#define AZX_REG_STATESTS 0x0e +#define AZX_REG_GSTS 0x10 +#define AZX_GSTS_FSTS (1 << 1) /* flush status */ +#define AZX_REG_INTCTL 0x20 +#define AZX_REG_INTSTS 0x24 +#define AZX_REG_WALLCLK 0x30 /* 24Mhz source */ +#define AZX_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */ +#define AZX_REG_SSYNC 0x38 +#define AZX_REG_CORBLBASE 0x40 +#define AZX_REG_CORBUBASE 0x44 +#define AZX_REG_CORBWP 0x48 +#define AZX_REG_CORBRP 0x4a +#define AZX_CORBRP_RST (1 << 15) /* read pointer reset */ +#define AZX_REG_CORBCTL 0x4c +#define AZX_CORBCTL_RUN (1 << 1) /* enable DMA */ +#define AZX_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */ +#define AZX_REG_CORBSTS 0x4d +#define AZX_CORBSTS_CMEI (1 << 0) /* memory error indication */ +#define AZX_REG_CORBSIZE 0x4e + +#define AZX_REG_RIRBLBASE 0x50 +#define AZX_REG_RIRBUBASE 0x54 +#define AZX_REG_RIRBWP 0x58 +#define AZX_RIRBWP_RST (1 << 15) /* write pointer reset */ +#define AZX_REG_RINTCNT 0x5a +#define AZX_REG_RIRBCTL 0x5c +#define AZX_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */ +#define AZX_RBCTL_DMA_EN (1 << 1) /* enable DMA */ +#define AZX_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */ +#define AZX_REG_RIRBSTS 0x5d +#define AZX_RBSTS_IRQ (1 << 0) /* response irq */ +#define AZX_RBSTS_OVERRUN (1 << 2) /* overrun irq */ +#define AZX_REG_RIRBSIZE 0x5e + +#define AZX_REG_IC 0x60 +#define AZX_REG_IR 0x64 +#define AZX_REG_IRS 0x68 +#define AZX_IRS_VALID (1<<1) +#define AZX_IRS_BUSY (1<<0) + +#define AZX_REG_DPLBASE 0x70 +#define AZX_REG_DPUBASE 0x74 +#define AZX_DPLBASE_ENABLE 0x1 /* Enable position buffer */ + +/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ +enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; + +/* stream register offsets from stream base */ +#define AZX_REG_SD_CTL 0x00 +#define AZX_REG_SD_STS 0x03 +#define AZX_REG_SD_LPIB 0x04 +#define AZX_REG_SD_CBL 0x08 +#define AZX_REG_SD_LVI 0x0c +#define AZX_REG_SD_FIFOW 0x0e +#define AZX_REG_SD_FIFOSIZE 0x10 +#define AZX_REG_SD_FORMAT 0x12 +#define AZX_REG_SD_BDLPL 0x18 +#define AZX_REG_SD_BDLPU 0x1c + +/* PCI space */ +#define AZX_PCIREG_TCSEL 0x44 + +/* + * other constants + */ + +/* max number of fragments - we may use more if allocating more pages for BDL */ +#define BDL_SIZE 4096 +#define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16) +#define AZX_MAX_FRAG 32 +/* max buffer size - no h/w limit, you can increase as you like */ +#define AZX_MAX_BUF_SIZE (1024*1024*1024) + +/* RIRB int mask: overrun[2], response[0] */ +#define RIRB_INT_RESPONSE 0x01 +#define RIRB_INT_OVERRUN 0x04 +#define RIRB_INT_MASK 0x05 + +/* STATESTS int mask: S3,SD2,SD1,SD0 */ +#define AZX_MAX_CODECS 8 +#define AZX_DEFAULT_CODECS 4 +#define STATESTS_INT_MASK ((1 << AZX_MAX_CODECS) - 1) + +/* SD_CTL bits */ +#define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */ +#define SD_CTL_DMA_START 0x02 /* stream DMA start bit */ +#define SD_CTL_STRIPE (3 << 16) /* stripe control */ +#define SD_CTL_TRAFFIC_PRIO (1 << 18) /* traffic priority */ +#define SD_CTL_DIR (1 << 19) /* bi-directional stream */ +#define SD_CTL_STREAM_TAG_MASK (0xf << 20) +#define SD_CTL_STREAM_TAG_SHIFT 20 + +/* SD_CTL and SD_STS */ +#define SD_INT_DESC_ERR 0x10 /* descriptor error interrupt */ +#define SD_INT_FIFO_ERR 0x08 /* FIFO error interrupt */ +#define SD_INT_COMPLETE 0x04 /* completion interrupt */ +#define SD_INT_MASK (SD_INT_DESC_ERR|SD_INT_FIFO_ERR|\ + SD_INT_COMPLETE) + +/* SD_STS */ +#define SD_STS_FIFO_READY 0x20 /* FIFO ready */ + +/* INTCTL and INTSTS */ +#define AZX_INT_ALL_STREAM 0xff /* all stream interrupts */ +#define AZX_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ +#define AZX_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ + +/* below are so far hardcoded - should read registers in future */ +#define AZX_MAX_CORB_ENTRIES 256 +#define AZX_MAX_RIRB_ENTRIES 256 + +/* driver quirks (capabilities) */ +/* bits 0-7 are used for indicating driver type */ +#define AZX_DCAPS_NO_TCSEL (1 << 8) /* No Intel TCSEL bit */ +#define AZX_DCAPS_NO_MSI (1 << 9) /* No MSI support */ +#define AZX_DCAPS_SNOOP_MASK (3 << 10) /* snoop type mask */ +#define AZX_DCAPS_SNOOP_OFF (1 << 12) /* snoop default off */ +#define AZX_DCAPS_RIRB_DELAY (1 << 13) /* Long delay in read loop */ +#define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14) /* Put a delay before read */ +#define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ +#define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ +#define AZX_DCAPS_POSFIX_VIA (1 << 17) /* Use VIACOMBO as default */ +#define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ +#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ +#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ +#define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21) /* no buffer size alignment */ +/* 22 unused */ +#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ +#define AZX_DCAPS_REVERSE_ASSIGN (1 << 24) /* Assign devices in reverse order */ +#define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ +#define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ +#define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ +#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ +#define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ +#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ + +enum { + AZX_SNOOP_TYPE_NONE, + AZX_SNOOP_TYPE_SCH, + AZX_SNOOP_TYPE_ATI, + AZX_SNOOP_TYPE_NVIDIA, +}; + +/* HD Audio class code */ +#define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403 + +struct azx_dev { + struct snd_dma_buffer bdl; /* BDL buffer */ + u32 *posbuf; /* position buffer pointer */ + + unsigned int bufsize; /* size of the play buffer in bytes */ + unsigned int period_bytes; /* size of the period in bytes */ + unsigned int frags; /* number for period in the play buffer */ + unsigned int fifo_size; /* FIFO size */ + unsigned long start_wallclk; /* start + minimum wallclk */ + unsigned long period_wallclk; /* wallclk for period */ + + void __iomem *sd_addr; /* stream descriptor pointer */ + + u32 sd_int_sta_mask; /* stream int status mask */ + + /* pcm support */ + struct snd_pcm_substream *substream; /* assigned substream, + * set in PCM open + */ + unsigned int format_val; /* format value to be set in the + * controller and the codec + */ + unsigned char stream_tag; /* assigned stream */ + unsigned char index; /* stream index */ + int assigned_key; /* last device# key assigned to */ + + unsigned int opened:1; + unsigned int running:1; + unsigned int irq_pending:1; + unsigned int prepared:1; + unsigned int locked:1; + /* + * For VIA: + * A flag to ensure DMA position is 0 + * when link position is not greater than FIFO size + */ + unsigned int insufficient:1; + unsigned int wc_marked:1; + unsigned int no_period_wakeup:1; + + struct timecounter azx_tc; + struct cyclecounter azx_cc; + + int delay_negative_threshold; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + /* Allows dsp load to have sole access to the playback stream. */ + struct mutex dsp_mutex; +#endif +}; + +/* CORB/RIRB */ +struct azx_rb { + u32 *buf; /* CORB/RIRB buffer + * Each CORB entry is 4byte, RIRB is 8byte + */ + dma_addr_t addr; /* physical address of CORB/RIRB buffer */ + /* for RIRB */ + unsigned short rp, wp; /* read/write pointers */ + int cmds[AZX_MAX_CODECS]; /* number of pending requests */ + u32 res[AZX_MAX_CODECS]; /* last read value */ +}; + +struct azx; + +/* Functions to read/write to hda registers. */ +struct hda_controller_ops { + /* Register Access */ + void (*reg_writel)(u32 value, u32 __iomem *addr); + u32 (*reg_readl)(u32 __iomem *addr); + void (*reg_writew)(u16 value, u16 __iomem *addr); + u16 (*reg_readw)(u16 __iomem *addr); + void (*reg_writeb)(u8 value, u8 __iomem *addr); + u8 (*reg_readb)(u8 __iomem *addr); + /* Disable msi if supported, PCI only */ + int (*disable_msi_reset_irq)(struct azx *); + /* Allocation ops */ + int (*dma_alloc_pages)(struct azx *chip, + int type, + size_t size, + struct snd_dma_buffer *buf); + void (*dma_free_pages)(struct azx *chip, struct snd_dma_buffer *buf); + int (*substream_alloc_pages)(struct azx *chip, + struct snd_pcm_substream *substream, + size_t size); + int (*substream_free_pages)(struct azx *chip, + struct snd_pcm_substream *substream); + void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream, + struct vm_area_struct *area); + /* Check if current position is acceptable */ + int (*position_check)(struct azx *chip, struct azx_dev *azx_dev); +}; + +struct azx_pcm { + struct azx *chip; + struct snd_pcm *pcm; + struct hda_codec *codec; + struct hda_pcm_stream *hinfo[2]; + struct list_head list; +}; + +typedef unsigned int (*azx_get_pos_callback_t)(struct azx *, struct azx_dev *); +typedef int (*azx_get_delay_callback_t)(struct azx *, struct azx_dev *, unsigned int pos); + +struct azx { + struct snd_card *card; + struct pci_dev *pci; + int dev_index; + + /* chip type specific */ + int driver_type; + unsigned int driver_caps; + int playback_streams; + int playback_index_offset; + int capture_streams; + int capture_index_offset; + int num_streams; + const int *jackpoll_ms; /* per-card jack poll interval */ + + /* Register interaction. */ + const struct hda_controller_ops *ops; + + /* position adjustment callbacks */ + azx_get_pos_callback_t get_position[2]; + azx_get_delay_callback_t get_delay[2]; + + /* pci resources */ + unsigned long addr; + void __iomem *remap_addr; + int irq; + + /* locks */ + spinlock_t reg_lock; + struct mutex open_mutex; /* Prevents concurrent open/close operations */ + + /* streams (x num_streams) */ + struct azx_dev *azx_dev; + + /* PCM */ + struct list_head pcm_list; /* azx_pcm list */ + + /* HD codec */ + unsigned short codec_mask; + int codec_probe_mask; /* copied from probe_mask option */ + struct hda_bus *bus; + unsigned int beep_mode; + + /* CORB/RIRB */ + struct azx_rb corb; + struct azx_rb rirb; + + /* CORB/RIRB and position buffers */ + struct snd_dma_buffer rb; + struct snd_dma_buffer posbuf; + +#ifdef CONFIG_SND_HDA_PATCH_LOADER + const struct firmware *fw; +#endif + + /* flags */ + const int *bdl_pos_adj; + int poll_count; + unsigned int running:1; + unsigned int initialized:1; + unsigned int single_cmd:1; + unsigned int polling_mode:1; + unsigned int msi:1; + unsigned int probing:1; /* codec probing phase */ + unsigned int snoop:1; + unsigned int align_buffer_size:1; + unsigned int region_requested:1; + unsigned int disabled:1; /* disabled by VGA-switcher */ + + /* for debugging */ + unsigned int last_cmd[AZX_MAX_CODECS]; + + /* reboot notifier (for mysterious hangup problem at power-down) */ + struct notifier_block reboot_notifier; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + struct azx_dev saved_azx_dev; +#endif +}; + +#ifdef CONFIG_X86 +#define azx_snoop(chip) ((chip)->snoop) +#else +#define azx_snoop(chip) true +#endif + +/* + * macros for easy use + */ + +#define azx_writel(chip, reg, value) \ + ((chip)->ops->reg_writel(value, (chip)->remap_addr + AZX_REG_##reg)) +#define azx_readl(chip, reg) \ + ((chip)->ops->reg_readl((chip)->remap_addr + AZX_REG_##reg)) +#define azx_writew(chip, reg, value) \ + ((chip)->ops->reg_writew(value, (chip)->remap_addr + AZX_REG_##reg)) +#define azx_readw(chip, reg) \ + ((chip)->ops->reg_readw((chip)->remap_addr + AZX_REG_##reg)) +#define azx_writeb(chip, reg, value) \ + ((chip)->ops->reg_writeb(value, (chip)->remap_addr + AZX_REG_##reg)) +#define azx_readb(chip, reg) \ + ((chip)->ops->reg_readb((chip)->remap_addr + AZX_REG_##reg)) + +#define azx_sd_writel(chip, dev, reg, value) \ + ((chip)->ops->reg_writel(value, (dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_readl(chip, dev, reg) \ + ((chip)->ops->reg_readl((dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_writew(chip, dev, reg, value) \ + ((chip)->ops->reg_writew(value, (dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_readw(chip, dev, reg) \ + ((chip)->ops->reg_readw((dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_writeb(chip, dev, reg, value) \ + ((chip)->ops->reg_writeb(value, (dev)->sd_addr + AZX_REG_##reg)) +#define azx_sd_readb(chip, dev, reg) \ + ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg)) + +#define azx_has_pm_runtime(chip) \ + (!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)) /* PCM setup */ static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream) diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c index 714894527e06..52a85d87c23c 100644 --- a/sound/pci/hda/hda_i915.c +++ b/sound/pci/hda/hda_i915.c @@ -22,7 +22,7 @@ #include #include #include -#include "hda_priv.h" +#include "hda_controller.h" #include "hda_intel.h" /* Intel HSW/BDW display HDA controller Extended Mode registers. diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 589883291f4e..ced44a75f8ea 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -62,7 +62,6 @@ #include #include "hda_codec.h" #include "hda_controller.h" -#include "hda_priv.h" #include "hda_intel.h" /* position fix mode */ diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 348611835476..d5231f7216a7 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -17,7 +17,7 @@ #define __SOUND_HDA_INTEL_H #include -#include "hda_priv.h" +#include "hda_controller.h" struct hda_intel { struct azx chip; diff --git a/sound/pci/hda/hda_priv.h b/sound/pci/hda/hda_priv.h deleted file mode 100644 index a7b4a25c571c..000000000000 --- a/sound/pci/hda/hda_priv.h +++ /dev/null @@ -1,409 +0,0 @@ -/* - * Common defines for the alsa driver code base for HD Audio. - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the Free - * Software Foundation; either version 2 of the License, or (at your option) - * any later version. - * - * This program is distributed in the hope that it will be useful, but WITHOUT - * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or - * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for - * more details. - */ - -#ifndef __SOUND_HDA_PRIV_H -#define __SOUND_HDA_PRIV_H - -#include -#include -#include - -/* - * registers - */ -#define AZX_REG_GCAP 0x00 -#define AZX_GCAP_64OK (1 << 0) /* 64bit address support */ -#define AZX_GCAP_NSDO (3 << 1) /* # of serial data out signals */ -#define AZX_GCAP_BSS (31 << 3) /* # of bidirectional streams */ -#define AZX_GCAP_ISS (15 << 8) /* # of input streams */ -#define AZX_GCAP_OSS (15 << 12) /* # of output streams */ -#define AZX_REG_VMIN 0x02 -#define AZX_REG_VMAJ 0x03 -#define AZX_REG_OUTPAY 0x04 -#define AZX_REG_INPAY 0x06 -#define AZX_REG_GCTL 0x08 -#define AZX_GCTL_RESET (1 << 0) /* controller reset */ -#define AZX_GCTL_FCNTRL (1 << 1) /* flush control */ -#define AZX_GCTL_UNSOL (1 << 8) /* accept unsol. response enable */ -#define AZX_REG_WAKEEN 0x0c -#define AZX_REG_STATESTS 0x0e -#define AZX_REG_GSTS 0x10 -#define AZX_GSTS_FSTS (1 << 1) /* flush status */ -#define AZX_REG_INTCTL 0x20 -#define AZX_REG_INTSTS 0x24 -#define AZX_REG_WALLCLK 0x30 /* 24Mhz source */ -#define AZX_REG_OLD_SSYNC 0x34 /* SSYNC for old ICH */ -#define AZX_REG_SSYNC 0x38 -#define AZX_REG_CORBLBASE 0x40 -#define AZX_REG_CORBUBASE 0x44 -#define AZX_REG_CORBWP 0x48 -#define AZX_REG_CORBRP 0x4a -#define AZX_CORBRP_RST (1 << 15) /* read pointer reset */ -#define AZX_REG_CORBCTL 0x4c -#define AZX_CORBCTL_RUN (1 << 1) /* enable DMA */ -#define AZX_CORBCTL_CMEIE (1 << 0) /* enable memory error irq */ -#define AZX_REG_CORBSTS 0x4d -#define AZX_CORBSTS_CMEI (1 << 0) /* memory error indication */ -#define AZX_REG_CORBSIZE 0x4e - -#define AZX_REG_RIRBLBASE 0x50 -#define AZX_REG_RIRBUBASE 0x54 -#define AZX_REG_RIRBWP 0x58 -#define AZX_RIRBWP_RST (1 << 15) /* write pointer reset */ -#define AZX_REG_RINTCNT 0x5a -#define AZX_REG_RIRBCTL 0x5c -#define AZX_RBCTL_IRQ_EN (1 << 0) /* enable IRQ */ -#define AZX_RBCTL_DMA_EN (1 << 1) /* enable DMA */ -#define AZX_RBCTL_OVERRUN_EN (1 << 2) /* enable overrun irq */ -#define AZX_REG_RIRBSTS 0x5d -#define AZX_RBSTS_IRQ (1 << 0) /* response irq */ -#define AZX_RBSTS_OVERRUN (1 << 2) /* overrun irq */ -#define AZX_REG_RIRBSIZE 0x5e - -#define AZX_REG_IC 0x60 -#define AZX_REG_IR 0x64 -#define AZX_REG_IRS 0x68 -#define AZX_IRS_VALID (1<<1) -#define AZX_IRS_BUSY (1<<0) - -#define AZX_REG_DPLBASE 0x70 -#define AZX_REG_DPUBASE 0x74 -#define AZX_DPLBASE_ENABLE 0x1 /* Enable position buffer */ - -/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */ -enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; - -/* stream register offsets from stream base */ -#define AZX_REG_SD_CTL 0x00 -#define AZX_REG_SD_STS 0x03 -#define AZX_REG_SD_LPIB 0x04 -#define AZX_REG_SD_CBL 0x08 -#define AZX_REG_SD_LVI 0x0c -#define AZX_REG_SD_FIFOW 0x0e -#define AZX_REG_SD_FIFOSIZE 0x10 -#define AZX_REG_SD_FORMAT 0x12 -#define AZX_REG_SD_BDLPL 0x18 -#define AZX_REG_SD_BDLPU 0x1c - -/* PCI space */ -#define AZX_PCIREG_TCSEL 0x44 - -/* - * other constants - */ - -/* max number of fragments - we may use more if allocating more pages for BDL */ -#define BDL_SIZE 4096 -#define AZX_MAX_BDL_ENTRIES (BDL_SIZE / 16) -#define AZX_MAX_FRAG 32 -/* max buffer size - no h/w limit, you can increase as you like */ -#define AZX_MAX_BUF_SIZE (1024*1024*1024) - -/* RIRB int mask: overrun[2], response[0] */ -#define RIRB_INT_RESPONSE 0x01 -#define RIRB_INT_OVERRUN 0x04 -#define RIRB_INT_MASK 0x05 - -/* STATESTS int mask: S3,SD2,SD1,SD0 */ -#define AZX_MAX_CODECS 8 -#define AZX_DEFAULT_CODECS 4 -#define STATESTS_INT_MASK ((1 << AZX_MAX_CODECS) - 1) - -/* SD_CTL bits */ -#define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */ -#define SD_CTL_DMA_START 0x02 /* stream DMA start bit */ -#define SD_CTL_STRIPE (3 << 16) /* stripe control */ -#define SD_CTL_TRAFFIC_PRIO (1 << 18) /* traffic priority */ -#define SD_CTL_DIR (1 << 19) /* bi-directional stream */ -#define SD_CTL_STREAM_TAG_MASK (0xf << 20) -#define SD_CTL_STREAM_TAG_SHIFT 20 - -/* SD_CTL and SD_STS */ -#define SD_INT_DESC_ERR 0x10 /* descriptor error interrupt */ -#define SD_INT_FIFO_ERR 0x08 /* FIFO error interrupt */ -#define SD_INT_COMPLETE 0x04 /* completion interrupt */ -#define SD_INT_MASK (SD_INT_DESC_ERR|SD_INT_FIFO_ERR|\ - SD_INT_COMPLETE) - -/* SD_STS */ -#define SD_STS_FIFO_READY 0x20 /* FIFO ready */ - -/* INTCTL and INTSTS */ -#define AZX_INT_ALL_STREAM 0xff /* all stream interrupts */ -#define AZX_INT_CTRL_EN 0x40000000 /* controller interrupt enable bit */ -#define AZX_INT_GLOBAL_EN 0x80000000 /* global interrupt enable bit */ - -/* below are so far hardcoded - should read registers in future */ -#define AZX_MAX_CORB_ENTRIES 256 -#define AZX_MAX_RIRB_ENTRIES 256 - -/* driver quirks (capabilities) */ -/* bits 0-7 are used for indicating driver type */ -#define AZX_DCAPS_NO_TCSEL (1 << 8) /* No Intel TCSEL bit */ -#define AZX_DCAPS_NO_MSI (1 << 9) /* No MSI support */ -#define AZX_DCAPS_SNOOP_MASK (3 << 10) /* snoop type mask */ -#define AZX_DCAPS_SNOOP_OFF (1 << 12) /* snoop default off */ -#define AZX_DCAPS_RIRB_DELAY (1 << 13) /* Long delay in read loop */ -#define AZX_DCAPS_RIRB_PRE_DELAY (1 << 14) /* Put a delay before read */ -#define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ -#define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ -#define AZX_DCAPS_POSFIX_VIA (1 << 17) /* Use VIACOMBO as default */ -#define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ -#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ -#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ -#define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21) /* no buffer size alignment */ -/* 22 unused */ -#define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23) /* BDLE in 4k boundary */ -#define AZX_DCAPS_REVERSE_ASSIGN (1 << 24) /* Assign devices in reverse order */ -#define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ -#define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -#define AZX_DCAPS_I915_POWERWELL (1 << 27) /* HSW i915 powerwell support */ -#define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ -#define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ -#define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ - -enum { - AZX_SNOOP_TYPE_NONE , - AZX_SNOOP_TYPE_SCH, - AZX_SNOOP_TYPE_ATI, - AZX_SNOOP_TYPE_NVIDIA, -}; - -/* HD Audio class code */ -#define PCI_CLASS_MULTIMEDIA_HD_AUDIO 0x0403 - -struct azx_dev { - struct snd_dma_buffer bdl; /* BDL buffer */ - u32 *posbuf; /* position buffer pointer */ - - unsigned int bufsize; /* size of the play buffer in bytes */ - unsigned int period_bytes; /* size of the period in bytes */ - unsigned int frags; /* number for period in the play buffer */ - unsigned int fifo_size; /* FIFO size */ - unsigned long start_wallclk; /* start + minimum wallclk */ - unsigned long period_wallclk; /* wallclk for period */ - - void __iomem *sd_addr; /* stream descriptor pointer */ - - u32 sd_int_sta_mask; /* stream int status mask */ - - /* pcm support */ - struct snd_pcm_substream *substream; /* assigned substream, - * set in PCM open - */ - unsigned int format_val; /* format value to be set in the - * controller and the codec - */ - unsigned char stream_tag; /* assigned stream */ - unsigned char index; /* stream index */ - int assigned_key; /* last device# key assigned to */ - - unsigned int opened:1; - unsigned int running:1; - unsigned int irq_pending:1; - unsigned int prepared:1; - unsigned int locked:1; - /* - * For VIA: - * A flag to ensure DMA position is 0 - * when link position is not greater than FIFO size - */ - unsigned int insufficient:1; - unsigned int wc_marked:1; - unsigned int no_period_wakeup:1; - - struct timecounter azx_tc; - struct cyclecounter azx_cc; - - int delay_negative_threshold; - -#ifdef CONFIG_SND_HDA_DSP_LOADER - /* Allows dsp load to have sole access to the playback stream. */ - struct mutex dsp_mutex; -#endif -}; - -/* CORB/RIRB */ -struct azx_rb { - u32 *buf; /* CORB/RIRB buffer - * Each CORB entry is 4byte, RIRB is 8byte - */ - dma_addr_t addr; /* physical address of CORB/RIRB buffer */ - /* for RIRB */ - unsigned short rp, wp; /* read/write pointers */ - int cmds[AZX_MAX_CODECS]; /* number of pending requests */ - u32 res[AZX_MAX_CODECS]; /* last read value */ -}; - -struct azx; - -/* Functions to read/write to hda registers. */ -struct hda_controller_ops { - /* Register Access */ - void (*reg_writel)(u32 value, u32 __iomem *addr); - u32 (*reg_readl)(u32 __iomem *addr); - void (*reg_writew)(u16 value, u16 __iomem *addr); - u16 (*reg_readw)(u16 __iomem *addr); - void (*reg_writeb)(u8 value, u8 __iomem *addr); - u8 (*reg_readb)(u8 __iomem *addr); - /* Disable msi if supported, PCI only */ - int (*disable_msi_reset_irq)(struct azx *); - /* Allocation ops */ - int (*dma_alloc_pages)(struct azx *chip, - int type, - size_t size, - struct snd_dma_buffer *buf); - void (*dma_free_pages)(struct azx *chip, struct snd_dma_buffer *buf); - int (*substream_alloc_pages)(struct azx *chip, - struct snd_pcm_substream *substream, - size_t size); - int (*substream_free_pages)(struct azx *chip, - struct snd_pcm_substream *substream); - void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream, - struct vm_area_struct *area); - /* Check if current position is acceptable */ - int (*position_check)(struct azx *chip, struct azx_dev *azx_dev); -}; - -struct azx_pcm { - struct azx *chip; - struct snd_pcm *pcm; - struct hda_codec *codec; - struct hda_pcm_stream *hinfo[2]; - struct list_head list; -}; - -typedef unsigned int (*azx_get_pos_callback_t)(struct azx *, struct azx_dev *); -typedef int (*azx_get_delay_callback_t)(struct azx *, struct azx_dev *, unsigned int pos); - -struct azx { - struct snd_card *card; - struct pci_dev *pci; - int dev_index; - - /* chip type specific */ - int driver_type; - unsigned int driver_caps; - int playback_streams; - int playback_index_offset; - int capture_streams; - int capture_index_offset; - int num_streams; - const int *jackpoll_ms; /* per-card jack poll interval */ - - /* Register interaction. */ - const struct hda_controller_ops *ops; - - /* position adjustment callbacks */ - azx_get_pos_callback_t get_position[2]; - azx_get_delay_callback_t get_delay[2]; - - /* pci resources */ - unsigned long addr; - void __iomem *remap_addr; - int irq; - - /* locks */ - spinlock_t reg_lock; - struct mutex open_mutex; /* Prevents concurrent open/close operations */ - - /* streams (x num_streams) */ - struct azx_dev *azx_dev; - - /* PCM */ - struct list_head pcm_list; /* azx_pcm list */ - - /* HD codec */ - unsigned short codec_mask; - int codec_probe_mask; /* copied from probe_mask option */ - struct hda_bus *bus; - unsigned int beep_mode; - - /* CORB/RIRB */ - struct azx_rb corb; - struct azx_rb rirb; - - /* CORB/RIRB and position buffers */ - struct snd_dma_buffer rb; - struct snd_dma_buffer posbuf; - -#ifdef CONFIG_SND_HDA_PATCH_LOADER - const struct firmware *fw; -#endif - - /* flags */ - const int *bdl_pos_adj; - int poll_count; - unsigned int running:1; - unsigned int initialized:1; - unsigned int single_cmd:1; - unsigned int polling_mode:1; - unsigned int msi:1; - unsigned int probing:1; /* codec probing phase */ - unsigned int snoop:1; - unsigned int align_buffer_size:1; - unsigned int region_requested:1; - unsigned int disabled:1; /* disabled by VGA-switcher */ - - /* for debugging */ - unsigned int last_cmd[AZX_MAX_CODECS]; - - /* reboot notifier (for mysterious hangup problem at power-down) */ - struct notifier_block reboot_notifier; - -#ifdef CONFIG_SND_HDA_DSP_LOADER - struct azx_dev saved_azx_dev; -#endif -}; - -#ifdef CONFIG_X86 -#define azx_snoop(chip) ((chip)->snoop) -#else -#define azx_snoop(chip) true -#endif - -/* - * macros for easy use - */ - -#define azx_writel(chip, reg, value) \ - ((chip)->ops->reg_writel(value, (chip)->remap_addr + AZX_REG_##reg)) -#define azx_readl(chip, reg) \ - ((chip)->ops->reg_readl((chip)->remap_addr + AZX_REG_##reg)) -#define azx_writew(chip, reg, value) \ - ((chip)->ops->reg_writew(value, (chip)->remap_addr + AZX_REG_##reg)) -#define azx_readw(chip, reg) \ - ((chip)->ops->reg_readw((chip)->remap_addr + AZX_REG_##reg)) -#define azx_writeb(chip, reg, value) \ - ((chip)->ops->reg_writeb(value, (chip)->remap_addr + AZX_REG_##reg)) -#define azx_readb(chip, reg) \ - ((chip)->ops->reg_readb((chip)->remap_addr + AZX_REG_##reg)) - -#define azx_sd_writel(chip, dev, reg, value) \ - ((chip)->ops->reg_writel(value, (dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_readl(chip, dev, reg) \ - ((chip)->ops->reg_readl((dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_writew(chip, dev, reg, value) \ - ((chip)->ops->reg_writew(value, (dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_readw(chip, dev, reg) \ - ((chip)->ops->reg_readw((dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_writeb(chip, dev, reg, value) \ - ((chip)->ops->reg_writeb(value, (dev)->sd_addr + AZX_REG_##reg)) -#define azx_sd_readb(chip, dev, reg) \ - ((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg)) - -#define azx_has_pm_runtime(chip) \ - (!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)) - -#endif /* __SOUND_HDA_PRIV_H */ diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 375e94f4cf52..7d0d04480f48 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -37,7 +37,6 @@ #include "hda_codec.h" #include "hda_controller.h" -#include "hda_priv.h" /* Defines for Nvidia Tegra HDA support */ #define HDA_BAR0 0x8000 From b8f28d53641f13902790904ab15028ff8ecd0882 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 18:06:45 +0100 Subject: [PATCH 022/162] ALSA: hda - Drop azx_mixer_create() It's just an indirection, so let the caller directly calling snd_hda_build_controls(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 7 ------- sound/pci/hda/hda_controller.h | 1 - sound/pci/hda/hda_intel.c | 2 +- sound/pci/hda/hda_tegra.c | 2 +- 4 files changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 6b3254d479e1..6fab39133051 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1923,13 +1923,6 @@ int azx_codec_configure(struct azx *chip) } EXPORT_SYMBOL_GPL(azx_codec_configure); -/* mixer creation - all stuff is implemented in hda module */ -int azx_mixer_create(struct azx *chip) -{ - return snd_hda_build_controls(chip->bus); -} -EXPORT_SYMBOL_GPL(azx_mixer_create); - static bool is_input_stream(struct azx *chip, unsigned char index) { diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index bd50b49cdd0d..79808e1a79ac 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -436,7 +436,6 @@ int azx_codec_create(struct azx *chip, const char *model, unsigned int max_slots, int *power_save_to); int azx_codec_configure(struct azx *chip); -int azx_mixer_create(struct azx *chip); int azx_init_stream(struct azx *chip); void azx_notifier_register(struct azx *chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ced44a75f8ea..b0a05691abe7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1923,7 +1923,7 @@ static int azx_probe_continue(struct azx *chip) goto out_free; /* create mixer controls */ - err = azx_mixer_create(chip); + err = snd_hda_build_controls(chip->bus); if (err < 0) goto out_free; diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 7d0d04480f48..f305c2a99206 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -516,7 +516,7 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; /* create mixer controls */ - err = azx_mixer_create(chip); + err = snd_hda_build_controls(chip->bus); if (err < 0) goto out_free; From 96d2bd6e3cdf57926f80605d6e28051bb6b24eb3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 18:12:22 +0100 Subject: [PATCH 023/162] ALSA: hda - Split azx_codec_create() to two phases azx_create_codec() function does actually two things: create a bus and probe codecs. For the future work, split this to two logical functions, azx_bus_create() and azx_probe_codecs(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 40 ++++++++++++++++++++-------------- sound/pci/hda/hda_controller.h | 5 ++--- sound/pci/hda/hda_intel.c | 10 +++++---- sound/pci/hda/hda_tegra.c | 6 ++++- 4 files changed, 37 insertions(+), 24 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 6fab39133051..2a674525e56f 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1829,13 +1829,11 @@ static struct hda_bus_ops bus_ops = { #endif }; -/* Codec initialization */ -int azx_codec_create(struct azx *chip, const char *model, - unsigned int max_slots, - int *power_save_to) +/* HD-audio bus initialization */ +int azx_bus_create(struct azx *chip, const char *model, int *power_save_to) { struct hda_bus *bus; - int c, codecs, err; + int err; err = snd_hda_bus_new(chip->card, &bus); if (err < 0) @@ -1855,6 +1853,26 @@ int azx_codec_create(struct azx *chip, const char *model, bus->needs_damn_long_delay = 1; } + /* AMD chipsets often cause the communication stalls upon certain + * sequence like the pin-detection. It seems that forcing the synced + * access works around the stall. Grrr... + */ + if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { + dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); + bus->sync_write = 1; + bus->allow_bus_reset = 1; + } + + return 0; +} +EXPORT_SYMBOL_GPL(azx_bus_create); + +/* Probe codecs */ +int azx_probe_codecs(struct azx *chip, unsigned int max_slots) +{ + struct hda_bus *bus = chip->bus; + int c, codecs, err; + codecs = 0; if (!max_slots) max_slots = AZX_DEFAULT_CODECS; @@ -1882,16 +1900,6 @@ int azx_codec_create(struct azx *chip, const char *model, } } - /* AMD chipsets often cause the communication stalls upon certain - * sequence like the pin-detection. It seems that forcing the synced - * access works around the stall. Grrr... - */ - if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { - dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); - bus->sync_write = 1; - bus->allow_bus_reset = 1; - } - /* Then create codec instances */ for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { @@ -1910,7 +1918,7 @@ int azx_codec_create(struct azx *chip, const char *model, } return 0; } -EXPORT_SYMBOL_GPL(azx_codec_create); +EXPORT_SYMBOL_GPL(azx_probe_codecs); /* configure each codec instance */ int azx_codec_configure(struct azx *chip) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 79808e1a79ac..0d09aa6b49ac 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -432,9 +432,8 @@ void azx_enter_link_reset(struct azx *chip); irqreturn_t azx_interrupt(int irq, void *dev_id); /* Codec interface */ -int azx_codec_create(struct azx *chip, const char *model, - unsigned int max_slots, - int *power_save_to); +int azx_bus_create(struct azx *chip, const char *model, int *power_save_to); +int azx_probe_codecs(struct azx *chip, unsigned int max_slots); int azx_codec_configure(struct azx *chip); int azx_init_stream(struct azx *chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b0a05691abe7..5e00cc4a722f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1893,12 +1893,14 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_codec_create(chip, model[dev], - azx_max_codecs[chip->driver_type], - power_save_addr); - + err = azx_bus_create(chip, model[dev], power_save_addr); if (err < 0) goto out_free; + + err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); + if (err < 0) + goto out_free; + #ifdef CONFIG_SND_HDA_PATCH_LOADER if (chip->fw) { err = snd_hda_load_patch(chip->bus, chip->fw->size, diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index f305c2a99206..1bd7a9e04046 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -502,7 +502,11 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; /* create codec instances */ - err = azx_codec_create(chip, NULL, 0, &power_save); + err = azx_bus_create(chip, NULL, &power_save); + if (err < 0) + goto out_free; + + err = azx_probe_codecs(chip, 0); if (err < 0) goto out_free; From 781c7b9615dc1441d6743124086eb3da14bf46e9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 10:26:25 +0100 Subject: [PATCH 024/162] ALSA: hda - Avoid unnecessary power-up at mixer amp changes When the mixer amp is touched by control elements, we don't have to power up always; if the codec was suspended at the time, we can just update the amp cache and it's reflected to the hardware upon resume. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 33 ++++++++++++++++----------------- 1 file changed, 16 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0734cb35887d..5e755eb6f19a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2148,11 +2148,10 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_read); static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val, - bool init_only) + bool init_only, bool cache_only) { struct hda_amp_info *info; unsigned int caps; - unsigned int cache_only; if (snd_BUG_ON(mask & ~0xff)) mask &= 0xff; @@ -2170,7 +2169,7 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, return 0; } info->vol[ch] = val; - cache_only = info->head.dirty = codec->cached_write; + info->head.dirty |= cache_only; caps = info->amp_caps; mutex_unlock(&codec->hash_mutex); if (!cache_only) @@ -2194,7 +2193,8 @@ static int codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, direction, idx, mask, val, false); + return codec_amp_update(codec, nid, ch, direction, idx, mask, val, + false, codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_update); @@ -2241,7 +2241,8 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_amp_stereo); int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch, int dir, int idx, int mask, int val) { - return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true); + return codec_amp_update(codec, nid, ch, dir, idx, mask, val, true, + codec->cached_write); } EXPORT_SYMBOL_GPL(snd_hda_codec_amp_init); @@ -2383,8 +2384,8 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, maxval = get_amp_max_value(codec, nid, dir, 0); if (val > maxval) val = maxval; - return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, - HDA_AMP_VOLMASK, val); + return codec_amp_update(codec, nid, ch, dir, idx, HDA_AMP_VOLMASK, val, + false, !hda_codec_is_power_on(codec)); } /** @@ -2434,14 +2435,12 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp); valp++; } if (chs & 2) change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_volume_put); @@ -3109,19 +3108,19 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, long *valp = ucontrol->value.integer.value; int change = 0; - snd_hda_power_up(codec); if (chs & 1) { - change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change = codec_amp_update(codec, nid, 0, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); valp++; } if (chs & 2) - change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - HDA_AMP_MUTE, - *valp ? 0 : HDA_AMP_MUTE); + change |= codec_amp_update(codec, nid, 1, dir, idx, + HDA_AMP_MUTE, + *valp ? 0 : HDA_AMP_MUTE, false, + !hda_codec_is_power_on(codec)); hda_call_check_power_status(codec, nid); - snd_hda_power_down(codec); return change; } EXPORT_SYMBOL_GPL(snd_hda_mixer_amp_switch_put); From 229d043096ea8e58829d37d35767afeac15997f5 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:03 -0600 Subject: [PATCH 025/162] ALSA: core: selection of audio_tstamp type and accuracy reports Audio timestamps can be extracted from sample counters, wall clocks, PHC clocks (Ethernet AVB), on-demand synchronized snapshots. This patch provides the ability to report timestamping capabilities, select timestamp types and retrieve timestamp accuracy, if supported. Details can be found in Documentations/sound/alsa/timestamping.txt This functionality is introduced by reclaiming the reserved_aligned field introduced by commit9c7066aef4a5eb8e4063de28f06c508bf6f2963a in snd_pcm_status to provide userspace with selection/query capabilities. Additional driver_tstamp and audio_tstamp_accuracy fields are also added. snd_pcm_mmap_status remains a read-only structure with only the audio timestamp value accessible from user space. The selection of audio timestamp type is done through snd_pcm_status only This commit does not impact ABI and does not impact the default behavior. By default audio timestamp is aligned with hw_pointer and reports the DMA position. Backwards compatibility is handled by using the HDAudio wall clock for playback and the hw_ptr for all other cases. For timestamp selection a new STATUS_EXT ioctl is introduced with read/write parameters. Alsa-lib will be modified to make use of STATUS_EXT. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++ include/sound/pcm.h | 60 +++++++ include/uapi/sound/asound.h | 34 +++- 3 files changed, 290 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt new file mode 100644 index 000000000000..0b191a23f534 --- /dev/null +++ b/Documentation/sound/alsa/timestamping.txt @@ -0,0 +1,200 @@ +The ALSA API can provide two different system timestamps: + +- Trigger_tstamp is the system time snapshot taken when the .trigger +callback is invoked. This snapshot is taken by the ALSA core in the +general case, but specific hardware may have synchronization +capabilities or conversely may only be able to provide a correct +estimate with a delay. In the latter two cases, the low-level driver +is responsible for updating the trigger_tstamp at the most appropriate +and precise moment. Applications should not rely solely on the first +trigger_tstamp but update their internal calculations if the driver +provides a refined estimate with a delay. + +- tstamp is the current system timestamp updated during the last +event or application query. +The difference (tstamp - trigger_tstamp) defines the elapsed time. + +The ALSA API provides reports two basic pieces of information, avail +and delay, which combined with the trigger and current system +timestamps allow for applications to keep track of the 'fullness' of +the ring buffer and the amount of queued samples. + +The use of these different pointers and time information depends on +the application needs: + +- 'avail' reports how much can be written in the ring buffer +- 'delay' reports the time it will take to hear a new sample after all +queued samples have been played out. + +When timestamps are enabled, the avail/delay information is reported +along with a snapshot of system time. Applications can select from +CLOCK_REALTIME (NTP corrections including going backwards), +CLOCK_MONOTONIC (NTP corrections but never going backwards), +CLOCK_MONOTIC_RAW (without NTP corrections) and change the mode +dynamically with sw_params + + +The ALSA API also provide an audio_tstamp which reflects the passage +of time as measured by different components of audio hardware. In +ascii-art, this could be represented as follows (for the playback +case): + + +--------------------------------------------------------------> time + ^ ^ ^ ^ ^ + | | | | | + analog link dma app FullBuffer + time time time time time + | | | | | + |< codec delay >|<--hw delay-->||<---avail->| + |<----------------- delay---------------------->| | + |<----ring buffer length---->| + +The analog time is taken at the last stage of the playback, as close +as possible to the actual transducer + +The link time is taken at the output of the SOC/chipset as the samples +are pushed on a link. The link time can be directly measured if +supported in hardware by sample counters or wallclocks (e.g. with +HDAudio 24MHz or PTP clock for networked solutions) or indirectly +estimated (e.g. with the frame counter in USB). + +The DMA time is measured using counters - typically the least reliable +of all measurements due to the bursty natured of DMA transfers. + +The app time corresponds to the time tracked by an application after +writing in the ring buffer. + +The application can query what the hardware supports, define which +audio time it wants reported by selecting the relevant settings in +audio_tstamp_config fields, get an estimate of the timestamp +accuracy. It can also request the delay-to-analog be included in the +measurement. Direct access to the link time is very interesting on +platforms that provide an embedded DSP; measuring directly the link +time with dedicated hardware, possibly synchronized with system time, +removes the need to keep track of internal DSP processing times and +latency. + +In case the application requests an audio tstamp that is not supported +in hardware/low-level driver, the type is overridden as DEFAULT and the +timestamp will report the DMA time based on the hw_pointer value. + +For backwards compatibility with previous implementations that did not +provide timestamp selection, with a zero-valued COMPAT timestamp type +the results will default to the HDAudio wall clock for playback +streams and to the DMA time (hw_ptr) in all other cases. + +The audio timestamp accuracy can be returned to user-space, so that +appropriate decisions are made: + +- for dma time (default), the granularity of the transfers can be + inferred from the steps between updates and in turn provide + information on how much the application pointer can be rewound + safely. + +- the link time can be used to track long-term drifts between audio + and system time using the (tstamp-trigger_tstamp)/audio_tstamp + ratio, the precision helps define how much smoothing/low-pass + filtering is required. The link time can be either reset on startup + or reported as is (the latter being useful to compare progress of + different streams - but may require the wallclock to be always + running and not wrap-around during idle periods). If supported in + hardware, the absolute link time could also be used to define a + precise start time (patches WIP) + +- including the delay in the audio timestamp may + counter-intuitively not increase the precision of timestamps, e.g. if a + codec includes variable-latency DSP processing or a chain of + hardware components the delay is typically not known with precision. + +The accuracy is reported in nanosecond units (using an unsigned 32-bit +word), which gives a max precision of 4.29s, more than enough for +audio applications... + +Due to the varied nature of timestamping needs, even for a single +application, the audio_tstamp_config can be changed dynamically. In +the STATUS ioctl, the parameters are read-only and do not allow for +any application selection. To work around this limitation without +impacting legacy applications, a new STATUS_EXT ioctl is introduced +with read/write parameters. ALSA-lib will be modified to make use of +STATUS_EXT and effectively deprecate STATUS. + +The ALSA API only allows for a single audio timestamp to be reported +at a time. This is a conscious design decision, reading the audio +timestamps from hardware registers or from IPC takes time, the more +timestamps are read the more imprecise the combined measurements +are. To avoid any interpretation issues, a single (system, audio) +timestamp is reported. Applications that need different timestamps +will be required to issue multiple queries and perform an +interpolation of the results + +In some hardware-specific configuration, the system timestamp is +latched by a low-level audio subsytem, and the information provided +back to the driver. Due to potential delays in the communication with +the hardware, there is a risk of misalignment with the avail and delay +information. To make sure applications are not confused, a +driver_timestamp field is added in the snd_pcm_status structure; this +timestamp shows when the information is put together by the driver +before returning from the STATUS and STATUS_EXT ioctl. in most cases +this driver_timestamp will be identical to the regular system tstamp. + +Examples of typestamping with HDaudio: + +1. DMA timestamp, no compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 +playback: systime: 341121338 nsec, audio time 342000000 nsec, systime delta -878662 +playback: systime: 426236663 nsec, audio time 427187500 nsec, systime delta -950837 +playback: systime: 597080580 nsec, audio time 598000000 nsec, systime delta -919420 +playback: systime: 682059782 nsec, audio time 683020833 nsec, systime delta -961051 +playback: systime: 852896415 nsec, audio time 853854166 nsec, systime delta -957751 +playback: systime: 937903344 nsec, audio time 938854166 nsec, systime delta -950822 + +2. DMA timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=1 -d +playback: systime: 341053347 nsec, audio time 341062500 nsec, systime delta -9153 +playback: systime: 426072447 nsec, audio time 426062500 nsec, systime delta 9947 +playback: systime: 596899518 nsec, audio time 596895833 nsec, systime delta 3685 +playback: systime: 681915317 nsec, audio time 681916666 nsec, systime delta -1349 +playback: systime: 852741306 nsec, audio time 852750000 nsec, systime delta -8694 + +3. link timestamp, compensation for DMA+analog delay +$ ./audio_time -p --ts_type=2 -d +playback: systime: 341060004 nsec, audio time 341062791 nsec, systime delta -2787 +playback: systime: 426242074 nsec, audio time 426244875 nsec, systime delta -2801 +playback: systime: 597080992 nsec, audio time 597084583 nsec, systime delta -3591 +playback: systime: 682084512 nsec, audio time 682088291 nsec, systime delta -3779 +playback: systime: 852936229 nsec, audio time 852940916 nsec, systime delta -4687 +playback: systime: 938107562 nsec, audio time 938112708 nsec, systime delta -5146 + +Example 1 shows that the timestamp at the DMA level is close to 1ms +ahead of the actual playback time (as a side time this sort of +measurement can help define rewind safeguards). Compensating for the +DMA-link delay in example 2 helps remove the hardware buffering abut +the information is still very jittery, with up to one sample of +error. In example 3 where the timestamps are measured with the link +wallclock, the timestamps show a monotonic behavior and a lower +dispersion. + +Example 3 and 4 are with USB audio class. Example 3 shows a high +offset between audio time and system time due to buffering. Example 4 +shows how compensating for the delay exposes a 1ms accuracy (due to +the use of the frame counter by the driver) + +Example 3: DMA timestamp, no compensation for delay, delta of ~5ms +$ ./audio_time -p -Dhw:1 -t1 +playback: systime: 120174019 nsec, audio time 125000000 nsec, systime delta -4825981 +playback: systime: 245041136 nsec, audio time 250000000 nsec, systime delta -4958864 +playback: systime: 370106088 nsec, audio time 375000000 nsec, systime delta -4893912 +playback: systime: 495040065 nsec, audio time 500000000 nsec, systime delta -4959935 +playback: systime: 620038179 nsec, audio time 625000000 nsec, systime delta -4961821 +playback: systime: 745087741 nsec, audio time 750000000 nsec, systime delta -4912259 +playback: systime: 870037336 nsec, audio time 875000000 nsec, systime delta -4962664 + +Example 4: DMA timestamp, compensation for delay, delay of ~1ms +$ ./audio_time -p -Dhw:1 -t1 -d +playback: systime: 120190520 nsec, audio time 120000000 nsec, systime delta 190520 +playback: systime: 245036740 nsec, audio time 244000000 nsec, systime delta 1036740 +playback: systime: 370034081 nsec, audio time 369000000 nsec, systime delta 1034081 +playback: systime: 495159907 nsec, audio time 494000000 nsec, systime delta 1159907 +playback: systime: 620098824 nsec, audio time 619000000 nsec, systime delta 1098824 +playback: systime: 745031847 nsec, audio time 744000000 nsec, systime delta 1031847 diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c0ddb7e69c28..60f0e48f7905 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -60,6 +60,9 @@ struct snd_pcm_hardware { struct snd_pcm_substream; +struct snd_pcm_audio_tstamp_config; /* definitions further down */ +struct snd_pcm_audio_tstamp_report; + struct snd_pcm_ops { int (*open)(struct snd_pcm_substream *substream); int (*close)(struct snd_pcm_substream *substream); @@ -281,6 +284,58 @@ struct snd_pcm_hw_constraint_ranges { struct snd_pcm_hwptr_log; +/* + * userspace-provided audio timestamp config to kernel, + * structure is for internal use only and filled with dedicated unpack routine + */ +struct snd_pcm_audio_tstamp_config { + /* 5 of max 16 bits used */ + u32 type_requested:4; + u32 report_delay:1; /* add total delay to A/D or D/A */ +}; + +static inline void snd_pcm_unpack_audio_tstamp_config(__u32 data, + struct snd_pcm_audio_tstamp_config *config) +{ + config->type_requested = data & 0xF; + config->report_delay = (data >> 4) & 1; +} + +/* + * kernel-provided audio timestamp report to user-space + * structure is for internal use only and read by dedicated pack routine + */ +struct snd_pcm_audio_tstamp_report { + /* 6 of max 16 bits used for bit-fields */ + + /* for backwards compatibility */ + u32 valid:1; + + /* actual type if hardware could not support requested timestamp */ + u32 actual_type:4; + + /* accuracy represented in ns units */ + u32 accuracy_report:1; /* 0 if accuracy unknown, 1 if accuracy field is valid */ + u32 accuracy; /* up to 4.29s, will be packed in separate field */ +}; + +static inline void snd_pcm_pack_audio_tstamp_report(__u32 *data, __u32 *accuracy, + const struct snd_pcm_audio_tstamp_report *report) +{ + u32 tmp; + + tmp = report->accuracy_report; + tmp <<= 4; + tmp |= report->actual_type; + tmp <<= 1; + tmp |= report->valid; + + *data &= 0xffff; /* zero-clear MSBs */ + *data |= (tmp << 16); + *accuracy = report->accuracy; +} + + struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; @@ -361,6 +416,11 @@ struct snd_pcm_runtime { struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */ + /* -- audio timestamp config -- */ + struct snd_pcm_audio_tstamp_config audio_tstamp_config; + struct snd_pcm_audio_tstamp_report audio_tstamp_report; + struct timespec driver_tstamp; + #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 0e88e7a0f0eb..acef4e4d2735 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -267,10 +267,17 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ -#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* (Deprecated)has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_LINK_ATIME 0x01000000 /* report hardware link audio time, reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ +#define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ + #define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ + + typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */ #define SNDRV_PCM_STATE_SETUP ((__force snd_pcm_state_t) 1) /* stream has a setup */ @@ -408,6 +415,22 @@ struct snd_pcm_channel_info { unsigned int step; /* samples distance in bits */ }; +enum { + /* + * first definition for backwards compatibility only, + * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else + */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, + + /* timestamp definitions */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /* DMA time, reported as per hw_ptr */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /* link time reported by sample or wallclock counter, reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /* link time reported by sample or wallclock counter, not reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /* link time estimated indirectly */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /* link time synchronized with system time */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED +}; + struct snd_pcm_status { snd_pcm_state_t state; /* stream state */ struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */ @@ -419,9 +442,11 @@ struct snd_pcm_status { snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ snd_pcm_state_t suspended_state; /* suspended stream state */ - __u32 reserved_alignment; /* must be filled with zero */ - struct timespec audio_tstamp; /* from sample counter or wall clock */ - unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */ + __u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */ + struct timespec audio_tstamp; /* sample counter, wall clock, PHC or on-demand sync'ed */ + struct timespec driver_tstamp; /* useful in case reference system tstamp is reported with delay */ + __u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */ + unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -534,6 +559,7 @@ enum { #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) #define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr) +#define SNDRV_PCM_IOCTL_STATUS_EXT _IOWR('A', 0x24, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info) #define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40) #define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41) From 38ca2a4d58bbc45973ee5cd14e4b803ee5ad69f0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:04 -0600 Subject: [PATCH 026/162] ALSA: core: pass audio tstamp config from userspace Let userspace select audio timestamp config when the STATUS_EXT ioctl is used, ignore and zero all other fields No change for the existing STATUS ioctl, parameters are treated as read-only. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index b03a638b420c..72323a8c35c8 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -753,12 +753,21 @@ int snd_pcm_status(struct snd_pcm_substream *substream, } static int snd_pcm_status_user(struct snd_pcm_substream *substream, - struct snd_pcm_status __user * _status) + struct snd_pcm_status __user * _status, + bool ext) { struct snd_pcm_status status; int res; - + memset(&status, 0, sizeof(status)); + /* + * with extension, parameters are read/write, + * get audio_tstamp_data from user, + * ignore rest of status structure + */ + if (ext && get_user(status.audio_tstamp_data, + (u32 __user *)(&_status->audio_tstamp_data))) + return -EFAULT; res = snd_pcm_status(substream, &status); if (res < 0) return res; @@ -2723,7 +2732,9 @@ static int snd_pcm_common_ioctl1(struct file *file, case SNDRV_PCM_IOCTL_SW_PARAMS: return snd_pcm_sw_params_user(substream, arg); case SNDRV_PCM_IOCTL_STATUS: - return snd_pcm_status_user(substream, arg); + return snd_pcm_status_user(substream, arg, false); + case SNDRV_PCM_IOCTL_STATUS_EXT: + return snd_pcm_status_user(substream, arg, true); case SNDRV_PCM_IOCTL_CHANNEL_INFO: return snd_pcm_channel_info_user(substream, arg); case SNDRV_PCM_IOCTL_PREPARE: From 5442a73a009231598fb5ea065c4e3f9daa30d8cc Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:05 -0600 Subject: [PATCH 027/162] ALSA: core: pass audio tstamp config from userspace in compat mode Let userspace select audio timestamp config, ignore and zero all other fields Use audio_tstamp_data to retrieve config and pass report back to user space Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/core/pcm_compat.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 2d957ba63557..b48b434444ed 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -194,18 +194,30 @@ struct snd_pcm_status32 { u32 avail_max; u32 overrange; s32 suspended_state; - u32 reserved_alignment; + u32 audio_tstamp_data; struct compat_timespec audio_tstamp; - unsigned char reserved[56-sizeof(struct compat_timespec)]; + struct compat_timespec driver_tstamp; + u32 audio_tstamp_accuracy; + unsigned char reserved[52-2*sizeof(struct compat_timespec)]; } __attribute__((packed)); static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, - struct snd_pcm_status32 __user *src) + struct snd_pcm_status32 __user *src, + bool ext) { struct snd_pcm_status status; int err; + memset(&status, 0, sizeof(status)); + /* + * with extension, parameters are read/write, + * get audio_tstamp_data from user, + * ignore rest of status structure + */ + if (ext && get_user(status.audio_tstamp_data, + (u32 __user *)(&src->audio_tstamp_data))) + return -EFAULT; err = snd_pcm_status(substream, &status); if (err < 0) return err; @@ -222,7 +234,10 @@ static int snd_pcm_status_user_compat(struct snd_pcm_substream *substream, put_user(status.avail_max, &src->avail_max) || put_user(status.overrange, &src->overrange) || put_user(status.suspended_state, &src->suspended_state) || - compat_put_timespec(&status.audio_tstamp, &src->audio_tstamp)) + put_user(status.audio_tstamp_data, &src->audio_tstamp_data) || + compat_put_timespec(&status.audio_tstamp, &src->audio_tstamp) || + compat_put_timespec(&status.driver_tstamp, &src->driver_tstamp) || + put_user(status.audio_tstamp_accuracy, &src->audio_tstamp_accuracy)) return -EFAULT; return err; @@ -457,6 +472,7 @@ enum { SNDRV_PCM_IOCTL_HW_PARAMS32 = _IOWR('A', 0x11, struct snd_pcm_hw_params32), SNDRV_PCM_IOCTL_SW_PARAMS32 = _IOWR('A', 0x13, struct snd_pcm_sw_params32), SNDRV_PCM_IOCTL_STATUS32 = _IOR('A', 0x20, struct snd_pcm_status32), + SNDRV_PCM_IOCTL_STATUS_EXT32 = _IOWR('A', 0x24, struct snd_pcm_status32), SNDRV_PCM_IOCTL_DELAY32 = _IOR('A', 0x21, s32), SNDRV_PCM_IOCTL_CHANNEL_INFO32 = _IOR('A', 0x32, struct snd_pcm_channel_info32), SNDRV_PCM_IOCTL_REWIND32 = _IOW('A', 0x46, u32), @@ -517,7 +533,9 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l case SNDRV_PCM_IOCTL_SW_PARAMS32: return snd_pcm_ioctl_sw_params_compat(substream, argp); case SNDRV_PCM_IOCTL_STATUS32: - return snd_pcm_status_user_compat(substream, argp); + return snd_pcm_status_user_compat(substream, argp, false); + case SNDRV_PCM_IOCTL_STATUS_EXT32: + return snd_pcm_status_user_compat(substream, argp, true); case SNDRV_PCM_IOCTL_SYNC_PTR32: return snd_pcm_ioctl_sync_ptr_compat(substream, argp); case SNDRV_PCM_IOCTL_CHANNEL_INFO32: From 3179f62001880e588e229db3006a59ad87b7792a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:06 -0600 Subject: [PATCH 028/162] ALSA: core: add .get_time_info Introduce more generic .get_time_info to retrieve system timestamp and audio timestamp in single routine. Backwards compatibility is preserved with same functionality as with .wall_clock method (to be removed in following commits to avoid breaking git bisect) Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 4 ++ sound/core/pcm_lib.c | 86 +++++++++++++++++++++++++++-------------- sound/core/pcm_native.c | 24 ++++++++++++ 3 files changed, 86 insertions(+), 28 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 60f0e48f7905..04f2d492ae57 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -76,6 +76,10 @@ struct snd_pcm_ops { snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); int (*wall_clock)(struct snd_pcm_substream *substream, struct timespec *audio_ts); + int (*get_time_info)(struct snd_pcm_substream *substream, + struct timespec *system_ts, struct timespec *audio_ts, + struct snd_pcm_audio_tstamp_config *audio_tstamp_config, + struct snd_pcm_audio_tstamp_report *audio_tstamp_report); int (*copy)(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *buf, snd_pcm_uframes_t count); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ffd656012ab8..ac6b33f3779c 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -232,6 +232,49 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return 0; } +static void update_audio_tstamp(struct snd_pcm_substream *substream, + struct timespec *curr_tstamp, + struct timespec *audio_tstamp) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + u64 audio_frames, audio_nsecs; + struct timespec driver_tstamp; + + if (runtime->tstamp_mode != SNDRV_PCM_TSTAMP_ENABLE) + return; + + if (!(substream->ops->get_time_info) || + (runtime->audio_tstamp_report.actual_type == + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) { + + /* + * provide audio timestamp derived from pointer position + * add delay only if requested + */ + + audio_frames = runtime->hw_ptr_wrap + runtime->status->hw_ptr; + + if (runtime->audio_tstamp_config.report_delay) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + audio_frames -= runtime->delay; + else + audio_frames += runtime->delay; + } + audio_nsecs = div_u64(audio_frames * 1000000000LL, + runtime->rate); + *audio_tstamp = ns_to_timespec(audio_nsecs); + } + runtime->status->audio_tstamp = *audio_tstamp; + runtime->status->tstamp = *curr_tstamp; + + /* + * re-take a driver timestamp to let apps detect if the reference tstamp + * read by low-level hardware was provided with a delay + */ + snd_pcm_gettime(substream->runtime, (struct timespec *)&driver_tstamp); + runtime->driver_tstamp = driver_tstamp; +} + static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, unsigned int in_interrupt) { @@ -256,11 +299,18 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, pos = substream->ops->pointer(substream); curr_jiffies = jiffies; if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { - snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); + if ((substream->ops->get_time_info) && + (runtime->audio_tstamp_config.type_requested != SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) { + substream->ops->get_time_info(substream, &curr_tstamp, + &audio_tstamp, + &runtime->audio_tstamp_config, + &runtime->audio_tstamp_report); - if ((runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK) && - (substream->ops->wall_clock)) - substream->ops->wall_clock(substream, &audio_tstamp); + /* re-test in case tstamp type is not supported in hardware and was demoted to DEFAULT */ + if (runtime->audio_tstamp_report.actual_type == SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT) + snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); + } else + snd_pcm_gettime(runtime, (struct timespec *)&curr_tstamp); } if (pos == SNDRV_PCM_POS_XRUN) { @@ -403,8 +453,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, } no_delta_check: - if (runtime->status->hw_ptr == new_hw_ptr) + if (runtime->status->hw_ptr == new_hw_ptr) { + update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return 0; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) @@ -426,30 +478,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, snd_BUG_ON(crossed_boundary != 1); runtime->hw_ptr_wrap += runtime->boundary; } - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { - runtime->status->tstamp = curr_tstamp; - if (!(runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK)) { - /* - * no wall clock available, provide audio timestamp - * derived from pointer position+delay - */ - u64 audio_frames, audio_nsecs; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - audio_frames = runtime->hw_ptr_wrap - + runtime->status->hw_ptr - - runtime->delay; - else - audio_frames = runtime->hw_ptr_wrap - + runtime->status->hw_ptr - + runtime->delay; - audio_nsecs = div_u64(audio_frames * 1000000000LL, - runtime->rate); - audio_tstamp = ns_to_timespec(audio_nsecs); - } - runtime->status->audio_tstamp = audio_tstamp; - } + update_audio_tstamp(substream, &curr_tstamp, &audio_tstamp); return snd_pcm_update_state(substream, runtime); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 72323a8c35c8..9c2c6f801fed 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -707,6 +707,23 @@ int snd_pcm_status(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_stream_lock_irq(substream); + + snd_pcm_unpack_audio_tstamp_config(status->audio_tstamp_data, + &runtime->audio_tstamp_config); + + /* backwards compatible behavior */ + if (runtime->audio_tstamp_config.type_requested == + SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT) { + if (runtime->hw.info & SNDRV_PCM_INFO_HAS_WALL_CLOCK) + runtime->audio_tstamp_config.type_requested = + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK; + else + runtime->audio_tstamp_config.type_requested = + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; + runtime->audio_tstamp_report.valid = 0; + } else + runtime->audio_tstamp_report.valid = 1; + status->state = runtime->status->state; status->suspended_state = runtime->status->suspended_state; if (status->state == SNDRV_PCM_STATE_OPEN) @@ -716,8 +733,15 @@ int snd_pcm_status(struct snd_pcm_substream *substream, snd_pcm_update_hw_ptr(substream); if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) { status->tstamp = runtime->status->tstamp; + status->driver_tstamp = runtime->driver_tstamp; status->audio_tstamp = runtime->status->audio_tstamp; + if (runtime->audio_tstamp_report.valid == 1) + /* backwards compatibility, no report provided in COMPAT mode */ + snd_pcm_pack_audio_tstamp_report(&status->audio_tstamp_data, + &status->audio_tstamp_accuracy, + &runtime->audio_tstamp_report); + goto _tstamp_end; } } else { From 9e94df3a624b1b485f2c2ac5ab94032da01e45b3 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:07 -0600 Subject: [PATCH 029/162] ALSA: hda: replace .wallclock by .get_time_info No real functional change, only take wall clock and system time in same routine and add accuracy report. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 40 ++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index dfcb5e929f9f..7806f1d297cb 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -732,17 +732,32 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) azx_get_position(chip, azx_dev)); } -static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, - struct timespec *ts) +static int azx_get_time_info(struct snd_pcm_substream *substream, + struct timespec *system_ts, struct timespec *audio_ts, + struct snd_pcm_audio_tstamp_config *audio_tstamp_config, + struct snd_pcm_audio_tstamp_report *audio_tstamp_report) { struct azx_dev *azx_dev = get_azx_dev(substream); u64 nsec; - nsec = timecounter_read(&azx_dev->azx_tc); - nsec = div_u64(nsec, 3); /* can be optimized */ - nsec = azx_adjust_codec_delay(substream, nsec); + if ((substream->runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_ATIME) && + (audio_tstamp_config->type_requested == SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK)) { - *ts = ns_to_timespec(nsec); + snd_pcm_gettime(substream->runtime, system_ts); + + nsec = timecounter_read(&azx_dev->azx_tc); + nsec = div_u64(nsec, 3); /* can be optimized */ + if (audio_tstamp_config->report_delay) + nsec = azx_adjust_codec_delay(substream, nsec); + + *audio_ts = ns_to_timespec(nsec); + + audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK; + audio_tstamp_report->accuracy_report = 1; /* rest of structure is valid */ + audio_tstamp_report->accuracy = 42; /* 24 MHz WallClock == 42ns resolution */ + + } else + audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; return 0; } @@ -756,7 +771,8 @@ static struct snd_pcm_hardware azx_pcm_hw = { /* SNDRV_PCM_INFO_RESUME |*/ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_SYNC_START | - SNDRV_PCM_INFO_HAS_WALL_CLOCK | + SNDRV_PCM_INFO_HAS_WALL_CLOCK | /* legacy */ + SNDRV_PCM_INFO_HAS_LINK_ATIME | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_48000, @@ -842,10 +858,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) return -EINVAL; } - /* disable WALLCLOCK timestamps for capture streams + /* disable LINK_ATIME timestamps for capture streams until we figure out how to handle digital inputs */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK; + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK; /* legacy */ + runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_LINK_ATIME; + } spin_lock_irqsave(&chip->reg_lock, flags); azx_dev->substream = substream; @@ -877,7 +895,7 @@ static struct snd_pcm_ops azx_pcm_ops = { .prepare = azx_pcm_prepare, .trigger = azx_pcm_trigger, .pointer = azx_pcm_pointer, - .wall_clock = azx_get_wallclock_tstamp, + .get_time_info = azx_get_time_info, .mmap = azx_pcm_mmap, .page = snd_pcm_sgbuf_ops_page, }; From 2d52a5abd5be35340296a251092c8a594679df54 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:08 -0600 Subject: [PATCH 030/162] ALSA: core: remove .wall_clock can be removed without breaking git-bisect now Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 04f2d492ae57..0cb7f3f5df7b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -74,8 +74,6 @@ struct snd_pcm_ops { int (*prepare)(struct snd_pcm_substream *substream); int (*trigger)(struct snd_pcm_substream *substream, int cmd); snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *substream); - int (*wall_clock)(struct snd_pcm_substream *substream, - struct timespec *audio_ts); int (*get_time_info)(struct snd_pcm_substream *substream, struct timespec *system_ts, struct timespec *audio_ts, struct snd_pcm_audio_tstamp_config *audio_tstamp_config, From c72638bdaabe9ea4b09003b9db7e1754f472fbed Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 13 Feb 2015 15:14:09 -0600 Subject: [PATCH 031/162] ALSA: bump PCM protocol to 2.0.13 Bump PCM protocol to enable use of STATUS_EXT ioctls, older apps will still use STATUS and audio timestamp configuration is not supported (backwards compatible behavior). Signed-off-by: Pierre-Louis Bossart Signed-off-by: Takashi Iwai --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index acef4e4d2735..3d46e9a0cd2e 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -140,7 +140,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 13) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; From ad876c862278be59147d4004f1a7c4d492e4ec96 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 16:26:00 +0100 Subject: [PATCH 032/162] ALSA: pcm: Minor refactoring in snd_pcm_attach_substream() No functional changes at all. Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 41 +++++++++++++++-------------------------- 1 file changed, 15 insertions(+), 26 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 0345e53a340c..d63d262c571e 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -919,6 +919,9 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, if (snd_BUG_ON(!pcm || !rsubstream)) return -ENXIO; + if (snd_BUG_ON(stream != SNDRV_PCM_STREAM_PLAYBACK && + stream != SNDRV_PCM_STREAM_CAPTURE)) + return -EINVAL; *rsubstream = NULL; pstr = &pcm->streams[stream]; if (pstr->substream == NULL || pstr->substream_count == 0) @@ -927,25 +930,14 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, card = pcm->card; prefer_subdevice = snd_ctl_get_preferred_subdevice(card, SND_CTL_SUBDEV_PCM); - switch (stream) { - case SNDRV_PCM_STREAM_PLAYBACK: - if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) { - for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; substream; substream = substream->next) { - if (SUBSTREAM_BUSY(substream)) - return -EAGAIN; - } + if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) { + int opposite = !stream; + + for (substream = pcm->streams[opposite].substream; substream; + substream = substream->next) { + if (SUBSTREAM_BUSY(substream)) + return -EAGAIN; } - break; - case SNDRV_PCM_STREAM_CAPTURE: - if (pcm->info_flags & SNDRV_PCM_INFO_HALF_DUPLEX) { - for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; substream; substream = substream->next) { - if (SUBSTREAM_BUSY(substream)) - return -EAGAIN; - } - } - break; - default: - return -EINVAL; } if (file->f_flags & O_APPEND) { @@ -968,15 +960,12 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, return 0; } - if (prefer_subdevice >= 0) { - for (substream = pstr->substream; substream; substream = substream->next) - if (!SUBSTREAM_BUSY(substream) && substream->number == prefer_subdevice) - goto __ok; - } - for (substream = pstr->substream; substream; substream = substream->next) - if (!SUBSTREAM_BUSY(substream)) + for (substream = pstr->substream; substream; substream = substream->next) { + if (!SUBSTREAM_BUSY(substream) && + (prefer_subdevice == -1 || + substream->number == prefer_subdevice)) break; - __ok: + } if (substream == NULL) return -EAGAIN; From b95bd3a454cf9e9e111b6b87c02550368fe6e802 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 16:49:04 +0100 Subject: [PATCH 033/162] ALSA: pcm: Don't add internal PCMs to PCM device list An internal PCM object shouldn't be added to the PCM device list, as it's never accessed directly from the user-space, and it has no proc or any similar accesses. Currently, it's excluded in snd_pcm_get() and snd_pcm_next(), but it's easier not to add such an object to the list. Actually, the whole snd_pcm_dev_register() can be skipped for an internal PCM. So this patch changes the code there, but also addresses the uninitialized list_head access. Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index d63d262c571e..d440629e08e2 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -49,8 +49,6 @@ static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { - if (pcm->internal) - continue; if (pcm->card == card && pcm->device == device) return pcm; } @@ -62,8 +60,6 @@ static int snd_pcm_next(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { - if (pcm->internal) - continue; if (pcm->card == card && pcm->device > device) return pcm->device; else if (pcm->card->number > card->number) @@ -76,6 +72,9 @@ static int snd_pcm_add(struct snd_pcm *newpcm) { struct snd_pcm *pcm; + if (newpcm->internal) + return 0; + list_for_each_entry(pcm, &snd_pcm_devices, list) { if (pcm->card == newpcm->card && pcm->device == newpcm->device) return -EBUSY; @@ -782,6 +781,9 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, pcm->card = card; pcm->device = device; pcm->internal = internal; + mutex_init(&pcm->open_mutex); + init_waitqueue_head(&pcm->open_wait); + INIT_LIST_HEAD(&pcm->list); if (id) strlcpy(pcm->id, id, sizeof(pcm->id)); if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { @@ -792,8 +794,6 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, snd_pcm_free(pcm); return err; } - mutex_init(&pcm->open_mutex); - init_waitqueue_head(&pcm->open_wait); if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) { snd_pcm_free(pcm); return err; @@ -1075,15 +1075,16 @@ static int snd_pcm_dev_register(struct snd_device *device) if (snd_BUG_ON(!device || !device->device_data)) return -ENXIO; pcm = device->device_data; + if (pcm->internal) + return 0; + mutex_lock(®ister_mutex); err = snd_pcm_add(pcm); - if (err) { - mutex_unlock(®ister_mutex); - return err; - } + if (err) + goto unlock; for (cidx = 0; cidx < 2; cidx++) { int devtype = -1; - if (pcm->streams[cidx].substream == NULL || pcm->internal) + if (pcm->streams[cidx].substream == NULL) continue; switch (cidx) { case SNDRV_PCM_STREAM_PLAYBACK: @@ -1098,9 +1099,8 @@ static int snd_pcm_dev_register(struct snd_device *device) &snd_pcm_f_ops[cidx], pcm, &pcm->streams[cidx].dev); if (err < 0) { - list_del(&pcm->list); - mutex_unlock(®ister_mutex); - return err; + list_del_init(&pcm->list); + goto unlock; } for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) @@ -1110,8 +1110,9 @@ static int snd_pcm_dev_register(struct snd_device *device) list_for_each_entry(notify, &snd_pcm_notify_list, list) notify->n_register(pcm); + unlock: mutex_unlock(®ister_mutex); - return 0; + return err; } static int snd_pcm_dev_disconnect(struct snd_device *device) From 646e1dd8f9f47c57560ce81c02fdd57ff0929bc6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 17:04:08 +0100 Subject: [PATCH 034/162] ALSA: pcm: Don't notify internal PCMs Notifier shouldn't listen to the changes of internal PCMs. Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index d440629e08e2..542dbc6f54e8 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -888,8 +888,9 @@ static int snd_pcm_free(struct snd_pcm *pcm) if (!pcm) return 0; - list_for_each_entry(notify, &snd_pcm_notify_list, list) { - notify->n_unregister(pcm); + if (!pcm->internal) { + list_for_each_entry(notify, &snd_pcm_notify_list, list) + notify->n_unregister(pcm); } if (pcm->private_free) pcm->private_free(pcm); @@ -1129,7 +1130,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) mutex_lock(&pcm->open_mutex); wake_up(&pcm->open_wait); list_del_init(&pcm->list); - for (cidx = 0; cidx < 2; cidx++) + for (cidx = 0; cidx < 2; cidx++) { for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) { snd_pcm_stream_lock_irq(substream); if (substream->runtime) { @@ -1139,8 +1140,10 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) } snd_pcm_stream_unlock_irq(substream); } - list_for_each_entry(notify, &snd_pcm_notify_list, list) { - notify->n_disconnect(pcm); + } + if (!pcm->internal) { + list_for_each_entry(notify, &snd_pcm_notify_list, list) + notify->n_disconnect(pcm); } for (cidx = 0; cidx < 2; cidx++) { snd_unregister_device(&pcm->streams[cidx].dev); From b20221385c40155f13068be75b865170a1ad5d1e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 17:05:27 +0100 Subject: [PATCH 035/162] ALSA: pcm: Don't ignore internal PCMs in snd_pcm_dev_disconnect() Some codes in snd_pcm_dev_disconnect() are still valid even for internal PCMs, but they are skipped because of the check of list_empty(&pcm->list) at the beginning. Remove this check and put pcm->internal checks appropriately for internal PCM object to process through this function. Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 542dbc6f54e8..e9b87465c73d 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1124,9 +1124,6 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) int cidx; mutex_lock(®ister_mutex); - if (list_empty(&pcm->list)) - goto unlock; - mutex_lock(&pcm->open_mutex); wake_up(&pcm->open_wait); list_del_init(&pcm->list); @@ -1146,14 +1143,14 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) notify->n_disconnect(pcm); } for (cidx = 0; cidx < 2; cidx++) { - snd_unregister_device(&pcm->streams[cidx].dev); + if (!pcm->internal) + snd_unregister_device(&pcm->streams[cidx].dev); if (pcm->streams[cidx].chmap_kctl) { snd_ctl_remove(pcm->card, pcm->streams[cidx].chmap_kctl); pcm->streams[cidx].chmap_kctl = NULL; } } mutex_unlock(&pcm->open_mutex); - unlock: mutex_unlock(®ister_mutex); return 0; } From 7371bd1f4aeb4e1c44b8c1ba8e36ebba4250b59c Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 17 Feb 2015 13:59:26 +0800 Subject: [PATCH 036/162] ASoC: rt5670: Add disabled item in dmic pin enum Currently, we will configure dmic related pin definition if pdata.dmic_en is true. However, there is no disable option in the enum. So, any dmic is used, all 3 dmic related pins will be configured. It may cause unexpected pin definition. This patch adds a disable item for each dmic enum and take it as default. So the driver will not set the pin configuration if we don't set dmicx_data_pin in platform data. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index d11b9c207e26..84857bdaa78b 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1967,17 +1967,20 @@ enum { }; enum { + RT5670_DMIC1_DISABLED, RT5670_DMIC_DATA_GPIO6, RT5670_DMIC_DATA_IN2P, RT5670_DMIC_DATA_GPIO7, }; enum { + RT5670_DMIC2_DISABLED, RT5670_DMIC_DATA_GPIO8, RT5670_DMIC_DATA_IN3N, }; enum { + RT5670_DMIC3_DISABLED, RT5670_DMIC_DATA_GPIO9, RT5670_DMIC_DATA_GPIO10, RT5670_DMIC_DATA_GPIO5, From 65d17a9ce9f24a3aaf7d614251fdcc1b2121765f Mon Sep 17 00:00:00 2001 From: Nikesh Oswal Date: Mon, 16 Feb 2015 15:25:48 +0000 Subject: [PATCH 037/162] ASoC: wm_adsp: Ensure DSP controls are always persistent Currently DSP controls are persistent (across DSP On/Off) only if they were set whilst the DSP is off. This change makes the controls persistent irrespective of when they are set. Signed-off-by: Nikesh Oswal Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 720d6e852986..14414ea23b55 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -420,10 +420,9 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol, memcpy(ctl->cache, p, ctl->len); - if (!ctl->enabled) { - ctl->set = 1; + ctl->set = 1; + if (!ctl->enabled) return 0; - } return wm_coeff_write_control(kcontrol, p, ctl->len); } From be951017453cba2f3eb789413f697b8f14393eec Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 16 Feb 2015 15:25:49 +0000 Subject: [PATCH 038/162] ASoC: wm_adsp: Improve round to next 4-byte boundary Whilst the existing code does correctly round to the next 4-byte boundary it does so rather inefficiently. This patch changes the rounding to be simpler and more efficient. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 14414ea23b55..e625cedb0fa9 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1184,7 +1184,6 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) int ret, pos, blocks, type, offset, reg; char *file; struct wm_adsp_buf *buf; - int tmp; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -1334,12 +1333,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } } - tmp = le32_to_cpu(blk->len) % 4; - if (tmp) - pos += le32_to_cpu(blk->len) + (4 - tmp) + sizeof(*blk); - else - pos += le32_to_cpu(blk->len) + sizeof(*blk); - + pos += (le32_to_cpu(blk->len) + sizeof(*blk) + 3) & ~0x03; blocks++; } From 7ff5eabce4231d199dadc14c23f14a6619f926c0 Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 17 Feb 2015 00:53:12 -0800 Subject: [PATCH 039/162] ASoC: max98357a: Remove use of DRV_NAME Remove use of DRV_NAME define. Signed-off-by: Kenneth Westfield Signed-off-by: Mark Brown --- sound/soc/codecs/max98357a.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index e9e6efbc21dd..4ee23fbc4e12 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -26,8 +26,6 @@ #include #include -#define DRV_NAME "max98357a" - static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -87,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = DRV_NAME, + .name = "max98357a", .playback = { - .stream_name = DRV_NAME "-playback", + .stream_name = "max98357a-playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, @@ -127,7 +125,7 @@ static int max98357a_platform_remove(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id max98357a_device_id[] = { - { .compatible = "maxim," DRV_NAME, }, + { .compatible = "maxim,max98357a" }, {} }; MODULE_DEVICE_TABLE(of, max98357a_device_id); @@ -135,7 +133,7 @@ MODULE_DEVICE_TABLE(of, max98357a_device_id); static struct platform_driver max98357a_platform_driver = { .driver = { - .name = DRV_NAME, + .name = "max98357a", .of_match_table = of_match_ptr(max98357a_device_id), }, .probe = max98357a_platform_probe, @@ -145,4 +143,3 @@ module_platform_driver(max98357a_platform_driver); MODULE_DESCRIPTION("Maxim MAX98357A Codec Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:" DRV_NAME); From 327ef4f02582d01f7eedb291794106823b44a0cf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2015 17:15:04 +0100 Subject: [PATCH 040/162] ALSA: hda - Decouple PCM and hwdep devices from codec object This is a preliminary patch for the hda_bus implementation, removing the parent device setup to codec device. Since the bus and the class devices can't be crossed over, leave the sound devices to the default parent device as is. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 1 - sound/pci/hda/hda_controller.c | 3 --- sound/pci/hda/hda_hwdep.c | 3 --- 3 files changed, 7 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 1e7de08e77cb..d6be4e852c4d 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -168,7 +168,6 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->evbit[0] = BIT_MASK(EV_SND); input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); input_dev->event = snd_hda_beep_event; - input_dev->dev.parent = &codec->dev; input_set_drvdata(input_dev, beep); beep->dev = input_dev; diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ebb7a644bd86..4c7a6f9bfcde 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -958,9 +958,6 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, chip->card->dev, size, MAX_PREALLOC_SIZE); - /* link to codec */ - for (s = 0; s < 2; s++) - pcm->streams[s].dev.parent = &codec->dev; return 0; } diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 11b5a42b4ec8..125f3420fa6a 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -116,9 +116,6 @@ int snd_hda_create_hwdep(struct hda_codec *codec) hwdep->ops.ioctl_compat = hda_hwdep_ioctl_compat; #endif - /* link to codec */ - hwdep->dev.parent = &codec->dev; - /* for sysfs */ hwdep->dev.groups = snd_hda_dev_attr_groups; dev_set_drvdata(&hwdep->dev, codec); From d8a766a16ed90c4b3bd7afa6e1417f8d715db507 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Feb 2015 15:25:37 +0100 Subject: [PATCH 041/162] ALSA: hda - Bind codecs via standard bus Now we create the standard HD-audio bus (/sys/bus/hdaudio), and bind the codec driver with the codec device over there. This is the first step of the whole transition so that the changes to each codec driver are kept as minimal as possible. Each codec driver needs to register hda_codec_driver struct containing the currently existing preset via the new helper macro module_hda_codec_driver(). The old hda_codec_preset_list is replaced with this infrastructure. The generic parsers (for HDMI and other) are also included in the preset with the special IDs to bind uniquely. In HD-audio core side, the device binding code is split to hda_bind.c. It provides the snd_hda_bus_type implementation to match the codec driver with the given codec vendor ID. It also manages the module auto-loading by itself like before: when the matching isn't found, it tries to probe the corresponding codec modules, and finally falls back to the generic drivers. (The special ID mentioned above is set at this stage.) The only visible change to outside is that the hdaudio sysfs entry now appears in /sys/bus/devices, not as a sound class device. More works to move the suspend/resume and remove ops will be (hopefully) done in later patches. Signed-off-by: Takashi Iwai --- sound/pci/hda/Makefile | 2 +- sound/pci/hda/hda_bind.c | 320 +++++++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.c | 307 ++----------------------------- sound/pci/hda/hda_codec.h | 27 ++- sound/pci/hda/hda_generic.c | 18 +- sound/pci/hda/hda_local.h | 18 +- sound/pci/hda/patch_analog.c | 16 +- sound/pci/hda/patch_ca0110.c | 16 +- sound/pci/hda/patch_ca0132.c | 16 +- sound/pci/hda/patch_cirrus.c | 16 +- sound/pci/hda/patch_cmedia.c | 16 +- sound/pci/hda/patch_conexant.c | 16 +- sound/pci/hda/patch_hdmi.c | 27 +-- sound/pci/hda/patch_realtek.c | 16 +- sound/pci/hda/patch_si3054.c | 16 +- sound/pci/hda/patch_sigmatel.c | 16 +- sound/pci/hda/patch_via.c | 16 +- 17 files changed, 401 insertions(+), 478 deletions(-) create mode 100644 sound/pci/hda/hda_bind.c diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 194f30935e77..96caaebfc19d 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -4,7 +4,7 @@ snd-hda-tegra-objs := hda_tegra.o # for haswell power well snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o -snd-hda-codec-y := hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o +snd-hda-codec-y := hda_bind.o hda_codec.o hda_jack.o hda_auto_parser.o hda_sysfs.o snd-hda-codec-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-codec-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o snd-hda-codec-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c new file mode 100644 index 000000000000..adf6b475dee1 --- /dev/null +++ b/sound/pci/hda/hda_bind.c @@ -0,0 +1,320 @@ +/* + * HD-audio codec driver binding + * Copyright (c) Takashi Iwai + */ + +#include +#include +#include +#include +#include +#include +#include "hda_codec.h" +#include "hda_local.h" + +/* codec vendor labels */ +struct hda_vendor_id { + unsigned int id; + const char *name; +}; + +static struct hda_vendor_id hda_vendor_ids[] = { + { 0x1002, "ATI" }, + { 0x1013, "Cirrus Logic" }, + { 0x1057, "Motorola" }, + { 0x1095, "Silicon Image" }, + { 0x10de, "Nvidia" }, + { 0x10ec, "Realtek" }, + { 0x1102, "Creative" }, + { 0x1106, "VIA" }, + { 0x111d, "IDT" }, + { 0x11c1, "LSI" }, + { 0x11d4, "Analog Devices" }, + { 0x13f6, "C-Media" }, + { 0x14f1, "Conexant" }, + { 0x17e8, "Chrontel" }, + { 0x1854, "LG" }, + { 0x1aec, "Wolfson Microelectronics" }, + { 0x1af4, "QEMU" }, + { 0x434d, "C-Media" }, + { 0x8086, "Intel" }, + { 0x8384, "SigmaTel" }, + {} /* terminator */ +}; + +/* + * find a matching codec preset + */ +static int hda_bus_match(struct device *dev, struct device_driver *drv) +{ + struct hda_codec *codec = container_of(dev, struct hda_codec, dev); + struct hda_codec_driver *driver = + container_of(drv, struct hda_codec_driver, driver); + const struct hda_codec_preset *preset; + /* check probe_id instead of vendor_id if set */ + u32 id = codec->probe_id ? codec->probe_id : codec->vendor_id; + + for (preset = driver->preset; preset->id; preset++) { + u32 mask = preset->mask; + + if (preset->afg && preset->afg != codec->afg) + continue; + if (preset->mfg && preset->mfg != codec->mfg) + continue; + if (!mask) + mask = ~0; + if (preset->id == (id & mask) && + (!preset->rev || preset->rev == codec->revision_id)) { + codec->preset = preset; + return 1; + } + } + return 0; +} + +/* reset the codec name from the preset */ +static int codec_refresh_name(struct hda_codec *codec, const char *name) +{ + char tmp[16]; + + kfree(codec->chip_name); + if (!name) { + sprintf(tmp, "ID %x", codec->vendor_id & 0xffff); + name = tmp; + } + codec->chip_name = kstrdup(name, GFP_KERNEL); + return codec->chip_name ? 0 : -ENOMEM; +} + +static int hda_codec_driver_probe(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + struct module *owner = dev->driver->owner; + int err; + + if (WARN_ON(!codec->preset)) + return -EINVAL; + + err = codec_refresh_name(codec, codec->preset->name); + if (err < 0) + goto error; + + if (!try_module_get(owner)) { + err = -EINVAL; + goto error; + } + + err = codec->preset->patch(codec); + if (err < 0) { + module_put(owner); + goto error; + } + + return 0; + + error: + codec->preset = NULL; + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + return err; +} + +static int hda_codec_driver_remove(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + + if (codec->patch_ops.free) + codec->patch_ops.free(codec); + codec->preset = NULL; + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + module_put(dev->driver->owner); + return 0; +} + +int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, + struct module *owner) +{ + drv->driver.name = name; + drv->driver.owner = owner; + drv->driver.bus = &snd_hda_bus_type; + drv->driver.probe = hda_codec_driver_probe; + drv->driver.remove = hda_codec_driver_remove; + /* TODO: PM and others */ + return driver_register(&drv->driver); +} +EXPORT_SYMBOL_GPL(__hda_codec_driver_register); + +void hda_codec_driver_unregister(struct hda_codec_driver *drv) +{ + driver_unregister(&drv->driver); +} +EXPORT_SYMBOL_GPL(hda_codec_driver_unregister); + +static inline bool codec_probed(struct hda_codec *codec) +{ + return device_attach(hda_codec_dev(codec)) > 0 && codec->preset; +} + +/* try to auto-load and bind the codec module */ +static void codec_bind_module(struct hda_codec *codec) +{ +#ifdef MODULE + request_module("snd-hda-codec-id:%08x", codec->vendor_id); + if (codec_probed(codec)) + return; + request_module("snd-hda-codec-id:%04x*", + (codec->vendor_id >> 16) & 0xffff); + if (codec_probed(codec)) + return; +#endif +} + +/* store the codec vendor name */ +static int get_codec_vendor_name(struct hda_codec *codec) +{ + const struct hda_vendor_id *c; + const char *vendor = NULL; + u16 vendor_id = codec->vendor_id >> 16; + char tmp[16]; + + for (c = hda_vendor_ids; c->id; c++) { + if (c->id == vendor_id) { + vendor = c->name; + break; + } + } + if (!vendor) { + sprintf(tmp, "Generic %04x", vendor_id); + vendor = tmp; + } + codec->vendor_name = kstrdup(vendor, GFP_KERNEL); + if (!codec->vendor_name) + return -ENOMEM; + return 0; +} + +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) +/* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */ +static bool is_likely_hdmi_codec(struct hda_codec *codec) +{ + hda_nid_t nid = codec->start_nid; + int i; + + for (i = 0; i < codec->num_nodes; i++, nid++) { + unsigned int wcaps = get_wcaps(codec, nid); + switch (get_wcaps_type(wcaps)) { + case AC_WID_AUD_IN: + return false; /* HDMI parser supports only HDMI out */ + case AC_WID_AUD_OUT: + if (!(wcaps & AC_WCAP_DIGITAL)) + return false; + break; + } + } + return true; +} +#else +/* no HDMI codec parser support */ +#define is_likely_hdmi_codec(codec) false +#endif /* CONFIG_SND_HDA_CODEC_HDMI */ + +static int codec_bind_generic(struct hda_codec *codec) +{ + if (codec->probe_id) + return -ENODEV; + + if (is_likely_hdmi_codec(codec)) { + codec->probe_id = HDA_CODEC_ID_GENERIC_HDMI; +#if IS_MODULE(CONFIG_SND_HDA_CODEC_HDMI) + request_module("snd-hda-codec-hdmi"); +#endif + if (codec_probed(codec)) + return 0; + } + + codec->probe_id = HDA_CODEC_ID_GENERIC; +#if IS_MODULE(CONFIG_SND_HDA_GENERIC) + request_module("snd-hda-codec-generic"); +#endif + if (codec_probed(codec)) + return 0; + return -ENODEV; +} + +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) +#define is_generic_config(codec) \ + (codec->modelname && !strcmp(codec->modelname, "generic")) +#else +#define is_generic_config(codec) 0 +#endif + +/** + * snd_hda_codec_configure - (Re-)configure the HD-audio codec + * @codec: the HDA codec + * + * Start parsing of the given codec tree and (re-)initialize the whole + * patch instance. + * + * Returns 0 if successful or a negative error code. + */ +int snd_hda_codec_configure(struct hda_codec *codec) +{ + int err; + + if (!codec->vendor_name) { + err = get_codec_vendor_name(codec); + if (err < 0) + return err; + } + + if (is_generic_config(codec)) + codec->probe_id = HDA_CODEC_ID_GENERIC; + else + codec->probe_id = 0; + + err = device_add(hda_codec_dev(codec)); + if (err < 0) + return err; + + if (!codec->preset) + codec_bind_module(codec); + if (!codec->preset) { + err = codec_bind_generic(codec); + if (err < 0) { + codec_err(codec, "Unable to bind the codec\n"); + goto error; + } + } + + /* audio codec should override the mixer name */ + if (codec->afg || !*codec->bus->card->mixername) + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); + return 0; + + error: + device_del(hda_codec_dev(codec)); + return err; +} +EXPORT_SYMBOL_GPL(snd_hda_codec_configure); + +/* + * bus registration + */ +struct bus_type snd_hda_bus_type = { + .name = "hdaudio", + .match = hda_bus_match, +}; + +static int __init hda_codec_init(void) +{ + return bus_register(&snd_hda_bus_type); +} + +static void __exit hda_codec_exit(void) +{ + bus_unregister(&snd_hda_bus_type); +} + +module_init(hda_codec_init); +module_exit(hda_codec_exit); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5e755eb6f19a..61c8174e8013 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -40,69 +40,6 @@ #define CREATE_TRACE_POINTS #include "hda_trace.h" -/* - * vendor / preset table - */ - -struct hda_vendor_id { - unsigned int id; - const char *name; -}; - -/* codec vendor labels */ -static struct hda_vendor_id hda_vendor_ids[] = { - { 0x1002, "ATI" }, - { 0x1013, "Cirrus Logic" }, - { 0x1057, "Motorola" }, - { 0x1095, "Silicon Image" }, - { 0x10de, "Nvidia" }, - { 0x10ec, "Realtek" }, - { 0x1102, "Creative" }, - { 0x1106, "VIA" }, - { 0x111d, "IDT" }, - { 0x11c1, "LSI" }, - { 0x11d4, "Analog Devices" }, - { 0x13f6, "C-Media" }, - { 0x14f1, "Conexant" }, - { 0x17e8, "Chrontel" }, - { 0x1854, "LG" }, - { 0x1aec, "Wolfson Microelectronics" }, - { 0x1af4, "QEMU" }, - { 0x434d, "C-Media" }, - { 0x8086, "Intel" }, - { 0x8384, "SigmaTel" }, - {} /* terminator */ -}; - -static DEFINE_MUTEX(preset_mutex); -static LIST_HEAD(hda_preset_tables); - -/** - * snd_hda_add_codec_preset - Add a codec preset to the chain - * @preset: codec preset table to add - */ -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset) -{ - mutex_lock(&preset_mutex); - list_add_tail(&preset->list, &hda_preset_tables); - mutex_unlock(&preset_mutex); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_add_codec_preset); - -/** - * snd_hda_delete_codec_preset - Delete a codec preset from the chain - * @preset: codec preset table to delete - */ -int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset) -{ - mutex_lock(&preset_mutex); - list_del(&preset->list); - mutex_unlock(&preset_mutex); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_delete_codec_preset); - #ifdef CONFIG_PM #define codec_in_pm(codec) ((codec)->in_pm) static void hda_power_work(struct work_struct *work); @@ -885,111 +822,6 @@ int snd_hda_bus_new(struct snd_card *card, } EXPORT_SYMBOL_GPL(snd_hda_bus_new); -#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) -#define is_generic_config(codec) \ - (codec->modelname && !strcmp(codec->modelname, "generic")) -#else -#define is_generic_config(codec) 0 -#endif - -#ifdef MODULE -#define HDA_MODREQ_MAX_COUNT 2 /* two request_modules()'s */ -#else -#define HDA_MODREQ_MAX_COUNT 0 /* all presets are statically linked */ -#endif - -/* - * find a matching codec preset - */ -static const struct hda_codec_preset * -find_codec_preset(struct hda_codec *codec) -{ - struct hda_codec_preset_list *tbl; - const struct hda_codec_preset *preset; - unsigned int mod_requested = 0; - - again: - mutex_lock(&preset_mutex); - list_for_each_entry(tbl, &hda_preset_tables, list) { - if (!try_module_get(tbl->owner)) { - codec_err(codec, "cannot module_get\n"); - continue; - } - for (preset = tbl->preset; preset->id; preset++) { - u32 mask = preset->mask; - if (preset->afg && preset->afg != codec->afg) - continue; - if (preset->mfg && preset->mfg != codec->mfg) - continue; - if (!mask) - mask = ~0; - if (preset->id == (codec->vendor_id & mask) && - (!preset->rev || - preset->rev == codec->revision_id)) { - mutex_unlock(&preset_mutex); - codec->owner = tbl->owner; - return preset; - } - } - module_put(tbl->owner); - } - mutex_unlock(&preset_mutex); - - if (mod_requested < HDA_MODREQ_MAX_COUNT) { - if (!mod_requested) - request_module("snd-hda-codec-id:%08x", - codec->vendor_id); - else - request_module("snd-hda-codec-id:%04x*", - (codec->vendor_id >> 16) & 0xffff); - mod_requested++; - goto again; - } - return NULL; -} - -/* - * get_codec_name - store the codec name - */ -static int get_codec_name(struct hda_codec *codec) -{ - const struct hda_vendor_id *c; - const char *vendor = NULL; - u16 vendor_id = codec->vendor_id >> 16; - char tmp[16]; - - if (codec->vendor_name) - goto get_chip_name; - - for (c = hda_vendor_ids; c->id; c++) { - if (c->id == vendor_id) { - vendor = c->name; - break; - } - } - if (!vendor) { - sprintf(tmp, "Generic %04x", vendor_id); - vendor = tmp; - } - codec->vendor_name = kstrdup(vendor, GFP_KERNEL); - if (!codec->vendor_name) - return -ENOMEM; - - get_chip_name: - if (codec->chip_name) - return 0; - - if (codec->preset && codec->preset->name) - codec->chip_name = kstrdup(codec->preset->name, GFP_KERNEL); - else { - sprintf(tmp, "ID %x", codec->vendor_id & 0xffff); - codec->chip_name = kstrdup(tmp, GFP_KERNEL); - } - if (!codec->chip_name) - return -ENOMEM; - return 0; -} - /* * look for an AFG and MFG nodes */ @@ -1300,20 +1132,6 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) return p; } -/* - * Dynamic symbol binding for the codec parsers - */ - -#define load_parser(codec, sym) \ - ((codec)->parser = (int (*)(struct hda_codec *))symbol_request(sym)) - -static void unload_parser(struct hda_codec *codec) -{ - if (codec->parser) - symbol_put_addr(codec->parser); - codec->parser = NULL; -} - /* * codec destructor */ @@ -1322,6 +1140,8 @@ static void snd_hda_codec_free(struct hda_codec *codec) if (!codec) return; cancel_delayed_work_sync(&codec->jackpoll_work); + if (device_is_registered(hda_codec_dev(codec))) + device_del(hda_codec_dev(codec)); snd_hda_jack_tbl_clear(codec); free_init_pincfgs(codec); #ifdef CONFIG_PM @@ -1335,12 +1155,8 @@ static void snd_hda_codec_free(struct hda_codec *codec) snd_array_free(&codec->spdif_out); remove_conn_list(codec); codec->bus->caddr_tbl[codec->addr] = NULL; - if (codec->patch_ops.free) - codec->patch_ops.free(codec); hda_call_pm_notify(codec, false); /* cancel leftover refcounts */ snd_hda_sysfs_clear(codec); - unload_parser(codec); - module_put(codec->owner); free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); kfree(codec->vendor_name); @@ -1348,7 +1164,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec->modelname); kfree(codec->wcaps); codec->bus->num_codecs--; - put_device(&codec->dev); + put_device(hda_codec_dev(codec)); } static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, @@ -1360,10 +1176,7 @@ static unsigned int hda_set_power_state(struct hda_codec *codec, static int snd_hda_codec_dev_register(struct snd_device *device) { struct hda_codec *codec = device->device_data; - int err = device_add(&codec->dev); - if (err < 0) - return err; snd_hda_register_beep_device(codec); return 0; } @@ -1373,7 +1186,6 @@ static int snd_hda_codec_dev_disconnect(struct snd_device *device) struct hda_codec *codec = device->device_data; snd_hda_detach_beep_device(codec); - device_del(&codec->dev); return 0; } @@ -1386,7 +1198,7 @@ static int snd_hda_codec_dev_free(struct snd_device *device) /* just free the container */ static void snd_hda_codec_dev_release(struct device *dev) { - kfree(container_of(dev, struct hda_codec, dev)); + kfree(dev_to_hda_codec(dev)); } /** @@ -1402,6 +1214,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct hda_codec **codecp) { struct hda_codec *codec; + struct device *dev; char component[31]; hda_nid_t fg; int err; @@ -1429,14 +1242,14 @@ int snd_hda_codec_new(struct hda_bus *bus, return -ENOMEM; } - device_initialize(&codec->dev); - codec->dev.parent = &bus->card->card_dev; - codec->dev.class = sound_class; - codec->dev.release = snd_hda_codec_dev_release; - codec->dev.groups = snd_hda_dev_attr_groups; - dev_set_name(&codec->dev, "hdaudioC%dD%d", bus->card->number, - codec_addr); - dev_set_drvdata(&codec->dev, codec); /* for sysfs */ + dev = hda_codec_dev(codec); + device_initialize(dev); + dev->parent = bus->card->dev; + dev->bus = &snd_hda_bus_type; + dev->release = snd_hda_codec_dev_release; + dev->groups = snd_hda_dev_attr_groups; + dev_set_name(dev, "hdaudioC%dD%d", bus->card->number, codec_addr); + dev_set_drvdata(dev, codec); /* for sysfs */ codec->bus = bus; codec->addr = codec_addr; @@ -1587,92 +1400,6 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_codec_update_widgets); - -#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) -/* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */ -static bool is_likely_hdmi_codec(struct hda_codec *codec) -{ - hda_nid_t nid = codec->start_nid; - int i; - - for (i = 0; i < codec->num_nodes; i++, nid++) { - unsigned int wcaps = get_wcaps(codec, nid); - switch (get_wcaps_type(wcaps)) { - case AC_WID_AUD_IN: - return false; /* HDMI parser supports only HDMI out */ - case AC_WID_AUD_OUT: - if (!(wcaps & AC_WCAP_DIGITAL)) - return false; - break; - } - } - return true; -} -#else -/* no HDMI codec parser support */ -#define is_likely_hdmi_codec(codec) false -#endif /* CONFIG_SND_HDA_CODEC_HDMI */ - -/** - * snd_hda_codec_configure - (Re-)configure the HD-audio codec - * @codec: the HDA codec - * - * Start parsing of the given codec tree and (re-)initialize the whole - * patch instance. - * - * Returns 0 if successful or a negative error code. - */ -int snd_hda_codec_configure(struct hda_codec *codec) -{ - int (*patch)(struct hda_codec *) = NULL; - int err; - - codec->preset = find_codec_preset(codec); - if (!codec->vendor_name || !codec->chip_name) { - err = get_codec_name(codec); - if (err < 0) - return err; - } - - if (!is_generic_config(codec) && codec->preset) - patch = codec->preset->patch; - if (!patch) { - unload_parser(codec); /* to be sure */ - if (is_likely_hdmi_codec(codec)) { -#if IS_MODULE(CONFIG_SND_HDA_CODEC_HDMI) - patch = load_parser(codec, snd_hda_parse_hdmi_codec); -#elif IS_BUILTIN(CONFIG_SND_HDA_CODEC_HDMI) - patch = snd_hda_parse_hdmi_codec; -#endif - } - if (!patch) { -#if IS_MODULE(CONFIG_SND_HDA_GENERIC) - patch = load_parser(codec, snd_hda_parse_generic_codec); -#elif IS_BUILTIN(CONFIG_SND_HDA_GENERIC) - patch = snd_hda_parse_generic_codec; -#endif - } - if (!patch) { - codec_err(codec, "No codec parser is available\n"); - return -ENODEV; - } - } - - err = patch(codec); - if (err < 0) { - unload_parser(codec); - return err; - } - - /* audio codec should override the mixer name */ - if (codec->afg || !*codec->bus->card->mixername) - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), - "%s %s", codec->vendor_name, codec->chip_name); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_codec_configure); - /* update the stream-id if changed */ static void update_pcm_stream_id(struct hda_codec *codec, struct hda_cvt_setup *p, hda_nid_t nid, @@ -2739,8 +2466,9 @@ int snd_hda_codec_reset(struct hda_codec *codec) } } snd_hda_detach_beep_device(codec); - if (codec->patch_ops.free) - codec->patch_ops.free(codec); + if (device_is_registered(hda_codec_dev(codec))) + device_del(hda_codec_dev(codec)); + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); snd_hda_jack_tbl_clear(codec); codec->proc_widget_hook = NULL; @@ -2759,9 +2487,6 @@ int snd_hda_codec_reset(struct hda_codec *codec) codec->preset = NULL; codec->slave_dig_outs = NULL; codec->spdif_status_reset = 0; - unload_parser(codec); - module_put(codec->owner); - codec->owner = NULL; /* allow device access again */ snd_hda_unlock_devices(bus); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 96421a3b32cd..3d42717e0e65 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -174,15 +174,22 @@ struct hda_codec_preset { int (*patch)(struct hda_codec *codec); }; -struct hda_codec_preset_list { +#define HDA_CODEC_ID_GENERIC_HDMI 0x00000101 +#define HDA_CODEC_ID_GENERIC 0x00000201 + +struct hda_codec_driver { + struct device_driver driver; const struct hda_codec_preset *preset; - struct module *owner; - struct list_head list; }; -/* initial hook */ -int snd_hda_add_codec_preset(struct hda_codec_preset_list *preset); -int snd_hda_delete_codec_preset(struct hda_codec_preset_list *preset); +int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, + struct module *owner); +#define hda_codec_driver_register(drv) \ + __hda_codec_driver_register(drv, KBUILD_MODNAME, THIS_MODULE) +void hda_codec_driver_unregister(struct hda_codec_driver *drv); +#define module_hda_codec_driver(drv) \ + module_driver(drv, hda_codec_driver_register, \ + hda_codec_driver_unregister) /* ops set by the preset patch */ struct hda_codec_ops { @@ -286,11 +293,10 @@ struct hda_codec { u32 vendor_id; u32 subsystem_id; u32 revision_id; + u32 probe_id; /* overridden id for probing */ /* detected preset */ const struct hda_codec_preset *preset; - struct module *owner; - int (*parser)(struct hda_codec *codec); const char *vendor_name; /* codec vendor name */ const char *chip_name; /* codec chip name */ const char *modelname; /* model name for preset */ @@ -408,6 +414,11 @@ struct hda_codec { struct snd_array verbs; }; +#define dev_to_hda_codec(_dev) container_of(_dev, struct hda_codec, dev) +#define hda_codec_dev(_dev) (&(_dev)->dev) + +extern struct bus_type snd_hda_bus_type; + /* direction */ enum { HDA_INPUT, HDA_OUTPUT diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b680b4ec6331..947d1a50f384 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5493,13 +5493,11 @@ static const struct hda_codec_ops generic_patch_ops = { #endif }; -/** +/* * snd_hda_parse_generic_codec - Generic codec parser * @codec: the HDA codec - * - * This should be called from the HDA codec core. */ -int snd_hda_parse_generic_codec(struct hda_codec *codec) +static int snd_hda_parse_generic_codec(struct hda_codec *codec) { struct hda_gen_spec *spec; int err; @@ -5525,7 +5523,17 @@ error: snd_hda_gen_free(codec); return err; } -EXPORT_SYMBOL_GPL(snd_hda_parse_generic_codec); + +static const struct hda_codec_preset snd_hda_preset_generic[] = { + { .id = HDA_CODEC_ID_GENERIC, .patch = snd_hda_parse_generic_codec }, + {} /* terminator */ +}; + +static struct hda_codec_driver generic_driver = { + .preset = snd_hda_preset_generic, +}; + +module_hda_codec_driver(generic_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Generic HD-audio codec parser"); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 49c08a7278c4..2f7d9646a41d 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -350,12 +350,6 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout); -/* - * generic codec parser - */ -int snd_hda_parse_generic_codec(struct hda_codec *codec); -int snd_hda_parse_hdmi_codec(struct hda_codec *codec); - /* * generic proc interface */ @@ -783,9 +777,13 @@ void snd_print_channel_allocation(int spk_alloc, char *buf, int buflen); /* */ -#define codec_err(codec, fmt, args...) dev_err(&(codec)->dev, fmt, ##args) -#define codec_warn(codec, fmt, args...) dev_warn(&(codec)->dev, fmt, ##args) -#define codec_info(codec, fmt, args...) dev_info(&(codec)->dev, fmt, ##args) -#define codec_dbg(codec, fmt, args...) dev_dbg(&(codec)->dev, fmt, ##args) +#define codec_err(codec, fmt, args...) \ + dev_err(hda_codec_dev(codec), fmt, ##args) +#define codec_warn(codec, fmt, args...) \ + dev_warn(hda_codec_dev(codec), fmt, ##args) +#define codec_info(codec, fmt, args...) \ + dev_info(hda_codec_dev(codec), fmt, ##args) +#define codec_dbg(codec, fmt, args...) \ + dev_dbg(hda_codec_dev(codec), fmt, ##args) #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d285904cdb64..af4c7be86c27 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1194,20 +1194,8 @@ MODULE_ALIAS("snd-hda-codec-id:11d4*"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Analog Devices HD-audio codec"); -static struct hda_codec_preset_list analog_list = { +static struct hda_codec_driver analog_driver = { .preset = snd_hda_preset_analog, - .owner = THIS_MODULE, }; -static int __init patch_analog_init(void) -{ - return snd_hda_add_codec_preset(&analog_list); -} - -static void __exit patch_analog_exit(void) -{ - snd_hda_delete_codec_preset(&analog_list); -} - -module_init(patch_analog_init) -module_exit(patch_analog_exit) +module_hda_codec_driver(analog_driver); diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 5e65999e0d8e..447302695195 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -98,20 +98,8 @@ MODULE_ALIAS("snd-hda-codec-id:1102000d"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Creative CA0110-IBG HD-audio codec"); -static struct hda_codec_preset_list ca0110_list = { +static struct hda_codec_driver ca0110_driver = { .preset = snd_hda_preset_ca0110, - .owner = THIS_MODULE, }; -static int __init patch_ca0110_init(void) -{ - return snd_hda_add_codec_preset(&ca0110_list); -} - -static void __exit patch_ca0110_exit(void) -{ - snd_hda_delete_codec_preset(&ca0110_list); -} - -module_init(patch_ca0110_init) -module_exit(patch_ca0110_exit) +module_hda_codec_driver(ca0110_driver); diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e0383eea9880..81991b4134cd 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4702,20 +4702,8 @@ MODULE_ALIAS("snd-hda-codec-id:11020011"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Creative Sound Core3D codec"); -static struct hda_codec_preset_list ca0132_list = { +static struct hda_codec_driver ca0132_driver = { .preset = snd_hda_preset_ca0132, - .owner = THIS_MODULE, }; -static int __init patch_ca0132_init(void) -{ - return snd_hda_add_codec_preset(&ca0132_list); -} - -static void __exit patch_ca0132_exit(void) -{ - snd_hda_delete_codec_preset(&ca0132_list); -} - -module_init(patch_ca0132_init) -module_exit(patch_ca0132_exit) +module_hda_codec_driver(ca0132_driver); diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 1589c9bcce3e..1af133933bc0 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1219,20 +1219,8 @@ MODULE_ALIAS("snd-hda-codec-id:10134213"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); -static struct hda_codec_preset_list cirrus_list = { +static struct hda_codec_driver cirrus_driver = { .preset = snd_hda_preset_cirrus, - .owner = THIS_MODULE, }; -static int __init patch_cirrus_init(void) -{ - return snd_hda_add_codec_preset(&cirrus_list); -} - -static void __exit patch_cirrus_exit(void) -{ - snd_hda_delete_codec_preset(&cirrus_list); -} - -module_init(patch_cirrus_init) -module_exit(patch_cirrus_exit) +module_hda_codec_driver(cirrus_driver); diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index c895a8f21192..617d9012e78a 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -137,20 +137,8 @@ MODULE_ALIAS("snd-hda-codec-id:434d4980"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("C-Media HD-audio codec"); -static struct hda_codec_preset_list cmedia_list = { +static struct hda_codec_driver cmedia_driver = { .preset = snd_hda_preset_cmedia, - .owner = THIS_MODULE, }; -static int __init patch_cmedia_init(void) -{ - return snd_hda_add_codec_preset(&cmedia_list); -} - -static void __exit patch_cmedia_exit(void) -{ - snd_hda_delete_codec_preset(&cmedia_list); -} - -module_init(patch_cmedia_init) -module_exit(patch_cmedia_exit) +module_hda_codec_driver(cmedia_driver); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index fd3ed18670e9..15a0a7d38c35 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1007,20 +1007,8 @@ MODULE_ALIAS("snd-hda-codec-id:14f151d7"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); -static struct hda_codec_preset_list conexant_list = { +static struct hda_codec_driver conexant_driver = { .preset = snd_hda_preset_conexant, - .owner = THIS_MODULE, }; -static int __init patch_conexant_init(void) -{ - return snd_hda_add_codec_preset(&conexant_list); -} - -static void __exit patch_conexant_exit(void) -{ - snd_hda_delete_codec_preset(&conexant_list); -} - -module_init(patch_conexant_init) -module_exit(patch_conexant_exit) +module_hda_codec_driver(conexant_driver); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b422e406a9cb..f1812aabd63e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3300,15 +3300,6 @@ static int patch_via_hdmi(struct hda_codec *codec) return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID); } -/* - * called from hda_codec.c for generic HDMI support - */ -int snd_hda_parse_hdmi_codec(struct hda_codec *codec) -{ - return patch_generic_hdmi(codec); -} -EXPORT_SYMBOL_GPL(snd_hda_parse_hdmi_codec); - /* * patch entries */ @@ -3373,6 +3364,8 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, +/* special ID for generic HDMI */ +{ .id = HDA_CODEC_ID_GENERIC_HDMI, .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -3442,20 +3435,8 @@ MODULE_ALIAS("snd-hda-codec-intelhdmi"); MODULE_ALIAS("snd-hda-codec-nvhdmi"); MODULE_ALIAS("snd-hda-codec-atihdmi"); -static struct hda_codec_preset_list intel_list = { +static struct hda_codec_driver hdmi_driver = { .preset = snd_hda_preset_hdmi, - .owner = THIS_MODULE, }; -static int __init patch_hdmi_init(void) -{ - return snd_hda_add_codec_preset(&intel_list); -} - -static void __exit patch_hdmi_exit(void) -{ - snd_hda_delete_codec_preset(&intel_list); -} - -module_init(patch_hdmi_init) -module_exit(patch_hdmi_exit) +module_hda_codec_driver(hdmi_driver); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b2b24a8b3dac..70808f92276a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6514,20 +6514,8 @@ MODULE_ALIAS("snd-hda-codec-id:10ec*"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Realtek HD-audio codec"); -static struct hda_codec_preset_list realtek_list = { +static struct hda_codec_driver realtek_driver = { .preset = snd_hda_preset_realtek, - .owner = THIS_MODULE, }; -static int __init patch_realtek_init(void) -{ - return snd_hda_add_codec_preset(&realtek_list); -} - -static void __exit patch_realtek_exit(void) -{ - snd_hda_delete_codec_preset(&realtek_list); -} - -module_init(patch_realtek_init) -module_exit(patch_realtek_exit) +module_hda_codec_driver(realtek_driver); diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 3208ad69583e..38a477333321 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -319,20 +319,8 @@ MODULE_ALIAS("snd-hda-codec-id:18540018"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Si3054 HD-audio modem codec"); -static struct hda_codec_preset_list si3054_list = { +static struct hda_codec_driver si3054_driver = { .preset = snd_hda_preset_si3054, - .owner = THIS_MODULE, }; -static int __init patch_si3054_init(void) -{ - return snd_hda_add_codec_preset(&si3054_list); -} - -static void __exit patch_si3054_exit(void) -{ - snd_hda_delete_codec_preset(&si3054_list); -} - -module_init(patch_si3054_init) -module_exit(patch_si3054_exit) +module_hda_codec_driver(si3054_driver); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 87eff3173ce9..6a2163056216 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5091,20 +5091,8 @@ MODULE_ALIAS("snd-hda-codec-id:111d*"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("IDT/Sigmatel HD-audio codec"); -static struct hda_codec_preset_list sigmatel_list = { +static struct hda_codec_driver sigmatel_driver = { .preset = snd_hda_preset_sigmatel, - .owner = THIS_MODULE, }; -static int __init patch_sigmatel_init(void) -{ - return snd_hda_add_codec_preset(&sigmatel_list); -} - -static void __exit patch_sigmatel_exit(void) -{ - snd_hda_delete_codec_preset(&sigmatel_list); -} - -module_init(patch_sigmatel_init) -module_exit(patch_sigmatel_exit) +module_hda_codec_driver(sigmatel_driver); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 3de6d3d779c9..2045f33b1ace 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1884,23 +1884,11 @@ static const struct hda_codec_preset snd_hda_preset_via[] = { MODULE_ALIAS("snd-hda-codec-id:1106*"); -static struct hda_codec_preset_list via_list = { +static struct hda_codec_driver via_driver = { .preset = snd_hda_preset_via, - .owner = THIS_MODULE, }; MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("VIA HD-audio codec"); -static int __init patch_via_init(void) -{ - return snd_hda_add_codec_preset(&via_list); -} - -static void __exit patch_via_exit(void) -{ - snd_hda_delete_codec_preset(&via_list); -} - -module_init(patch_via_init) -module_exit(patch_via_exit) +module_hda_codec_driver(via_driver); From 59ed1eade1d6ec24751baca99305f9713a5d779e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Feb 2015 15:39:59 +0100 Subject: [PATCH 042/162] ALSA: hda - Move codec suspend/resume to codec driver This patch moves the suspend/resume mechanisms down to each codec driver level, as we have a proper codec driver bound on the bus now. Then we get the asynchronous PM gratis without fiddling much in the driver level. As a soft-landing transition, implement the common suspend/resume pm ops for hda_codec_driver and keep the each codec driver intact. Only the callers of suspend/resume in the controller side (azx_suspend() and azx_resume()) are removed. Another involved place is azx_bus_reset() calling the temporary suspend and resume as a hackish method of bus reset. The HD-audio core provide a helper function snd_hda_bus_reset() instead. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 3 +- sound/pci/hda/hda_codec.c | 93 ++++++++++++---------------------- sound/pci/hda/hda_codec.h | 6 +-- sound/pci/hda/hda_controller.c | 11 +--- sound/pci/hda/hda_intel.c | 6 --- sound/pci/hda/hda_tegra.c | 6 --- 6 files changed, 39 insertions(+), 86 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index adf6b475dee1..ce2dd7b0dc07 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -8,6 +8,7 @@ #include #include #include +#include #include #include "hda_codec.h" #include "hda_local.h" @@ -138,7 +139,7 @@ int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, drv->driver.bus = &snd_hda_bus_type; drv->driver.probe = hda_codec_driver_probe; drv->driver.remove = hda_codec_driver_remove; - /* TODO: PM and others */ + drv->driver.pm = &hda_codec_driver_pm; return driver_register(&drv->driver); } EXPORT_SYMBOL_GPL(__hda_codec_driver_register); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 61c8174e8013..3d6ff60e6c14 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1250,6 +1250,7 @@ int snd_hda_codec_new(struct hda_bus *bus, dev->groups = snd_hda_dev_attr_groups; dev_set_name(dev, "hdaudioC%dD%d", bus->card->number, codec_addr); dev_set_drvdata(dev, codec); /* for sysfs */ + device_enable_async_suspend(dev); codec->bus = bus; codec->addr = codec_addr; @@ -3970,8 +3971,31 @@ static void hda_call_codec_resume(struct hda_codec *codec) codec->in_pm = 0; snd_hda_power_down(codec); /* flag down before returning */ } + +static int hda_codec_driver_suspend(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + int i; + + cancel_delayed_work_sync(&codec->jackpoll_work); + for (i = 0; i < codec->num_pcms; i++) + snd_pcm_suspend_all(codec->pcm_info[i].pcm); + hda_call_codec_suspend(codec, false); + return 0; +} + +static int hda_codec_driver_resume(struct device *dev) +{ + hda_call_codec_resume(dev_to_hda_codec(dev)); + return 0; +} #endif /* CONFIG_PM */ +/* referred in hda_bind.c */ +const struct dev_pm_ops hda_codec_driver_pm = { + SET_SYSTEM_SLEEP_PM_OPS(hda_codec_driver_suspend, + hda_codec_driver_resume) +}; /** * snd_hda_build_controls - build mixer controls @@ -5505,77 +5529,26 @@ int snd_hda_add_imux_item(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_add_imux_item); - -#ifdef CONFIG_PM -/* - * power management - */ - - -static void hda_async_suspend(void *data, async_cookie_t cookie) -{ - hda_call_codec_suspend(data, false); -} - -static void hda_async_resume(void *data, async_cookie_t cookie) -{ - hda_call_codec_resume(data); -} - /** - * snd_hda_suspend - suspend the codecs - * @bus: the HDA bus - * - * Returns 0 if successful. + * snd_hda_bus_reset - Reset the bus + * @bus: HD-audio bus */ -int snd_hda_suspend(struct hda_bus *bus) +void snd_hda_bus_reset(struct hda_bus *bus) { struct hda_codec *codec; - ASYNC_DOMAIN_EXCLUSIVE(domain); list_for_each_entry(codec, &bus->codec_list, list) { + /* FIXME: maybe a better way needed for forced reset */ cancel_delayed_work_sync(&codec->jackpoll_work); +#ifdef CONFIG_PM if (hda_codec_is_power_on(codec)) { - if (bus->num_codecs > 1) - async_schedule_domain(hda_async_suspend, codec, - &domain); - else - hda_call_codec_suspend(codec, false); - } - } - - if (bus->num_codecs > 1) - async_synchronize_full_domain(&domain); - - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_suspend); - -/** - * snd_hda_resume - resume the codecs - * @bus: the HDA bus - * - * Returns 0 if successful. - */ -int snd_hda_resume(struct hda_bus *bus) -{ - struct hda_codec *codec; - ASYNC_DOMAIN_EXCLUSIVE(domain); - - list_for_each_entry(codec, &bus->codec_list, list) { - if (bus->num_codecs > 1) - async_schedule_domain(hda_async_resume, codec, &domain); - else + hda_call_codec_suspend(codec, false); hda_call_codec_resume(codec); + } +#endif } - - if (bus->num_codecs > 1) - async_synchronize_full_domain(&domain); - - return 0; } -EXPORT_SYMBOL_GPL(snd_hda_resume); -#endif /* CONFIG_PM */ +EXPORT_SYMBOL_GPL(snd_hda_bus_reset); /* * generic arrays diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 3d42717e0e65..1fa3dd9baf52 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -567,14 +567,12 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, int snd_hda_lock_devices(struct hda_bus *bus); void snd_hda_unlock_devices(struct hda_bus *bus); +void snd_hda_bus_reset(struct hda_bus *bus); /* * power management */ -#ifdef CONFIG_PM -int snd_hda_suspend(struct hda_bus *bus); -int snd_hda_resume(struct hda_bus *bus); -#endif +extern const struct dev_pm_ops hda_codec_driver_pm; static inline int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 4c7a6f9bfcde..30ddb751806a 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1780,15 +1780,8 @@ static void azx_bus_reset(struct hda_bus *bus) bus->in_reset = 1; azx_stop_chip(chip); azx_init_chip(chip, true); -#ifdef CONFIG_PM - if (chip->initialized) { - struct azx_pcm *p; - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); - snd_hda_suspend(chip->bus); - snd_hda_resume(chip->bus); - } -#endif + if (chip->initialized) + snd_hda_bus_reset(chip->bus); bus->in_reset = 0; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5e00cc4a722f..9db1b078801f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -772,7 +772,6 @@ static int azx_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; struct hda_intel *hda; - struct azx_pcm *p; if (!card) return 0; @@ -784,10 +783,6 @@ static int azx_suspend(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); azx_clear_irq_pending(chip); - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); - if (chip->initialized) - snd_hda_suspend(chip->bus); azx_stop_chip(chip); azx_enter_link_reset(chip); if (chip->irq >= 0) { @@ -830,7 +825,6 @@ static int azx_resume(struct device *dev) azx_init_chip(chip, true); - snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 1bd7a9e04046..f6949e413a50 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -249,14 +249,9 @@ static int hda_tegra_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; - struct azx_pcm *p; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - list_for_each_entry(p, &chip->pcm_list, list) - snd_pcm_suspend_all(p->pcm); - if (chip->initialized) - snd_hda_suspend(chip->bus); azx_stop_chip(chip); azx_enter_link_reset(chip); @@ -277,7 +272,6 @@ static int hda_tegra_resume(struct device *dev) azx_init_chip(chip, 1); - snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; From cc72da7d4d063ab9e690e56e0ef1ca1c24ee1635 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 16:00:22 +0100 Subject: [PATCH 043/162] ALSA: hda - Use standard runtime PM for codec power-save control Like the previous transition of suspend/resume, now move the power-save code to the standard runtime PM. As usual for runtime PM, it's a bit tricky, but this simplified codes a lot in the end. For keeping the usage compatibility, power_save module option still controls the whole power-saving behavior on all codecs. The value is translated to pm_runtime_*_autosuspend() and pm_runtime_allow() / pm_runtime_forbid() calls. snd_hda_power_up() and snd_hda_power_down() are translated to pm_runtime_get_sync() and pm_runtime_put_autosuspend(), respectively. Since we can do call pm_runtime_get_sync() more reliably, the sync version is used always and snd_hda_power_up_d3wait() is dropped. Another slight difference is that snd_hda_power_up()/down() don't call runtime_pm code during the suspend/resume transition phase. Calling them there isn't safe unlike our own code, resulted in unexpected behavior (endless wakeups). The hda_power_count tracepoint was removed, as it doesn't match well with the new code. Last but not least, we need to set ignore_children flag in the parent dev.power field so that the runtime PM of the controller chip won't get confused. The notification is still done in the bus pm_notify callback. We'll get rid of this hack in the later patch. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 251 +++++++++++++-------------------- sound/pci/hda/hda_codec.h | 67 +-------- sound/pci/hda/hda_controller.c | 2 +- sound/pci/hda/hda_intel.c | 2 + sound/pci/hda/hda_trace.h | 24 ---- 5 files changed, 111 insertions(+), 235 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3d6ff60e6c14..d0dbc62c9147 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -26,6 +26,8 @@ #include #include #include +#include +#include #include #include "hda_codec.h" #include @@ -41,10 +43,9 @@ #include "hda_trace.h" #ifdef CONFIG_PM -#define codec_in_pm(codec) ((codec)->in_pm) -static void hda_power_work(struct work_struct *work); -static void hda_keep_power_on(struct hda_codec *codec); -#define hda_codec_is_power_on(codec) ((codec)->power_on) +#define codec_in_pm(codec) atomic_read(&(codec)->in_pm) +#define hda_codec_is_power_on(codec) \ + (!pm_runtime_suspended(hda_codec_dev(codec))) static void hda_call_pm_notify(struct hda_codec *codec, bool power_up) { @@ -60,7 +61,6 @@ static void hda_call_pm_notify(struct hda_codec *codec, bool power_up) #else #define codec_in_pm(codec) 0 -static inline void hda_keep_power_on(struct hda_codec *codec) {} #define hda_codec_is_power_on(codec) 1 #define hda_call_pm_notify(codec, state) {} #endif @@ -1144,10 +1144,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) device_del(hda_codec_dev(codec)); snd_hda_jack_tbl_clear(codec); free_init_pincfgs(codec); -#ifdef CONFIG_PM - cancel_delayed_work(&codec->power_work); flush_workqueue(codec->bus->workq); -#endif list_del(&codec->list); snd_array_free(&codec->mixers); snd_array_free(&codec->nids); @@ -1178,6 +1175,10 @@ static int snd_hda_codec_dev_register(struct snd_device *device) struct hda_codec *codec = device->device_data; snd_hda_register_beep_device(codec); + if (device_is_registered(hda_codec_dev(codec))) { + snd_hda_power_sync(codec); + pm_runtime_enable(hda_codec_dev(codec)); + } return 0; } @@ -1274,13 +1275,14 @@ int snd_hda_codec_new(struct hda_bus *bus, codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM - spin_lock_init(&codec->power_lock); - INIT_DELAYED_WORK(&codec->power_work, hda_power_work); /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. * the caller has to power down appropriatley after initialization * phase. */ - hda_keep_power_on(codec); + pm_runtime_set_active(hda_codec_dev(codec)); + pm_runtime_get_noresume(hda_codec_dev(codec)); + codec->power_jiffies = jiffies; + hda_call_pm_notify(codec, true); #endif snd_hda_sysfs_init(codec); @@ -2453,10 +2455,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) /* OK, let it free */ cancel_delayed_work_sync(&codec->jackpoll_work); -#ifdef CONFIG_PM - cancel_delayed_work_sync(&codec->power_work); flush_workqueue(bus->workq); -#endif snd_hda_ctls_clear(codec); /* release PCMs */ for (i = 0; i < codec->num_pcms; i++) { @@ -3893,31 +3892,40 @@ static inline void hda_exec_init_verbs(struct hda_codec *codec) {} #endif #ifdef CONFIG_PM +/* update the power on/off account with the current jiffies */ +static void update_power_acct(struct hda_codec *codec, bool on) +{ + unsigned long delta = jiffies - codec->power_jiffies; + + if (on) + codec->power_on_acct += delta; + else + codec->power_off_acct += delta; + codec->power_jiffies += delta; +} + +void snd_hda_update_power_acct(struct hda_codec *codec) +{ + update_power_acct(codec, hda_codec_is_power_on(codec)); +} + /* * call suspend and power-down; used both from PM and power-save * this function returns the power state in the end */ -static unsigned int hda_call_codec_suspend(struct hda_codec *codec, bool in_wq) +static unsigned int hda_call_codec_suspend(struct hda_codec *codec) { unsigned int state; - codec->in_pm = 1; + atomic_inc(&codec->in_pm); if (codec->patch_ops.suspend) codec->patch_ops.suspend(codec); hda_cleanup_all_streams(codec); state = hda_set_power_state(codec, AC_PWRST_D3); - /* Cancel delayed work if we aren't currently running from it. */ - if (!in_wq) - cancel_delayed_work_sync(&codec->power_work); - spin_lock(&codec->power_lock); - snd_hda_update_power_acct(codec); trace_hda_power_down(codec); - codec->power_on = 0; - codec->power_transition = 0; - codec->power_jiffies = jiffies; - spin_unlock(&codec->power_lock); - codec->in_pm = 0; + update_power_acct(codec, true); + atomic_dec(&codec->in_pm); return state; } @@ -3942,14 +3950,14 @@ static void hda_mark_cmd_cache_dirty(struct hda_codec *codec) */ static void hda_call_codec_resume(struct hda_codec *codec) { - codec->in_pm = 1; + atomic_inc(&codec->in_pm); + trace_hda_power_up(codec); hda_mark_cmd_cache_dirty(codec); - /* set as if powered on for avoiding re-entering the resume - * in the resume / power-save sequence - */ - hda_keep_power_on(codec); + codec->power_jiffies = jiffies; + hda_call_pm_notify(codec, true); + hda_set_power_state(codec, AC_PWRST_D0); restore_shutup_pins(codec); hda_exec_init_verbs(codec); @@ -3967,34 +3975,38 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_jackpoll_work(&codec->jackpoll_work.work); else snd_hda_jack_report_sync(codec); - - codec->in_pm = 0; - snd_hda_power_down(codec); /* flag down before returning */ + atomic_dec(&codec->in_pm); } -static int hda_codec_driver_suspend(struct device *dev) +static int hda_codec_runtime_suspend(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + unsigned int state; int i; cancel_delayed_work_sync(&codec->jackpoll_work); for (i = 0; i < codec->num_pcms; i++) snd_pcm_suspend_all(codec->pcm_info[i].pcm); - hda_call_codec_suspend(codec, false); + state = hda_call_codec_suspend(codec); + if (!codec->bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) + hda_call_pm_notify(codec, false); return 0; } -static int hda_codec_driver_resume(struct device *dev) +static int hda_codec_runtime_resume(struct device *dev) { hda_call_codec_resume(dev_to_hda_codec(dev)); + pm_runtime_mark_last_busy(dev); return 0; } #endif /* CONFIG_PM */ /* referred in hda_bind.c */ const struct dev_pm_ops hda_codec_driver_pm = { - SET_SYSTEM_SLEEP_PM_OPS(hda_codec_driver_suspend, - hda_codec_driver_resume) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) + SET_RUNTIME_PM_OPS(hda_codec_runtime_suspend, hda_codec_runtime_resume, + NULL) }; /** @@ -4733,127 +4745,66 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_add_new_ctls); #ifdef CONFIG_PM -static void hda_power_work(struct work_struct *work) +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the usage counter and resume the device if not done yet. + */ +void snd_hda_power_up(struct hda_codec *codec) { - struct hda_codec *codec = - container_of(work, struct hda_codec, power_work.work); - struct hda_bus *bus = codec->bus; - unsigned int state; + struct device *dev = hda_codec_dev(codec); - spin_lock(&codec->power_lock); - if (codec->power_transition > 0) { /* during power-up sequence? */ - spin_unlock(&codec->power_lock); + if (codec_in_pm(codec)) return; - } - if (!codec->power_on || codec->power_count) { - codec->power_transition = 0; - spin_unlock(&codec->power_lock); - return; - } - spin_unlock(&codec->power_lock); - - state = hda_call_codec_suspend(codec, true); - if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) - hda_call_pm_notify(codec, false); -} - -static void hda_keep_power_on(struct hda_codec *codec) -{ - spin_lock(&codec->power_lock); - codec->power_count++; - codec->power_on = 1; - codec->power_jiffies = jiffies; - spin_unlock(&codec->power_lock); - hda_call_pm_notify(codec, true); -} - -/* update the power on/off account with the current jiffies */ -void snd_hda_update_power_acct(struct hda_codec *codec) -{ - unsigned long delta = jiffies - codec->power_jiffies; - if (codec->power_on) - codec->power_on_acct += delta; - else - codec->power_off_acct += delta; - codec->power_jiffies += delta; -} - -/* Transition to powered up, if wait_power_down then wait for a pending - * transition to D3 to complete. A pending D3 transition is indicated - * with power_transition == -1. */ -/* call this with codec->power_lock held! */ -static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) -{ - /* Return if power_on or transitioning to power_on, unless currently - * powering down. */ - if ((codec->power_on || codec->power_transition > 0) && - !(wait_power_down && codec->power_transition < 0)) - return; - spin_unlock(&codec->power_lock); - - cancel_delayed_work_sync(&codec->power_work); - - spin_lock(&codec->power_lock); - /* If the power down delayed work was cancelled above before starting, - * then there is no need to go through power up here. - */ - if (codec->power_on) { - if (codec->power_transition < 0) - codec->power_transition = 0; - return; - } - - trace_hda_power_up(codec); - snd_hda_update_power_acct(codec); - codec->power_on = 1; - codec->power_jiffies = jiffies; - codec->power_transition = 1; /* avoid reentrance */ - spin_unlock(&codec->power_lock); - - hda_call_codec_resume(codec); - - spin_lock(&codec->power_lock); - codec->power_transition = 0; -} - -#define power_save(codec) \ - ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) - -/* Transition to powered down */ -static void __snd_hda_power_down(struct hda_codec *codec) -{ - if (!codec->power_on || codec->power_count || codec->power_transition) - return; - - if (power_save(codec)) { - codec->power_transition = -1; /* avoid reentrance */ - queue_delayed_work(codec->bus->workq, &codec->power_work, - msecs_to_jiffies(power_save(codec) * 1000)); - } + pm_runtime_get_sync(dev); } +EXPORT_SYMBOL_GPL(snd_hda_power_up); /** - * snd_hda_power_save - Power-up/down/sync the codec + * snd_hda_power_down - Power-down the codec * @codec: HD-audio codec - * @delta: the counter delta to change - * @d3wait: sync for D3 transition complete * - * Change the power-up counter via @delta, and power up or down the hardware - * appropriately. For the power-down, queue to the delayed action. - * Passing zero to @delta means to synchronize the power state. + * Decrement the usage counter and schedules the autosuspend if none used. */ -void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait) +void snd_hda_power_down(struct hda_codec *codec) { - spin_lock(&codec->power_lock); - codec->power_count += delta; - trace_hda_power_count(codec); - if (delta > 0) - __snd_hda_power_up(codec, d3wait); - else - __snd_hda_power_down(codec); - spin_unlock(&codec->power_lock); + struct device *dev = hda_codec_dev(codec); + + if (codec_in_pm(codec)) + return; + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); } -EXPORT_SYMBOL_GPL(snd_hda_power_save); +EXPORT_SYMBOL_GPL(snd_hda_power_down); + +/** + * snd_hda_power_sync - Synchronize the power_save option + * @codec: HD-audio codec + * + * Synchronize the runtime PM autosuspend state from the power_save option. + */ +void snd_hda_power_sync(struct hda_codec *codec) +{ + struct device *dev = hda_codec_dev(codec); + int delay; + + if (!codec->bus->power_save) + return; + + delay = *codec->bus->power_save * 1000; + if (delay > 0) { + pm_runtime_set_autosuspend_delay(dev, delay); + pm_runtime_use_autosuspend(dev); + pm_runtime_allow(dev); + if (!pm_runtime_suspended(dev)) + pm_runtime_mark_last_busy(dev); + } else { + pm_runtime_dont_use_autosuspend(dev); + pm_runtime_forbid(dev); + } +} +EXPORT_SYMBOL_GPL(snd_hda_power_sync); /** * snd_hda_check_amp_list_power - Check the amp list and update the power @@ -5542,7 +5493,7 @@ void snd_hda_bus_reset(struct hda_bus *bus) cancel_delayed_work_sync(&codec->jackpoll_work); #ifdef CONFIG_PM if (hda_codec_is_power_on(codec)) { - hda_call_codec_suspend(codec, false); + hda_call_codec_suspend(codec); hda_call_codec_resume(codec); } #endif diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 1fa3dd9baf52..593956fc384b 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -372,17 +372,12 @@ struct hda_codec { unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ #ifdef CONFIG_PM - unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_up_notified:1; /* PM notified to controller */ - unsigned int in_pm:1; /* suspend/resume being performed */ - int power_transition; /* power-state in transition */ - int power_count; /* current (global) power refcount */ - struct delayed_work power_work; /* delayed task for powerdown */ + atomic_t in_pm; /* suspend/resume being performed */ unsigned long power_on_acct; unsigned long power_off_acct; unsigned long power_jiffies; - spinlock_t power_lock; #endif /* filter the requested power state per nid */ @@ -595,64 +590,16 @@ const char *snd_hda_get_jack_location(u32 cfg); * power saving */ #ifdef CONFIG_PM -void snd_hda_power_save(struct hda_codec *codec, int delta, bool d3wait); +void snd_hda_power_up(struct hda_codec *codec); +void snd_hda_power_down(struct hda_codec *codec); +void snd_hda_power_sync(struct hda_codec *codec); void snd_hda_update_power_acct(struct hda_codec *codec); #else -static inline void snd_hda_power_save(struct hda_codec *codec, int delta, - bool d3wait) {} +static inline void snd_hda_power_up(struct hda_codec *codec) {} +static inline void snd_hda_power_down(struct hda_codec *codec) {} +static inline void snd_hda_power_sync(struct hda_codec *codec) {} #endif -/** - * snd_hda_power_up - Power-up the codec - * @codec: HD-audio codec - * - * Increment the power-up counter and power up the hardware really when - * not turned on yet. - */ -static inline void snd_hda_power_up(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 1, false); -} - -/** - * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending - * D3 transition to complete. This differs from snd_hda_power_up() when - * power_transition == -1. snd_hda_power_up sees this case as a nop, - * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers - * back up. - * @codec: HD-audio codec - * - * Cancel any power down operation hapenning on the work queue, then power up. - */ -static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 1, true); -} - -/** - * snd_hda_power_down - Power-down the codec - * @codec: HD-audio codec - * - * Decrement the power-up counter and schedules the power-off work if - * the counter rearches to zero. - */ -static inline void snd_hda_power_down(struct hda_codec *codec) -{ - snd_hda_power_save(codec, -1, false); -} - -/** - * snd_hda_power_sync - Synchronize the power-save status - * @codec: HD-audio codec - * - * Synchronize the actual power state with the power account; - * called when power_save parameter is changed - */ -static inline void snd_hda_power_sync(struct hda_codec *codec) -{ - snd_hda_power_save(codec, 0, false); -} - #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 30ddb751806a..522c54f94740 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -836,7 +836,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) buff_step); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); - snd_hda_power_up_d3wait(apcm->codec); + snd_hda_power_up(apcm->codec); err = hinfo->ops.open(hinfo, apcm->codec, substream); if (err < 0) { azx_release_device(azx_dev); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9db1b078801f..26510e60cbe2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1087,6 +1087,7 @@ static int azx_free(struct azx *chip) azx_stop_chip(chip); } + pci->dev.power.ignore_children = 0; /* FIXME */ if (chip->irq >= 0) free_irq(chip->irq, (void*)chip); if (chip->msi) @@ -1796,6 +1797,7 @@ static int azx_probe(struct pci_dev *pci, return err; } + pci->dev.power.ignore_children = 1; /* FIXME */ err = azx_create(card, pci, dev, pci_id->driver_data, &pci_hda_ops, &chip); if (err < 0) diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h index 3a1c63161eb1..c0e1c7d24dbe 100644 --- a/sound/pci/hda/hda_trace.h +++ b/sound/pci/hda/hda_trace.h @@ -87,30 +87,6 @@ DEFINE_EVENT(hda_power, hda_power_up, TP_PROTO(struct hda_codec *codec), TP_ARGS(codec) ); - -TRACE_EVENT(hda_power_count, - TP_PROTO(struct hda_codec *codec), - TP_ARGS(codec), - TP_STRUCT__entry( - __field( unsigned int, card ) - __field( unsigned int, addr ) - __field( int, power_count ) - __field( int, power_on ) - __field( int, power_transition ) - ), - - TP_fast_assign( - __entry->card = (codec)->bus->card->number; - __entry->addr = (codec)->addr; - __entry->power_count = (codec)->power_count; - __entry->power_on = (codec)->power_on; - __entry->power_transition = (codec)->power_transition; - ), - - TP_printk("[%d:%d] power_count=%d, power_on=%d, power_transition=%d", - __entry->card, __entry->addr, __entry->power_count, - __entry->power_on, __entry->power_transition) -); #endif /* CONFIG_PM */ TRACE_EVENT(hda_unsol_event, From 052a9f698268e606ca01eb1ce2a672e548f2ce11 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Mon, 9 Feb 2015 00:18:11 -0800 Subject: [PATCH 044/162] ALSA: Add params_set_format helper Add a helper to set pcm format directly from params Signed-off-by: Fang, Yang A Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/pcm_params.h | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 3c45f3924ba7..c704357775fc 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -366,4 +366,11 @@ static inline int params_physical_width(const struct snd_pcm_hw_params *p) return snd_pcm_format_physical_width(params_format(p)); } +static inline void +params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt) +{ + snd_mask_set(hw_param_mask(p, SNDRV_PCM_HW_PARAM_FORMAT), + (__force int)fmt); +} + #endif /* __SOUND_PCM_PARAMS_H */ From 369a9f5f397fe3258ab937ec7a9c2229d0b1a015 Mon Sep 17 00:00:00 2001 From: "Fang, Yang A" Date: Mon, 9 Feb 2015 00:18:12 -0800 Subject: [PATCH 045/162] ASoC: Intel: fix machine driver warnings this patch will fix below sparse warnings warning: incorrect type in argument 2 (different base types) expected unsigned int [unsigned] val got restricted snd_pcm_format_t [usertype] sound/soc/intel/haswell.c:61:37 sound/soc/intel/broadwell.c:115:37: sound/soc/intel/bytcr_dpcm_rt5640.c:118:37: sound/soc/intel/cht_bsw_rt5672.c:183:37: sound/soc/intel/cht_bsw_rt5645.c:208:37: Signed-off-by: Fang, Yang A Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/intel/broadwell.c | 4 +--- sound/soc/intel/bytcr_dpcm_rt5640.c | 4 +--- sound/soc/intel/cht_bsw_rt5645.c | 4 +--- sound/soc/intel/cht_bsw_rt5672.c | 4 +--- sound/soc/intel/haswell.c | 4 +--- 5 files changed, 5 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 9cf7d01479ad..fba2ef5dac42 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -110,9 +110,7 @@ static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } diff --git a/sound/soc/intel/bytcr_dpcm_rt5640.c b/sound/soc/intel/bytcr_dpcm_rt5640.c index 59308629043e..3b262d01c1b3 100644 --- a/sound/soc/intel/bytcr_dpcm_rt5640.c +++ b/sound/soc/intel/bytcr_dpcm_rt5640.c @@ -113,9 +113,7 @@ static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c index bd29617a9ab9..dd935255a020 100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -203,9 +203,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index ff016621583a..c56f9dfe2129 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -178,9 +178,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP2 to 24-bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S24_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; } diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c index 35edf51a52aa..00fddd3f5dfb 100644 --- a/sound/soc/intel/haswell.c +++ b/sound/soc/intel/haswell.c @@ -56,9 +56,7 @@ static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, channels->min = channels->max = 2; /* set SSP0 to 16 bit */ - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - - SNDRV_PCM_HW_PARAM_FIRST_MASK], - SNDRV_PCM_FORMAT_S16_LE); + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); return 0; } From 48c7699fb2c799d084ce490bceea14fe04ad12a1 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:53 +0530 Subject: [PATCH 046/162] ASoC: core: allow pcms to be registered as nonatomic ALSA core with commit 257f8cce5d40 - "ALSA: pcm: Allow nonatomic trigger operations" allows trigger ops to implemented as nonatomic. For ASoC, we can specify this in dailinks and is updated while snd_pcm is created Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Cc: Takashi Iwai Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/soc-pcm.c | 1 + 2 files changed, 4 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade195628..76bc944dcb5c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -954,6 +954,9 @@ struct snd_soc_dai_link { unsigned int symmetric_channels:1; unsigned int symmetric_samplebits:1; + /* Mark this pcm with non atomic ops */ + bool nonatomic; + /* Do not create a PCM for this DAI link (Backend link) */ unsigned int no_pcm:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6b0136e7cb88..6e3781e88f9a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2511,6 +2511,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + pcm->nonatomic = rtd->dai_link->nonatomic; rtd->pcm = pcm; pcm->private_data = rtd; From 76ca1c2cd8fc0b8764c6360263e2fbca43495ab2 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:54 +0530 Subject: [PATCH 047/162] ASoC: Intel: mark cht machine driver with nonatomic trigger The DSP messages are sent with nonatomic context, which include trigger messages, so mark the driver as nonatomic Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index c56f9dfe2129..a5098d6f988b 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -216,6 +216,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", .ignore_suspend = 1, + .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, .dpcm_capture = 1, @@ -238,6 +239,7 @@ static struct snd_soc_dai_link cht_dailink[] = { .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, + .nonatomic = true, .codec_dai_name = "rt5670-aif1", .codec_name = "i2c-10EC5670:00", .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF From 7b9ca9d7e561ebdc93b43277eb69d20a0dc8f5cd Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:55 +0530 Subject: [PATCH 048/162] ASoC: Intel: update MMX ID to 3 The updated firmware expects the MMX ID to be used as 3, so update the driver as well Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-atom-controls.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index dfebfdd5eb2a..daecc58f28af 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -150,7 +150,7 @@ enum sst_cmd_type { enum sst_task { SST_TASK_SBA = 1, - SST_TASK_MMX, + SST_TASK_MMX = 3, }; enum sst_type { From e0b87d476bc13fc55e7000a84cd1d87c8fdc1f2f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:56 +0530 Subject: [PATCH 049/162] ASoC: Intel: add support for pause and resume in sst This adds missing pcm pause and resume ops in the driver Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_drv_interface.c | 31 +++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 5f75ef3cdd22..5d56fcdd58d8 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -572,6 +572,35 @@ static int sst_stream_drop(struct device *dev, int str_id) return sst_drop_stream(ctx, str_id); } +static int sst_stream_pause(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + + return sst_pause_stream(ctx, str_id); +} + +static int sst_stream_resume(struct device *dev, int str_id) +{ + struct stream_info *str_info; + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + + if (ctx->sst_state != SST_FW_RUNNING) + return 0; + + str_info = get_stream_info(ctx, str_id); + if (!str_info) + return -EINVAL; + return sst_resume_stream(ctx, str_id); +} + static int sst_stream_init(struct device *dev, struct pcm_stream_info *str_info) { int str_id = 0; @@ -633,6 +662,8 @@ static struct sst_ops pcm_ops = { .stream_init = sst_stream_init, .stream_start = sst_stream_start, .stream_drop = sst_stream_drop, + .stream_pause = sst_stream_pause, + .stream_pause_release = sst_stream_resume, .stream_read_tstamp = sst_read_timestamp, .send_byte_stream = sst_send_byte_stream, .close = sst_close_pcm_stream, From fc9406ab9b4a9aac0ab9ad213993824cbe9c65ac Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:57 +0530 Subject: [PATCH 050/162] ASoC: Intel: add support for pcm stream suspend/resume The driver didn't implement support for pcm stream suspend and resume, so add it Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7523cbef8780..ea0fa4b90bb0 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -594,11 +594,13 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, ret_val = stream->ops->stream_drop(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: dev_dbg(rtd->dev, "sst: in pause\n"); status = SST_PLATFORM_PAUSED; ret_val = stream->ops->stream_pause(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: dev_dbg(rtd->dev, "sst: in pause release\n"); status = SST_PLATFORM_RUNNING; ret_val = stream->ops->stream_pause_release(sst->dev, str_id); From 54e6beccc67129c474aad7578951112c6cf28e01 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Thu, 12 Feb 2015 09:59:58 +0530 Subject: [PATCH 051/162] ASoC: Intel: add support for platform suspend This adds support for platform suspend and resume. We ensure all pcms are suspended by invoking snd_soc_suspend() and then stop the DSP Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst-mfld-platform-pcm.c | 58 +++++++++++++++++++++++++ sound/soc/intel/sst-mfld-platform.h | 1 + 2 files changed, 59 insertions(+) diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index ea0fa4b90bb0..2fbaf2c75d17 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -667,6 +667,9 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) static int sst_soc_probe(struct snd_soc_platform *platform) { + struct sst_data *drv = dev_get_drvdata(platform->dev); + + drv->soc_card = platform->component.card; return sst_dsp_init_v2_dpcm(platform); } @@ -729,9 +732,64 @@ static int sst_platform_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP + +static int sst_soc_prepare(struct device *dev) +{ + struct sst_data *drv = dev_get_drvdata(dev); + int i; + + /* suspend all pcms first */ + snd_soc_suspend(drv->soc_card->dev); + snd_soc_poweroff(drv->soc_card->dev); + + /* set the SSPs to idle */ + for (i = 0; i < drv->soc_card->num_rtd; i++) { + struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + + if (dai->active) { + send_ssp_cmd(dai, dai->name, 0); + sst_handle_vb_timer(dai, false); + } + } + + return 0; +} + +static void sst_soc_complete(struct device *dev) +{ + struct sst_data *drv = dev_get_drvdata(dev); + int i; + + /* restart SSPs */ + for (i = 0; i < drv->soc_card->num_rtd; i++) { + struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + + if (dai->active) { + sst_handle_vb_timer(dai, true); + send_ssp_cmd(dai, dai->name, 1); + } + } + snd_soc_resume(drv->soc_card->dev); +} + +#else + +#define sst_soc_prepare NULL +#define sst_soc_complete NULL + +#endif + + +static const struct dev_pm_ops sst_platform_pm = { + .prepare = sst_soc_prepare, + .complete = sst_soc_complete, +}; + static struct platform_driver sst_platform_driver = { .driver = { .name = "sst-mfld-platform", + .pm = &sst_platform_pm, }, .probe = sst_platform_probe, .remove = sst_platform_remove, diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 79c8d1246a8f..9094314be2b0 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -174,6 +174,7 @@ struct sst_data { struct sst_platform_data *pdata; struct snd_sst_bytes_v2 *byte_stream; struct mutex lock; + struct snd_soc_card *soc_card; }; int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); From 5c88b4e91d3b6a3d701d7b134fa945e6309e7068 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Thu, 12 Feb 2015 10:00:01 +0530 Subject: [PATCH 052/162] ASoC: Intel: Add memcpy32_fromio as well Export 32-bit version of memcpy for use in suspend/resume. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.h | 3 +++ sound/soc/intel/sst/sst_loader.c | 10 +++++++++- 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index 562bc483d6b7..f793780a50a2 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -544,4 +544,7 @@ int sst_alloc_drv_context(struct intel_sst_drv **ctx, int sst_context_init(struct intel_sst_drv *ctx); void sst_context_cleanup(struct intel_sst_drv *ctx); void sst_configure_runtime_pm(struct intel_sst_drv *ctx); +void memcpy32_toio(void __iomem *dst, const void *src, int count); +void memcpy32_fromio(void *dst, const void __iomem *src, int count); + #endif diff --git a/sound/soc/intel/sst/sst_loader.c b/sound/soc/intel/sst/sst_loader.c index 7888cd707853..e88907ae8b15 100644 --- a/sound/soc/intel/sst/sst_loader.c +++ b/sound/soc/intel/sst/sst_loader.c @@ -39,7 +39,15 @@ #include "sst.h" #include "../sst-dsp.h" -static inline void memcpy32_toio(void __iomem *dst, const void *src, int count) +void memcpy32_toio(void __iomem *dst, const void *src, int count) +{ + /* __iowrite32_copy uses 32-bit count values so divide by 4 for + * right count in words + */ + __iowrite32_copy(dst, src, count/4); +} + +void memcpy32_fromio(void *dst, const void __iomem *src, int count) { /* __iowrite32_copy uses 32-bit count values so divide by 4 for * right count in words From 4a8448d4289d7210053a43f9f21e42929beb159b Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 24 Feb 2015 11:39:44 +0530 Subject: [PATCH 053/162] ASoC: Intel: add pm support in sst ipc driver This adds support for system pm support. We need to save the dsp memory which gets lost on suspend and restore that on resume Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 128 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst/sst.h | 9 +++ 2 files changed, 137 insertions(+) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8a8d56a146e7..8f938112a01f 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -415,6 +415,83 @@ static int intel_sst_runtime_suspend(struct device *dev) return ret; } +static int intel_sst_suspend(struct device *dev) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + struct sst_fw_save *fw_save; + int i, ret = 0; + + /* check first if we are already in SW reset */ + if (ctx->sst_state == SST_RESET) + return 0; + + /* + * check if any stream is active and running + * they should already by suspend by soc_suspend + */ + for (i = 1; i <= ctx->info.max_streams; i++) { + struct stream_info *stream = &ctx->streams[i]; + + if (stream->status == STREAM_RUNNING) { + dev_err(dev, "stream %d is running, cant susupend, abort\n", i); + return -EBUSY; + } + } + synchronize_irq(ctx->irq_num); + flush_workqueue(ctx->post_msg_wq); + + /* Move the SST state to Reset */ + sst_set_fw_state_locked(ctx, SST_RESET); + + /* tell DSP we are suspending */ + if (ctx->ops->save_dsp_context(ctx)) + return -EBUSY; + + /* save the memories */ + fw_save = kzalloc(sizeof(*fw_save), GFP_KERNEL); + if (!fw_save) + return -ENOMEM; + fw_save->iram = kzalloc(ctx->iram_end - ctx->iram_base, GFP_KERNEL); + if (!fw_save->iram) { + ret = -ENOMEM; + goto iram; + } + fw_save->dram = kzalloc(ctx->dram_end - ctx->dram_base, GFP_KERNEL); + if (!fw_save->dram) { + ret = -ENOMEM; + goto dram; + } + fw_save->sram = kzalloc(SST_MAILBOX_SIZE, GFP_KERNEL); + if (!fw_save->sram) { + ret = -ENOMEM; + goto sram; + } + + fw_save->ddr = kzalloc(ctx->ddr_end - ctx->ddr_base, GFP_KERNEL); + if (!fw_save->ddr) { + ret = -ENOMEM; + goto ddr; + } + + memcpy32_fromio(fw_save->iram, ctx->iram, ctx->iram_end - ctx->iram_base); + memcpy32_fromio(fw_save->dram, ctx->dram, ctx->dram_end - ctx->dram_base); + memcpy32_fromio(fw_save->sram, ctx->mailbox, SST_MAILBOX_SIZE); + memcpy32_fromio(fw_save->ddr, ctx->ddr, ctx->ddr_end - ctx->ddr_base); + + ctx->fw_save = fw_save; + ctx->ops->reset(ctx); + return 0; +ddr: + kfree(fw_save->sram); +sram: + kfree(fw_save->dram); +dram: + kfree(fw_save->iram); +iram: + kfree(fw_save); + return ret; +} + static int intel_sst_runtime_resume(struct device *dev) { int ret = 0; @@ -430,7 +507,58 @@ static int intel_sst_runtime_resume(struct device *dev) return ret; } +static int intel_sst_resume(struct device *dev) +{ + struct intel_sst_drv *ctx = dev_get_drvdata(dev); + struct sst_fw_save *fw_save = ctx->fw_save; + int ret = 0; + struct sst_block *block; + + if (!fw_save) + return intel_sst_runtime_resume(dev); + + sst_set_fw_state_locked(ctx, SST_FW_LOADING); + + /* we have to restore the memory saved */ + ctx->ops->reset(ctx); + + ctx->fw_save = NULL; + + memcpy32_toio(ctx->iram, fw_save->iram, ctx->iram_end - ctx->iram_base); + memcpy32_toio(ctx->dram, fw_save->dram, ctx->dram_end - ctx->dram_base); + memcpy32_toio(ctx->mailbox, fw_save->sram, SST_MAILBOX_SIZE); + memcpy32_toio(ctx->ddr, fw_save->ddr, ctx->ddr_end - ctx->ddr_base); + + kfree(fw_save->sram); + kfree(fw_save->dram); + kfree(fw_save->iram); + kfree(fw_save->ddr); + kfree(fw_save); + + block = sst_create_block(ctx, 0, FW_DWNL_ID); + if (block == NULL) + return -ENOMEM; + + + /* start and wait for ack */ + ctx->ops->start(ctx); + ret = sst_wait_timeout(ctx, block); + if (ret) { + dev_err(ctx->dev, "fw download failed %d\n", ret); + /* FW download failed due to timeout */ + ret = -EBUSY; + + } else { + sst_set_fw_state_locked(ctx, SST_FW_RUNNING); + } + + sst_free_block(ctx, block); + return ret; +} + const struct dev_pm_ops intel_sst_pm = { + .suspend = intel_sst_suspend, + .resume = intel_sst_resume, .runtime_suspend = intel_sst_runtime_suspend, .runtime_resume = intel_sst_runtime_resume, }; diff --git a/sound/soc/intel/sst/sst.h b/sound/soc/intel/sst/sst.h index f793780a50a2..3f493862e98d 100644 --- a/sound/soc/intel/sst/sst.h +++ b/sound/soc/intel/sst/sst.h @@ -337,6 +337,13 @@ struct sst_shim_regs64 { u64 csr2; }; +struct sst_fw_save { + void *iram; + void *dram; + void *sram; + void *ddr; +}; + /** * struct intel_sst_drv - driver ops * @@ -428,6 +435,8 @@ struct intel_sst_drv { * persistent till worker thread gets called */ char firmware_name[FW_NAME_SIZE]; + + struct sst_fw_save *fw_save; }; /* misc definitions */ From 9308d1456e30e374d93957e3376a09370be9dc52 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 24 Feb 2015 11:39:45 +0530 Subject: [PATCH 054/162] ASoC: Intel: Move the fw download to power_control Thus removing the runtime_resume handler. Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst.c | 18 +---------------- sound/soc/intel/sst/sst_drv_interface.c | 27 +++++++++++++++++++++---- 2 files changed, 24 insertions(+), 21 deletions(-) diff --git a/sound/soc/intel/sst/sst.c b/sound/soc/intel/sst/sst.c index 8f938112a01f..4d8f73ac5dd9 100644 --- a/sound/soc/intel/sst/sst.c +++ b/sound/soc/intel/sst/sst.c @@ -492,21 +492,6 @@ iram: return ret; } -static int intel_sst_runtime_resume(struct device *dev) -{ - int ret = 0; - struct intel_sst_drv *ctx = dev_get_drvdata(dev); - - if (ctx->sst_state == SST_RESET) { - ret = sst_load_fw(ctx); - if (ret) { - dev_err(dev, "FW download fail %d\n", ret); - sst_set_fw_state_locked(ctx, SST_RESET); - } - } - return ret; -} - static int intel_sst_resume(struct device *dev) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); @@ -515,7 +500,7 @@ static int intel_sst_resume(struct device *dev) struct sst_block *block; if (!fw_save) - return intel_sst_runtime_resume(dev); + return 0; sst_set_fw_state_locked(ctx, SST_FW_LOADING); @@ -560,6 +545,5 @@ const struct dev_pm_ops intel_sst_pm = { .suspend = intel_sst_suspend, .resume = intel_sst_resume, .runtime_suspend = intel_sst_runtime_suspend, - .runtime_resume = intel_sst_runtime_resume, }; EXPORT_SYMBOL_GPL(intel_sst_pm); diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 5d56fcdd58d8..549af7d7f6d0 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -138,12 +138,31 @@ int sst_get_stream(struct intel_sst_drv *ctx, static int sst_power_control(struct device *dev, bool state) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); + int ret = 0; - dev_dbg(ctx->dev, "state:%d", state); - if (state == true) - return pm_runtime_get_sync(dev); - else + if (state == true) { + ret = pm_runtime_get_sync(dev); + dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", + atomic_read(&dev->power.usage_count)); + if (ret < 0) { + dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); + return ret; + } + if ((ctx->sst_state == SST_RESET) && + (atomic_read(&dev->power.usage_count) == 1)) { + ret = sst_load_fw(ctx); + if (ret) { + dev_err(dev, "FW download fail %d\n", ret); + sst_set_fw_state_locked(ctx, SST_RESET); + ret = sst_pm_runtime_put(ctx); + } + } + } else { + dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", + atomic_read(&dev->power.usage_count)); return sst_pm_runtime_put(ctx); + } + return ret; } /* From 583e58a0f0e996008780fe4df0f7640890a9b69a Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 24 Feb 2015 11:39:46 +0530 Subject: [PATCH 055/162] ASoC: Intel: Remove ignore suspend support In our platform we want platform and codec driver routines to get invoked and don't need the machine routines so remove here Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index a5098d6f988b..67db5101bc89 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -215,7 +215,6 @@ static struct snd_soc_dai_link cht_dailink[] = { .codec_dai_name = "snd-soc-dummy-dai", .codec_name = "snd-soc-dummy", .platform_name = "sst-mfld-platform", - .ignore_suspend = 1, .nonatomic = true, .dynamic = 1, .dpcm_playback = 1, @@ -246,7 +245,6 @@ static struct snd_soc_dai_link cht_dailink[] = { | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, .be_hw_params_fixup = cht_codec_fixup, - .ignore_suspend = 1, .dpcm_playback = 1, .dpcm_capture = 1, .ops = &cht_be_ssp2_ops, From 3f2dcbeaeb2badb951a68e7d525ff4e55d0b0678 Mon Sep 17 00:00:00 2001 From: "Subhransu S. Prusty" Date: Tue, 24 Feb 2015 11:39:47 +0530 Subject: [PATCH 056/162] ASoC: Intel: Remove soc pm handling to allow platform driver handle it Signed-off-by: Subhransu S. Prusty Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5672.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index 67db5101bc89..bc8dcacd5e6a 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -283,7 +283,6 @@ static int snd_cht_mc_probe(struct platform_device *pdev) static struct platform_driver snd_cht_mc_driver = { .driver = { .name = "cht-bsw-rt5672", - .pm = &snd_soc_pm_ops, }, .probe = snd_cht_mc_probe, }; From 34d7c3905adb9a9d8f8155857c76314125510817 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Sat, 21 Feb 2015 16:33:24 +0100 Subject: [PATCH 057/162] ASoC: improve usage of gpiod API MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Since 39b2bbe3d715 (gpio: add flags argument to gpiod_get*() functions) which appeared in v3.17-rc1, the gpiod_get* functions take an additional parameter that allows to specify direction and initial value for output. Simplify drivers accordingly. Also there is an *_optional variant that serves well here. The sematics is slightly changed here by using it as error checking is more strict now: If GPIOLIB is not enabled an error is returned instead of just ignoring the gpio. On one hand this is bad for devices that don't "have" the respective gpio as the driver is failing now. On the other hand there is no means to assert that this gpio is really not needed or if only the driver to control it is not available. The latter is a real reason to fail and so it's defensive to fail here, too. Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown --- sound/soc/codecs/adau1977.c | 17 +++++------------ sound/soc/codecs/cs35l32.c | 17 +++++------------ sound/soc/codecs/cs4265.c | 17 +++++------------ sound/soc/codecs/sta350.c | 28 ++++++++-------------------- sound/soc/codecs/tas2552.c | 13 +++---------- 5 files changed, 26 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c index 70ab35744aba..7ad8e156e2df 100644 --- a/sound/soc/codecs/adau1977.c +++ b/sound/soc/codecs/adau1977.c @@ -938,22 +938,15 @@ int adau1977_probe(struct device *dev, struct regmap *regmap, adau1977->dvdd_reg = NULL; } - adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); - if (IS_ERR(adau1977->reset_gpio)) { - ret = PTR_ERR(adau1977->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return PTR_ERR(adau1977->reset_gpio); - adau1977->reset_gpio = NULL; - } + adau1977->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(adau1977->reset_gpio)) + return PTR_ERR(adau1977->reset_gpio); dev_set_drvdata(dev, adau1977); - if (adau1977->reset_gpio) { - ret = gpiod_direction_output(adau1977->reset_gpio, 0); - if (ret) - return ret; + if (adau1977->reset_gpio) ndelay(100); - } ret = adau1977_power_enable(adau1977); if (ret) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index f2b8aad21274..60598b230341 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -437,20 +437,13 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, } /* Reset the Device */ - cs35l32->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs35l32->reset_gpio)) { - ret = PTR_ERR(cs35l32->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; + cs35l32->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs35l32->reset_gpio)) + return PTR_ERR(cs35l32->reset_gpio); - cs35l32->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs35l32->reset_gpio, 0); - if (ret) - return ret; + if (cs35l32->reset_gpio) gpiod_set_value_cansleep(cs35l32->reset_gpio, 1); - } /* initialize codec */ ret = regmap_read(cs35l32->regmap, CS35L32_DEVID_AB, ®); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index ce6086835ebd..cac48ddf3ba6 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -605,21 +605,14 @@ static int cs4265_i2c_probe(struct i2c_client *i2c_client, return ret; } - cs4265->reset_gpio = devm_gpiod_get(&i2c_client->dev, - "reset-gpios"); - if (IS_ERR(cs4265->reset_gpio)) { - ret = PTR_ERR(cs4265->reset_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; + cs4265->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs4265->reset_gpio)) + return PTR_ERR(cs4265->reset_gpio); - cs4265->reset_gpio = NULL; - } else { - ret = gpiod_direction_output(cs4265->reset_gpio, 0); - if (ret) - return ret; + if (cs4265->reset_gpio) { mdelay(1); gpiod_set_value_cansleep(cs4265->reset_gpio, 1); - } i2c_set_clientdata(i2c_client, cs4265); diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index bda2ee18769e..669e3228241e 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -1213,27 +1213,15 @@ static int sta350_i2c_probe(struct i2c_client *i2c, #endif /* GPIOs */ - sta350->gpiod_nreset = devm_gpiod_get(dev, "reset"); - if (IS_ERR(sta350->gpiod_nreset)) { - ret = PTR_ERR(sta350->gpiod_nreset); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; + sta350->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_nreset)) + return PTR_ERR(sta350->gpiod_nreset); - sta350->gpiod_nreset = NULL; - } else { - gpiod_direction_output(sta350->gpiod_nreset, 0); - } - - sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down"); - if (IS_ERR(sta350->gpiod_power_down)) { - ret = PTR_ERR(sta350->gpiod_power_down); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - sta350->gpiod_power_down = NULL; - } else { - gpiod_direction_output(sta350->gpiod_power_down, 0); - } + sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down", + GPIOD_OUT_LOW); + if (IS_ERR(sta350->gpiod_power_down)) + return PTR_ERR(sta350->gpiod_power_down); /* regulators */ for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index ae23acdd2708..dfb4ff5cc9ea 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -485,16 +485,9 @@ static int tas2552_probe(struct i2c_client *client, if (data == NULL) return -ENOMEM; - data->enable_gpio = devm_gpiod_get(dev, "enable"); - if (IS_ERR(data->enable_gpio)) { - ret = PTR_ERR(data->enable_gpio); - if (ret != -ENOENT && ret != -ENOSYS) - return ret; - - data->enable_gpio = NULL; - } else { - gpiod_direction_output(data->enable_gpio, 0); - } + data->enable_gpio = devm_gpiod_get(dev, "enable", GPIOD_OUT_LOW); + if (IS_ERR(data->enable_gpio)) + return PTR_ERR(data->enable_gpio); data->tas2552_client = client; data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); From 5890bd5256bc026c425361fa087dc05c7a24d853 Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 16 Feb 2015 22:02:47 +0100 Subject: [PATCH 058/162] ASoC: pcm512x: Rearrange to not repeat dacsrc_rate / dac_div Signed-off-by: Peter Rosin Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 9974f201a08f..f11c76f1acfe 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -863,28 +863,29 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, dacsrc_rate = sck_rate; } + osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate); + if (osr_div > 128) { + dev_err(dev, "Failed to find OSR divider\n"); + return -EINVAL; + } + dac_div = DIV_ROUND_CLOSEST(dacsrc_rate, dac_rate); if (dac_div > 128) { dev_err(dev, "Failed to find DAC divider\n"); return -EINVAL; } + dac_rate = dacsrc_rate / dac_div; - ncp_div = DIV_ROUND_CLOSEST(dacsrc_rate / dac_div, 1536000); - if (ncp_div > 128 || dacsrc_rate / dac_div / ncp_div > 2048000) { + ncp_div = DIV_ROUND_CLOSEST(dac_rate, 1536000); + if (ncp_div > 128 || dac_rate / ncp_div > 2048000) { /* run NCP no faster than 2048000 Hz, but why? */ - ncp_div = DIV_ROUND_UP(dacsrc_rate / dac_div, 2048000); + ncp_div = DIV_ROUND_UP(dac_rate, 2048000); if (ncp_div > 128) { dev_err(dev, "Failed to find NCP divider\n"); return -EINVAL; } } - osr_div = DIV_ROUND_CLOSEST(dac_rate, osr_rate); - if (osr_div > 128) { - dev_err(dev, "Failed to find OSR divider\n"); - return -EINVAL; - } - idac = mck_rate / (dsp_div * sample_rate); ret = regmap_write(pcm512x->regmap, PCM512x_DSP_CLKDIV, dsp_div - 1); From f29933c9ae4b8f30c713186d3babb630c7cfb4f2 Mon Sep 17 00:00:00 2001 From: Peter Rosin Date: Mon, 23 Feb 2015 21:03:33 +0100 Subject: [PATCH 059/162] ASoC: pcm512x: Allow independently overclocking PLL, DAC and DSP When using non-standard rates, a relatively small amount of overclocking can make a big difference to a number of cases. - Not all rates are possible to achieve with the PLL, due to divider restrictions. - The higher oversampling rates that can be used by the DAC, the simpler the analog output filters get (mirror frequencies move up, away from the desired spectrum). - The more work the DSP can perform per sample, the better. For standard rates, there is little to gain as everything is designed just right, and the needed overclocking to make a real difference would be significant. Signed-off-by: Peter Rosin Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 161 ++++++++++++++++++++++++++++++++++--- 1 file changed, 150 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index f11c76f1acfe..4b5f1fe9be97 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -54,6 +54,9 @@ struct pcm512x_priv { int pll_d; int pll_p; unsigned long real_pll; + unsigned long overclock_pll; + unsigned long overclock_dac; + unsigned long overclock_dsp; }; /* @@ -224,6 +227,90 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) } } +static int pcm512x_overclock_pll_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_pll; + return 0; +} + +static int pcm512x_overclock_pll_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_pll = ucontrol->value.integer.value[0]; + return 0; +} + +static int pcm512x_overclock_dsp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_dsp; + return 0; +} + +static int pcm512x_overclock_dsp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_dsp = ucontrol->value.integer.value[0]; + return 0; +} + +static int pcm512x_overclock_dac_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = pcm512x->overclock_dac; + return 0; +} + +static int pcm512x_overclock_dac_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); + + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + case SND_SOC_BIAS_STANDBY: + break; + default: + return -EBUSY; + } + + pcm512x->overclock_dac = ucontrol->value.integer.value[0]; + return 0; +} + static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0); @@ -328,6 +415,13 @@ SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf), SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus), SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf), SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds), + +SOC_SINGLE_EXT("Max Overclock PLL", SND_SOC_NOPM, 0, 20, 0, + pcm512x_overclock_pll_get, pcm512x_overclock_pll_put), +SOC_SINGLE_EXT("Max Overclock DSP", SND_SOC_NOPM, 0, 40, 0, + pcm512x_overclock_dsp_get, pcm512x_overclock_dsp_put), +SOC_SINGLE_EXT("Max Overclock DAC", SND_SOC_NOPM, 0, 40, 0, + pcm512x_overclock_dac_get, pcm512x_overclock_dac_put), }; static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = { @@ -346,6 +440,45 @@ static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = { { "OUTR", NULL, "DACR" }, }; +static unsigned long pcm512x_pll_max(struct pcm512x_priv *pcm512x) +{ + return 25000000 + 25000000 * pcm512x->overclock_pll / 100; +} + +static unsigned long pcm512x_dsp_max(struct pcm512x_priv *pcm512x) +{ + return 50000000 + 50000000 * pcm512x->overclock_dsp / 100; +} + +static unsigned long pcm512x_dac_max(struct pcm512x_priv *pcm512x, + unsigned long rate) +{ + return rate + rate * pcm512x->overclock_dac / 100; +} + +static unsigned long pcm512x_sck_max(struct pcm512x_priv *pcm512x) +{ + if (!pcm512x->pll_out) + return 25000000; + return pcm512x_pll_max(pcm512x); +} + +static unsigned long pcm512x_ncp_target(struct pcm512x_priv *pcm512x, + unsigned long dac_rate) +{ + /* + * If the DAC is not actually overclocked, use the good old + * NCP target rate... + */ + if (dac_rate <= 6144000) + return 1536000; + /* + * ...but if the DAC is in fact overclocked, bump the NCP target + * rate to get the recommended dividers even when overclocking. + */ + return pcm512x_dac_max(pcm512x, 1536000); +} + static const u32 pcm512x_dai_rates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, 384000, @@ -359,6 +492,7 @@ static const struct snd_pcm_hw_constraint_list constraints_slave = { static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { + struct pcm512x_priv *pcm512x = rule->private; struct snd_interval ranges[2]; int frame_size; @@ -377,7 +511,7 @@ static int pcm512x_hw_rule_rate(struct snd_pcm_hw_params *params, */ memset(ranges, 0, sizeof(ranges)); ranges[0].min = 8000; - ranges[0].max = 25000000 / frame_size / 2; + ranges[0].max = pcm512x_sck_max(pcm512x) / frame_size / 2; ranges[1].min = DIV_ROUND_UP(16000000, frame_size); ranges[1].max = 384000; break; @@ -408,7 +542,7 @@ static int pcm512x_dai_startup_master(struct snd_pcm_substream *substream, return snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, pcm512x_hw_rule_rate, - NULL, + pcm512x, SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -517,6 +651,8 @@ static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai, unsigned long bclk_rate) { struct device *dev = dai->dev; + struct snd_soc_codec *codec = dai->codec; + struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec); unsigned long sck_rate; int pow2; @@ -527,9 +663,10 @@ static unsigned long pcm512x_find_sck(struct snd_soc_dai *dai, * as many factors of 2 as possible, as that makes it easier * to find a fast DAC rate */ - pow2 = 1 << fls((25000000 - 16000000) / bclk_rate); + pow2 = 1 << fls((pcm512x_pll_max(pcm512x) - 16000000) / bclk_rate); for (; pow2; pow2 >>= 1) { - sck_rate = rounddown(25000000, bclk_rate * pow2); + sck_rate = rounddown(pcm512x_pll_max(pcm512x), + bclk_rate * pow2); if (sck_rate >= 16000000) break; } @@ -678,7 +815,7 @@ static unsigned long pcm512x_pllin_dac_rate(struct snd_soc_dai *dai, return 0; /* futile, quit early */ /* run DAC no faster than 6144000 Hz */ - for (dac_rate = rounddown(6144000, osr_rate); + for (dac_rate = rounddown(pcm512x_dac_max(pcm512x, 6144000), osr_rate); dac_rate; dac_rate -= osr_rate) { @@ -805,7 +942,7 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, osr_rate = 16 * sample_rate; /* run DSP no faster than 50 MHz */ - dsp_div = mck_rate > 50000000 ? 2 : 1; + dsp_div = mck_rate > pcm512x_dsp_max(pcm512x) ? 2 : 1; dac_rate = pcm512x_pllin_dac_rate(dai, osr_rate, pllin_rate); if (dac_rate) { @@ -836,7 +973,8 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, dacsrc_rate = pllin_rate; } else { /* run DAC no faster than 6144000 Hz */ - unsigned long dac_mul = 6144000 / osr_rate; + unsigned long dac_mul = pcm512x_dac_max(pcm512x, 6144000) + / osr_rate; unsigned long sck_mul = sck_rate / osr_rate; for (; dac_mul; dac_mul--) { @@ -876,7 +1014,8 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, } dac_rate = dacsrc_rate / dac_div; - ncp_div = DIV_ROUND_CLOSEST(dac_rate, 1536000); + ncp_div = DIV_ROUND_CLOSEST(dac_rate, + pcm512x_ncp_target(pcm512x, dac_rate)); if (ncp_div > 128 || dac_rate / ncp_div > 2048000) { /* run NCP no faster than 2048000 Hz, but why? */ ncp_div = DIV_ROUND_UP(dac_rate, 2048000); @@ -938,11 +1077,11 @@ static int pcm512x_set_dividers(struct snd_soc_dai *dai, return ret; } - if (sample_rate <= 48000) + if (sample_rate <= pcm512x_dac_max(pcm512x, 48000)) fssp = PCM512x_FSSP_48KHZ; - else if (sample_rate <= 96000) + else if (sample_rate <= pcm512x_dac_max(pcm512x, 96000)) fssp = PCM512x_FSSP_96KHZ; - else if (sample_rate <= 192000) + else if (sample_rate <= pcm512x_dac_max(pcm512x, 192000)) fssp = PCM512x_FSSP_192KHZ; else fssp = PCM512x_FSSP_384KHZ; From f23e860edbb3f2208c0ab3448e756689bb4a3760 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Sat, 14 Feb 2015 17:22:49 -0800 Subject: [PATCH 060/162] ASoC: core: Add extra dapm properties for Device Tree The current helper functions, snd_soc_of_parse_audio_simple_widgets() and snd_soc_of_parse_audio_routing(), set dapm_widgets and dapm_routes without caring if they are already set by using build-in widgets and routes in the card driver. So there could be one of them, build-in one or Device Tree one, overrided by the other depending on which one was assigned later. This patch adds an extra pair of dapm_widgets and dapm_routes for DT use only so as to prevent unexpected overriding. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 16 ++++++++++++---- 2 files changed, 17 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade195628..f66a1ef98a40 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1071,11 +1071,16 @@ struct snd_soc_card { /* * Card-specific routes and widgets. + * Note: of_dapm_xxx for Device Tree; Otherwise for driver build-in. */ const struct snd_soc_dapm_widget *dapm_widgets; int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; int num_dapm_routes; + const struct snd_soc_dapm_widget *of_dapm_widgets; + int num_of_dapm_widgets; + const struct snd_soc_dapm_route *of_dapm_routes; + int num_of_dapm_routes; bool fully_routed; struct work_struct deferred_resume_work; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30579ca5bacb..5c0658d49609 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1561,6 +1561,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, card->num_dapm_widgets); + if (card->of_dapm_widgets) + snd_soc_dapm_new_controls(&card->dapm, card->of_dapm_widgets, + card->num_of_dapm_widgets); + /* initialise the sound card only once */ if (card->probe) { ret = card->probe(card); @@ -1616,6 +1620,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + if (card->of_dapm_routes) + snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, + card->num_of_dapm_routes); + for (i = 0; i < card->num_links; i++) { if (card->dai_link[i].dai_fmt) snd_soc_runtime_set_dai_fmt(&card->rtd[i], @@ -3223,8 +3231,8 @@ int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, widgets[i].name = wname; } - card->dapm_widgets = widgets; - card->num_dapm_widgets = num_widgets; + card->of_dapm_widgets = widgets; + card->num_of_dapm_widgets = num_widgets; return 0; } @@ -3308,8 +3316,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } } - card->num_dapm_routes = num_routes; - card->dapm_routes = routes; + card->num_of_dapm_routes = num_routes; + card->of_dapm_routes = routes; return 0; } From 3185878a70e721644b0e32ebbc0a039616551949 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Sat, 14 Feb 2015 17:22:50 -0800 Subject: [PATCH 061/162] ASoC: fsl-asoc-card: Add snd_soc_of_parse_audio_routing() This patch adds snd_soc_of_parse_audio_routing() to get dapm routes configurations via Device Tree. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 3f6959c8e2f7..de438871040b 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -512,6 +512,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + ret = snd_soc_of_parse_audio_routing(&priv->card, "audio-routing"); + if (ret) { + dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); + goto asrc_fail; + } + /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; From e2cef68d5903cc2052e9f6e46b323b7ead695e73 Mon Sep 17 00:00:00 2001 From: Jie Yang Date: Tue, 10 Feb 2015 17:01:56 +0800 Subject: [PATCH 062/162] ASoC: rt286: add jack detection disable with NULL jack passed Some platforms, e.g. WSB, don't need jack detection when system is in Suspend, for power save reason. Here add headphone/mic jack detection disable feature with NULL jack passed in, when disabled, it will disable interrupt, and disable LDO1, which is used for jack detection when headphone is plugged in. Signed-off-by: Jie Yang Reviewed-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt286.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index f374840a5a7c..16723b167fbf 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -395,9 +395,20 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) rt286->jack = jack; - /* Send an initial empty report */ - snd_soc_jack_report(rt286->jack, 0, - SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + if (jack) { + /* enable IRQ */ + if (rt286->jack->status | SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO1"); + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2); + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, rt286->jack->status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + } else { + /* disable IRQ */ + regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x0); + snd_soc_dapm_disable_pin(&codec->dapm, "LDO1"); + } + snd_soc_dapm_sync(&codec->dapm); return 0; } From 19449593d60b75654fe33a98c4fb8ff8a38ac1e0 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:27:10 +0300 Subject: [PATCH 063/162] sound: sys_timer: indent poll_def_tmr() correctly The indenting here was really whacky and not consistent from one line to the next. I also reverse the "if (opened)" and "if (tmr_running)" tests so that I could remove two indent levels. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/sys_timer.c | 33 ++++++++++++++------------------- 1 file changed, 14 insertions(+), 19 deletions(-) diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c index 9f039831114c..2226dda0eff0 100644 --- a/sound/oss/sys_timer.c +++ b/sound/oss/sys_timer.c @@ -50,29 +50,24 @@ tmr2ticks(int tmr_value) static void poll_def_tmr(unsigned long dummy) { + if (!opened) + return; + def_tmr.expires = (1) + jiffies; + add_timer(&def_tmr); - if (opened) - { + if (!tmr_running) + return; - { - def_tmr.expires = (1) + jiffies; - add_timer(&def_tmr); - } + spin_lock(&lock); + tmr_ctr++; + curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); - if (tmr_running) - { - spin_lock(&lock); - tmr_ctr++; - curr_ticks = ticks_offs + tmr2ticks(tmr_ctr); + if (curr_ticks >= next_event_time) { + next_event_time = (unsigned long) -1; + sequencer_timer(0); + } - if (curr_ticks >= next_event_time) - { - next_event_time = (unsigned long) -1; - sequencer_timer(0); - } - spin_unlock(&lock); - } - } + spin_unlock(&lock); } static void From df403869e35f88e1e45483d31ee70b4aa3ed8896 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:30:36 +0300 Subject: [PATCH 064/162] sound/oss/opl3: remove some stray whitespace Removed an extra tab and a extra space character. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/opl3.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 607cee4d545e..b6d19adf8f41 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -666,7 +666,7 @@ static int opl3_start_note (int dev, int voice, int note, int volume) opl3_command(map->ioaddr, FNUM_LOW + map->voice_num, data); data = 0x20 | ((block & 0x7) << 2) | ((fnum >> 8) & 0x3); - devc->voc[voice].keyon_byte = data; + devc->voc[voice].keyon_byte = data; opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num, data); if (voice_mode == 4) opl3_command(map->ioaddr, KEYON_BLOCK + map->voice_num + 3, data); @@ -717,7 +717,7 @@ static void freq_to_fnum (int freq, int *block, int *fnum) static void opl3_command (int io_addr, unsigned int addr, unsigned int val) { - int i; + int i; /* * The original 2-OP synth requires a quite long delay after writing to a From e214e5183d9da3b61f775d3ae7202ea8aa10ebed Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:31:43 +0300 Subject: [PATCH 065/162] sound/sb_ess: white space cleanups These weren't aligned on the same lines as the surrounding code and the printk was quite messy. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/sb_ess.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index b47a69026f1b..57f7d25a2cd3 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -604,7 +604,7 @@ static void ess_audio_output_block_audio2 ess_chgmixer (devc, 0x78, 0x03, 0x03); /* Go */ devc->irq_mode_16 = IMODE_OUTPUT; - devc->intr_active_16 = 1; + devc->intr_active_16 = 1; } static void ess_audio_output_block @@ -1183,17 +1183,12 @@ FKS_test (devc); chip = "ES1688"; } - printk ( KERN_INFO "ESS chip %s %s%s\n" - , chip - , ( devc->sbmo.esstype == ESSTYPE_DETECT || devc->sbmo.esstype == ESSTYPE_LIKE20 - ? "detected" - : "specified" - ) - , ( devc->sbmo.esstype == ESSTYPE_LIKE20 - ? " (kernel 2.0 compatible)" - : "" - ) - ); + printk(KERN_INFO "ESS chip %s %s%s\n", chip, + (devc->sbmo.esstype == ESSTYPE_DETECT || + devc->sbmo.esstype == ESSTYPE_LIKE20) ? + "detected" : "specified", + devc->sbmo.esstype == ESSTYPE_LIKE20 ? + " (kernel 2.0 compatible)" : ""); sprintf(name,"ESS %s AudioDrive (rev %d)", chip, ess_minor & 0x0f); } else { From f0418d46d6ad25be991d557c6c50d1e61b4ba690 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:32:16 +0300 Subject: [PATCH 066/162] sound/sb_midi: a couple indenting fixes Let's make things line up a little bit better. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/oss/sb_midi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/oss/sb_midi.c b/sound/oss/sb_midi.c index f139028e85c0..551ee7557b4e 100644 --- a/sound/oss/sb_midi.c +++ b/sound/oss/sb_midi.c @@ -179,14 +179,14 @@ void sb_dsp_midi_init(sb_devc * devc, struct module *owner) { printk(KERN_WARNING "Sound Blaster: failed to allocate MIDI memory.\n"); sound_unload_mididev(dev); - return; + return; } memcpy((char *) midi_devs[dev], (char *) &sb_midi_operations, sizeof(struct midi_operations)); if (owner) - midi_devs[dev]->owner = owner; - + midi_devs[dev]->owner = owner; + midi_devs[dev]->devc = devc; From 7f788e0cc07dba7f4eee6ffea30edee3af86e2a5 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:32:45 +0300 Subject: [PATCH 067/162] ALSA: azt3328: some indenting cleanups A few minor tweaks to make things line up correctly. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/azt3328.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index a40a2b4c8fd7..33b2a0af1b59 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1385,8 +1385,8 @@ snd_azf3328_ctrl_codec_activity(struct snd_azf3328 *chip, .running) && (!chip->codecs[peer_codecs[codec_type].other2] .running)); - } - if (call_function) + } + if (call_function) snd_azf3328_ctrl_enable_codecs(chip, enable); /* ...and adjust clock, too @@ -2126,7 +2126,8 @@ static struct snd_pcm_ops snd_azf3328_i2s_out_ops = { static int snd_azf3328_pcm(struct snd_azf3328 *chip) { -enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; /* pcm devices */ + /* pcm devices */ + enum { AZF_PCMDEV_STD, AZF_PCMDEV_I2S_OUT, NUM_AZF_PCMDEVS }; struct snd_pcm *pcm; int err; From 9603cded0e2cef003a822985d84b5daff1c7232f Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 25 Feb 2015 16:33:57 +0300 Subject: [PATCH 068/162] ALSA: cmipci: remove a stray space character Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/cmipci.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 1d0f2cad2f5a..6cf464d9043d 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2062,7 +2062,7 @@ static int snd_cmipci_get_volume(struct snd_kcontrol *kcontrol, val = (snd_cmipci_mixer_read(cm, reg.right_reg) >> reg.right_shift) & reg.mask; if (reg.invert) val = reg.mask - val; - ucontrol->value.integer.value[1] = val; + ucontrol->value.integer.value[1] = val; } spin_unlock_irq(&cm->reg_lock); return 0; From 5bb400ce4a9b100a2dd3f5db17c4c76877cecade Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Wed, 25 Feb 2015 21:37:52 +0530 Subject: [PATCH 069/162] ASoC: Intel: wrap runtime_pm usage count under CONFIG_PM The struct dev_pm_ops defines usage_count only when CONFIG_PM is defined. So we should use this variable only in cases where this falg is true. So we define a local variable and read the value under this flag. In non PM cases, we set this to 1. Reported-by: kbuild test robot Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/sst/sst_drv_interface.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/sst/sst_drv_interface.c b/sound/soc/intel/sst/sst_drv_interface.c index 549af7d7f6d0..f0e4b99b3aeb 100644 --- a/sound/soc/intel/sst/sst_drv_interface.c +++ b/sound/soc/intel/sst/sst_drv_interface.c @@ -139,17 +139,23 @@ static int sst_power_control(struct device *dev, bool state) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); int ret = 0; + int usage_count = 0; + +#ifdef CONFIG_PM + usage_count = atomic_read(&dev->power.usage_count); +#else + usage_count = 1; +#endif if (state == true) { ret = pm_runtime_get_sync(dev); - dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", - atomic_read(&dev->power.usage_count)); + + dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); return ret; } - if ((ctx->sst_state == SST_RESET) && - (atomic_read(&dev->power.usage_count) == 1)) { + if ((ctx->sst_state == SST_RESET) && (usage_count == 1)) { ret = sst_load_fw(ctx); if (ret) { dev_err(dev, "FW download fail %d\n", ret); @@ -158,8 +164,7 @@ static int sst_power_control(struct device *dev, bool state) } } } else { - dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", - atomic_read(&dev->power.usage_count)); + dev_dbg(ctx->dev, "Disable: pm usage count: %d\n", usage_count); return sst_pm_runtime_put(ctx); } return ret; From 92b2ad2c9e18ca2bfa8727af7edcd372d9acaac4 Mon Sep 17 00:00:00 2001 From: Kenneth Westfield Date: Tue, 24 Feb 2015 22:39:04 -0800 Subject: [PATCH 070/162] ASoC: max98357a: Use standard DAI names Use the standard naming convention for the codec DAI. Signed-off-by: Kenneth Westfield Signed-off-by: Mark Brown --- sound/soc/codecs/max98357a.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index 4ee23fbc4e12..bf3e933ee895 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -85,9 +85,9 @@ static struct snd_soc_dai_ops max98357a_dai_ops = { }; static struct snd_soc_dai_driver max98357a_dai_driver = { - .name = "max98357a", + .name = "HiFi", .playback = { - .stream_name = "max98357a-playback", + .stream_name = "HiFi Playback", .formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32, From bb573928e187fc5b1f91c3a7684791d5dfcca640 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 09:26:04 +0100 Subject: [PATCH 071/162] ALSA: hda - Drop power_save value indirection in hda_bus We used to pass the power_save option value to hda_bus via a given pointer. This was needed to refer to the value from the HD-audio core side. However, after the transition to the runtime PM, this is no longer needed. This patch drops the power_save value indirection in hda_bus above, and let the controller driver reprograms the autosuspend value explicitly by a new helper, snd_hda_set_power_save(). Without this call, the HD-audio core doesn't set up the autosuspend and flip the runtime PM. (User may still be able to set up via sysfs, though.) Along with this change, the pointer argument of azx_bus_create() is dropped as well. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 34 ++++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 5 ++--- sound/pci/hda/hda_controller.c | 5 +---- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 10 ++++------ sound/pci/hda/hda_tegra.c | 5 +++-- 6 files changed, 29 insertions(+), 32 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d0dbc62c9147..36cebe0e76b2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1175,10 +1175,8 @@ static int snd_hda_codec_dev_register(struct snd_device *device) struct hda_codec *codec = device->device_data; snd_hda_register_beep_device(codec); - if (device_is_registered(hda_codec_dev(codec))) { - snd_hda_power_sync(codec); + if (device_is_registered(hda_codec_dev(codec))) pm_runtime_enable(hda_codec_dev(codec)); - } return 0; } @@ -4778,21 +4776,10 @@ void snd_hda_power_down(struct hda_codec *codec) } EXPORT_SYMBOL_GPL(snd_hda_power_down); -/** - * snd_hda_power_sync - Synchronize the power_save option - * @codec: HD-audio codec - * - * Synchronize the runtime PM autosuspend state from the power_save option. - */ -void snd_hda_power_sync(struct hda_codec *codec) +static void codec_set_power_save(struct hda_codec *codec, int delay) { struct device *dev = hda_codec_dev(codec); - int delay; - if (!codec->bus->power_save) - return; - - delay = *codec->bus->power_save * 1000; if (delay > 0) { pm_runtime_set_autosuspend_delay(dev, delay); pm_runtime_use_autosuspend(dev); @@ -4804,7 +4791,22 @@ void snd_hda_power_sync(struct hda_codec *codec) pm_runtime_forbid(dev); } } -EXPORT_SYMBOL_GPL(snd_hda_power_sync); + +/** + * snd_hda_set_power_save - reprogram autosuspend for the given delay + * @bus: HD-audio bus + * @delay: autosuspend delay in msec, 0 = off + * + * Synchronize the runtime PM autosuspend state from the power_save option. + */ +void snd_hda_set_power_save(struct hda_bus *bus, int delay) +{ + struct hda_codec *c; + + list_for_each_entry(c, &bus->codec_list, list) + codec_set_power_save(c, delay); +} +EXPORT_SYMBOL_GPL(snd_hda_set_power_save); /** * snd_hda_check_amp_list_power - Check the amp list and update the power diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 593956fc384b..89908f5b9601 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -122,7 +122,6 @@ struct hda_bus { void *private_data; struct pci_dev *pci; const char *modelname; - int *power_save; struct hda_bus_ops ops; /* codec linked list */ @@ -592,12 +591,12 @@ const char *snd_hda_get_jack_location(u32 cfg); #ifdef CONFIG_PM void snd_hda_power_up(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); -void snd_hda_power_sync(struct hda_codec *codec); +void snd_hda_set_power_save(struct hda_bus *bus, int delay); void snd_hda_update_power_acct(struct hda_codec *codec); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} -static inline void snd_hda_power_sync(struct hda_codec *codec) {} +static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} #endif #ifdef CONFIG_SND_HDA_PATCH_LOADER diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 522c54f94740..cfe2c55296b6 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1838,7 +1838,7 @@ static struct hda_bus_ops bus_ops = { }; /* HD-audio bus initialization */ -int azx_bus_create(struct azx *chip, const char *model, int *power_save_to) +int azx_bus_create(struct azx *chip, const char *model) { struct hda_bus *bus; int err; @@ -1852,9 +1852,6 @@ int azx_bus_create(struct azx *chip, const char *model, int *power_save_to) bus->pci = chip->pci; bus->modelname = model; bus->ops = bus_ops; -#ifdef CONFIG_PM - bus->power_save = power_save_to; -#endif if (chip->driver_caps & AZX_DCAPS_RIRB_DELAY) { dev_dbg(chip->card->dev, "Enable delay in RIRB handling\n"); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 0d09aa6b49ac..e4f46a21698a 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -432,7 +432,7 @@ void azx_enter_link_reset(struct azx *chip); irqreturn_t azx_interrupt(int irq, void *dev_id); /* Codec interface */ -int azx_bus_create(struct azx *chip, const char *model, int *power_save_to); +int azx_bus_create(struct azx *chip, const char *model); int azx_probe_codecs(struct azx *chip, unsigned int max_slots); int azx_codec_configure(struct azx *chip); int azx_init_stream(struct azx *chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 26510e60cbe2..40540048b002 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -173,7 +173,6 @@ static struct kernel_param_ops param_ops_xint = { #define param_check_xint param_check_int static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; -static int *power_save_addr = &power_save; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); @@ -186,7 +185,7 @@ static bool power_save_controller = 1; module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #else -static int *power_save_addr; +#define power_save 0 #endif /* CONFIG_PM */ static int align_buffer_size = -1; @@ -740,7 +739,6 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) { struct hda_intel *hda; struct azx *chip; - struct hda_codec *c; int prev = power_save; int ret = param_set_int(val, kp); @@ -752,8 +750,7 @@ static int param_set_xint(const char *val, const struct kernel_param *kp) chip = &hda->chip; if (!chip->bus || chip->disabled) continue; - list_for_each_entry(c, &chip->bus->codec_list, list) - snd_hda_power_sync(c); + snd_hda_set_power_save(chip->bus, power_save * 1000); } mutex_unlock(&card_list_lock); return 0; @@ -1889,7 +1886,7 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_bus_create(chip, model[dev], power_save_addr); + err = azx_bus_create(chip, model[dev]); if (err < 0) goto out_free; @@ -1933,6 +1930,7 @@ static int azx_probe_continue(struct azx *chip) power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); + snd_hda_set_power_save(chip->bus, power_save * 1000); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index f6949e413a50..42bc17655df0 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -81,7 +81,7 @@ module_param(power_save, bint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout (in seconds, 0 = disable)."); #else -static int power_save = 0; +#define power_save 0 #endif /* @@ -496,7 +496,7 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; /* create codec instances */ - err = azx_bus_create(chip, NULL, &power_save); + err = azx_bus_create(chip, NULL); if (err < 0) goto out_free; @@ -525,6 +525,7 @@ static int hda_tegra_probe(struct platform_device *pdev) chip->running = 1; power_down_all_codecs(chip); azx_notifier_register(chip); + snd_hda_set_power_save(chip->bus, power_save * 1000); return 0; From 55ed9cd1feee80764937913afe760161b86cfb11 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 19 Feb 2015 17:35:32 +0100 Subject: [PATCH 072/162] ALSA: hda - Replace bus pm_notify with the standard runtime PM framework Now the final bit of runtime PM cleanup: instead of manual notification of the power up/down of the codec via hda_bus pm_notify ops, use the standard runtime PM feature. The child codec device will kick off the runtime PM of the parent (PCI) device upon suspend/resume automatically. For managing whether the link can be really turned off, we use the bit flags bus->codec_powered instead of the earlier bus->power_keep_link_on. flag. Each codec driver is responsible to set/clear the bit flag, and the controller device can be turned off only when all these bits are cleared. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 33 ++++++++------------------------- sound/pci/hda/hda_codec.h | 7 +------ sound/pci/hda/hda_controller.c | 19 ------------------- sound/pci/hda/hda_intel.c | 5 ++--- sound/pci/hda/patch_sigmatel.c | 4 +++- 5 files changed, 14 insertions(+), 54 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 36cebe0e76b2..33b8b71f8eaf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -46,23 +46,9 @@ #define codec_in_pm(codec) atomic_read(&(codec)->in_pm) #define hda_codec_is_power_on(codec) \ (!pm_runtime_suspended(hda_codec_dev(codec))) - -static void hda_call_pm_notify(struct hda_codec *codec, bool power_up) -{ - struct hda_bus *bus = codec->bus; - - if ((power_up && codec->pm_up_notified) || - (!power_up && !codec->pm_up_notified)) - return; - if (bus->ops.pm_notify) - bus->ops.pm_notify(bus, power_up); - codec->pm_up_notified = power_up; -} - #else #define codec_in_pm(codec) 0 #define hda_codec_is_power_on(codec) 1 -#define hda_call_pm_notify(codec, state) {} #endif /** @@ -1152,7 +1138,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) snd_array_free(&codec->spdif_out); remove_conn_list(codec); codec->bus->caddr_tbl[codec->addr] = NULL; - hda_call_pm_notify(codec, false); /* cancel leftover refcounts */ + clear_bit(codec->addr, &codec->bus->codec_powered); snd_hda_sysfs_clear(codec); free_hda_cache(&codec->amp_cache); free_hda_cache(&codec->cmd_cache); @@ -1277,10 +1263,10 @@ int snd_hda_codec_new(struct hda_bus *bus, * the caller has to power down appropriatley after initialization * phase. */ + set_bit(codec->addr, &bus->codec_powered); pm_runtime_set_active(hda_codec_dev(codec)); pm_runtime_get_noresume(hda_codec_dev(codec)); codec->power_jiffies = jiffies; - hda_call_pm_notify(codec, true); #endif snd_hda_sysfs_init(codec); @@ -1340,11 +1326,6 @@ int snd_hda_codec_new(struct hda_bus *bus, #endif codec->epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); -#ifdef CONFIG_PM - if (!codec->d3_stop_clk || !codec->epss) - bus->power_keep_link_on = 1; -#endif - /* power-up all before initialization */ hda_set_power_state(codec, AC_PWRST_D0); @@ -3954,7 +3935,6 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_mark_cmd_cache_dirty(codec); codec->power_jiffies = jiffies; - hda_call_pm_notify(codec, true); hda_set_power_state(codec, AC_PWRST_D0); restore_shutup_pins(codec); @@ -3986,14 +3966,17 @@ static int hda_codec_runtime_suspend(struct device *dev) for (i = 0; i < codec->num_pcms; i++) snd_pcm_suspend_all(codec->pcm_info[i].pcm); state = hda_call_codec_suspend(codec); - if (!codec->bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) - hda_call_pm_notify(codec, false); + if (codec->d3_stop_clk && codec->epss && (state & AC_PWRST_CLK_STOP_OK)) + clear_bit(codec->addr, &codec->bus->codec_powered); return 0; } static int hda_codec_runtime_resume(struct device *dev) { - hda_call_codec_resume(dev_to_hda_codec(dev)); + struct hda_codec *codec = dev_to_hda_codec(dev); + + set_bit(codec->addr, &codec->bus->codec_powered); + hda_call_codec_resume(codec); pm_runtime_mark_last_busy(dev); return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 89908f5b9601..457fc589eb46 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -83,10 +83,6 @@ struct hda_bus_ops { struct hda_pcm *pcm); /* reset bus for retry verb */ void (*bus_reset)(struct hda_bus *bus); -#ifdef CONFIG_PM - /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_bus *bus, bool power_up); -#endif #ifdef CONFIG_SND_HDA_DSP_LOADER /* prepare DSP transfer */ int (*load_dsp_prepare)(struct hda_bus *bus, unsigned int format, @@ -150,10 +146,10 @@ struct hda_bus { unsigned int rirb_error:1; /* error in codec communication */ unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ - unsigned int power_keep_link_on:1; /* don't power off HDA link */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ int primary_dig_out_type; /* primary digital out PCM type */ + unsigned long codec_powered; /* bit flags of powered codecs */ }; /* @@ -372,7 +368,6 @@ struct hda_codec { unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ #ifdef CONFIG_PM unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ - unsigned int pm_up_notified:1; /* PM notified to controller */ atomic_t in_pm; /* suspend/resume being performed */ unsigned long power_on_acct; unsigned long power_off_acct; diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index cfe2c55296b6..789ca66c3094 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1785,22 +1785,6 @@ static void azx_bus_reset(struct hda_bus *bus) bus->in_reset = 0; } -#ifdef CONFIG_PM -/* power-up/down the controller */ -static void azx_power_notify(struct hda_bus *bus, bool power_up) -{ - struct azx *chip = bus->private_data; - - if (!azx_has_pm_runtime(chip)) - return; - - if (power_up) - pm_runtime_get_sync(chip->card->dev); - else - pm_runtime_put_sync(chip->card->dev); -} -#endif - static int get_jackpoll_interval(struct azx *chip) { int i; @@ -1827,9 +1811,6 @@ static struct hda_bus_ops bus_ops = { .get_response = azx_get_response, .attach_pcm = azx_attach_pcm_stream, .bus_reset = azx_bus_reset, -#ifdef CONFIG_PM - .pm_notify = azx_power_notify, -#endif #ifdef CONFIG_SND_HDA_DSP_LOADER .load_dsp_prepare = azx_load_dsp_prepare, .load_dsp_trigger = azx_load_dsp_trigger, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 40540048b002..738d332351d5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -918,7 +918,8 @@ static int azx_runtime_idle(struct device *dev) if (chip->disabled || hda->init_failed) return 0; - if (!power_save_controller || !azx_has_pm_runtime(chip)) + if (!power_save_controller || !azx_has_pm_runtime(chip) || + chip->bus->codec_powered) return -EBUSY; return 0; @@ -1084,7 +1085,6 @@ static int azx_free(struct azx *chip) azx_stop_chip(chip); } - pci->dev.power.ignore_children = 0; /* FIXME */ if (chip->irq >= 0) free_irq(chip->irq, (void*)chip); if (chip->msi) @@ -1794,7 +1794,6 @@ static int azx_probe(struct pci_dev *pci, return err; } - pci->dev.power.ignore_children = 1; /* FIXME */ err = azx_create(card, pci, dev, pci_id->driver_data, &pci_hda_ops, &chip); if (err < 0) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6a2163056216..2956a6ba6bf0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2132,8 +2132,10 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ +#ifdef CONFIG_PM /* resetting controller clears GPIO, so we need to keep on */ - codec->bus->power_keep_link_on = 1; + codec->d3_stop_clk = 0; +#endif } } From 709949fbe9632941585dcacabc8a66010030ed10 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 09:58:14 +0100 Subject: [PATCH 073/162] ALSA: hda - Power down codec automatically at registration So far, we let the controller driver power down the all codecs at the end of probe. But this can be done better in the codec's dev_register callback. This results in the reduction of duplicated codes in each control driver. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 5 +++-- sound/pci/hda/hda_intel.c | 14 -------------- sound/pci/hda/hda_tegra.c | 12 ------------ 3 files changed, 3 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 33b8b71f8eaf..6580a367023e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1163,6 +1163,8 @@ static int snd_hda_codec_dev_register(struct snd_device *device) snd_hda_register_beep_device(codec); if (device_is_registered(hda_codec_dev(codec))) pm_runtime_enable(hda_codec_dev(codec)); + /* it was powered up in snd_hda_codec_new(), now all done */ + snd_hda_power_down(codec); return 0; } @@ -1260,8 +1262,7 @@ int snd_hda_codec_new(struct hda_bus *bus, #ifdef CONFIG_PM /* snd_hda_codec_new() marks the codec as power-up, and leave it as is. - * the caller has to power down appropriatley after initialization - * phase. + * it's powered down later in snd_hda_codec_dev_register(). */ set_bit(codec->addr, &bus->codec_powered); pm_runtime_set_active(hda_codec_dev(codec)); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 738d332351d5..e75e8137e296 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1604,19 +1604,6 @@ static int azx_first_init(struct azx *chip) return 0; } -static void power_down_all_codecs(struct azx *chip) -{ -#ifdef CONFIG_PM - /* The codecs were powered up in snd_hda_codec_new(). - * Now all initialization done, so turn them down if possible - */ - struct hda_codec *codec; - list_for_each_entry(codec, &chip->bus->codec_list, list) { - snd_hda_power_down(codec); - } -#endif -} - #ifdef CONFIG_SND_HDA_PATCH_LOADER /* callback from request_firmware_nowait() */ static void azx_firmware_cb(const struct firmware *fw, void *context) @@ -1926,7 +1913,6 @@ static int azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; - power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); snd_hda_set_power_save(chip->bus, power_save * 1000); diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 42bc17655df0..1359fdd20f02 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -337,17 +337,6 @@ static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) return 0; } -/* - * The codecs were powered up in snd_hda_codec_new(). - * Now all initialization done, so turn them down if possible - */ -static void power_down_all_codecs(struct azx *chip) -{ - struct hda_codec *codec; - list_for_each_entry(codec, &chip->bus->codec_list, list) - snd_hda_power_down(codec); -} - static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) { struct snd_card *card = chip->card; @@ -523,7 +512,6 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; chip->running = 1; - power_down_all_codecs(chip); azx_notifier_register(chip); snd_hda_set_power_save(chip->bus, power_save * 1000); From 777ae1946813f1dea24d908c8f318754f090f41f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2015 14:35:59 +0100 Subject: [PATCH 074/162] ALSA: hda - Set parent of input beep devices Set the card device as the parent like other sound devices instead of leaving it empty. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index d6be4e852c4d..e98438e95e79 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -160,6 +160,7 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->name = "HDA Digital PCBeep"; input_dev->phys = beep->phys; input_dev->id.bustype = BUS_PCI; + input_dev->dev.parent = &codec->bus->card->card_dev; input_dev->id.vendor = codec->vendor_id >> 16; input_dev->id.product = codec->vendor_id & 0xffff; From 7e40b80da452770878943edfe7da80f10f8d25da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 26 Feb 2015 09:42:04 +0100 Subject: [PATCH 075/162] ALSA: hda - Remove channel mode helper functions They are no longer used, let's kill them. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 82 --------------------------------------- sound/pci/hda/hda_local.h | 23 ----------- 2 files changed, 105 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6580a367023e..db86b446743c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4843,88 +4843,6 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); #endif -/* - * Channel mode helper - */ - -/** - * snd_hda_ch_mode_info - Info callback helper for the channel mode enum - * @codec: the HDA codec - * @uinfo: pointer to get/store the data - * @chmode: channel mode array - * @num_chmodes: channel mode array size - */ -int snd_hda_ch_mode_info(struct hda_codec *codec, - struct snd_ctl_elem_info *uinfo, - const struct hda_channel_mode *chmode, - int num_chmodes) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = num_chmodes; - if (uinfo->value.enumerated.item >= num_chmodes) - uinfo->value.enumerated.item = num_chmodes - 1; - sprintf(uinfo->value.enumerated.name, "%dch", - chmode[uinfo->value.enumerated.item].channels); - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_ch_mode_info); - -/** - * snd_hda_ch_mode_get - Get callback helper for the channel mode enum - * @codec: the HDA codec - * @ucontrol: pointer to get/store the data - * @chmode: channel mode array - * @num_chmodes: channel mode array size - * @max_channels: max number of channels - */ -int snd_hda_ch_mode_get(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int max_channels) -{ - int i; - - for (i = 0; i < num_chmodes; i++) { - if (max_channels == chmode[i].channels) { - ucontrol->value.enumerated.item[0] = i; - break; - } - } - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_ch_mode_get); - -/** - * snd_hda_ch_mode_put - Put callback helper for the channel mode enum - * @codec: the HDA codec - * @ucontrol: pointer to get/store the data - * @chmode: channel mode array - * @num_chmodes: channel mode array size - * @max_channelsp: pointer to store the max channels - */ -int snd_hda_ch_mode_put(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int *max_channelsp) -{ - unsigned int mode; - - mode = ucontrol->value.enumerated.item[0]; - if (mode >= num_chmodes) - return -EINVAL; - if (*max_channelsp == chmode[mode].channels) - return 0; - /* change the current channel setting */ - *max_channelsp = chmode[mode].channels; - if (chmode[mode].sequence) - snd_hda_sequence_write_cache(codec, chmode[mode].sequence); - return 1; -} -EXPORT_SYMBOL_GPL(snd_hda_ch_mode_put); - /* * input MUX helper */ diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 2f7d9646a41d..8588813163e3 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -272,29 +272,6 @@ int snd_hda_add_imux_item(struct hda_codec *codec, struct hda_input_mux *imux, const char *label, int index, int *type_index_ret); -/* - * Channel mode helper - */ -struct hda_channel_mode { - int channels; - const struct hda_verb *sequence; -}; - -int snd_hda_ch_mode_info(struct hda_codec *codec, - struct snd_ctl_elem_info *uinfo, - const struct hda_channel_mode *chmode, - int num_chmodes); -int snd_hda_ch_mode_get(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int max_channels); -int snd_hda_ch_mode_put(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol, - const struct hda_channel_mode *chmode, - int num_chmodes, - int *max_channelsp); - /* * Multi-channel / digital-out PCM helper */ From 820cc6cf2c552155ea919e596a85e1f4e5dfa2b5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Feb 2015 12:50:46 +0100 Subject: [PATCH 076/162] ALSA: hda - Clear pcm pointer assigned to hda_pcm at device removal We leave the pcm field of struct hda_pcm at removal of each device, so far. This hasn't been a problem since unbinding the codec driver isn't supposed to happen and another route via snd_hda_codec_reset() clears all the once. However, for a proper unbind implementation, we need to care about it. This patch does the thing above properly: - Include struct hda_pcm pointer instead of struct hda_pcm_stream pointers in struct azx_dev. This allows us to point the hda_pcm object at dev_free callback. - Introduce to_hda_pcm_stream() macro for better readability. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 22 +++++++++++++++------- sound/pci/hda/hda_controller.h | 2 +- 2 files changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 789ca66c3094..1695f0e2bd9d 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -258,11 +258,18 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } +static inline struct hda_pcm_stream * +to_hda_pcm_stream(struct snd_pcm_substream *substream) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + return &apcm->info->stream[substream->stream]; +} + static u64 azx_adjust_codec_delay(struct snd_pcm_substream *substream, u64 nsec) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); u64 codec_frames, codec_nsecs; if (!hinfo->ops.get_delay) @@ -398,7 +405,7 @@ static int azx_setup_periods(struct azx *chip, static int azx_pcm_close(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); struct azx *chip = apcm->chip; struct azx_dev *azx_dev = get_azx_dev(substream); unsigned long flags; @@ -440,7 +447,7 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx_dev *azx_dev = get_azx_dev(substream); struct azx *chip = apcm->chip; - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); int err; /* reset BDL address */ @@ -467,7 +474,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) struct azx_pcm *apcm = snd_pcm_substream_chip(substream); struct azx *chip = apcm->chip; struct azx_dev *azx_dev = get_azx_dev(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int bufsize, period_bytes, format_val, stream_tag; int err; @@ -707,7 +714,7 @@ unsigned int azx_get_position(struct azx *chip, if (substream->runtime) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); if (chip->get_delay[stream]) delay += chip->get_delay[stream](chip, azx_dev, pos); @@ -790,7 +797,7 @@ static struct snd_pcm_hardware azx_pcm_hw = { static int azx_pcm_open(struct snd_pcm_substream *substream) { struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; + struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); struct azx *chip = apcm->chip; struct azx_dev *azx_dev; struct snd_pcm_runtime *runtime = substream->runtime; @@ -904,6 +911,7 @@ static void azx_pcm_free(struct snd_pcm *pcm) struct azx_pcm *apcm = pcm->private_data; if (apcm) { list_del(&apcm->list); + apcm->info->pcm = NULL; kfree(apcm); } } @@ -940,6 +948,7 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, apcm->chip = chip; apcm->pcm = pcm; apcm->codec = codec; + apcm->info = cpcm; pcm->private_data = apcm; pcm->private_free = azx_pcm_free; if (cpcm->pcm_type == HDA_PCM_TYPE_MODEM) @@ -947,7 +956,6 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, list_add_tail(&apcm->list, &chip->pcm_list); cpcm->pcm = pcm; for (s = 0; s < 2; s++) { - apcm->hinfo[s] = &cpcm->stream[s]; if (cpcm->stream[s].substreams) snd_pcm_set_ops(pcm, s, &azx_pcm_ops); } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index e4f46a21698a..94c1a4719f7f 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -283,7 +283,7 @@ struct azx_pcm { struct azx *chip; struct snd_pcm *pcm; struct hda_codec *codec; - struct hda_pcm_stream *hinfo[2]; + struct hda_pcm *info; struct list_head list; }; From 223c055aa0eb7e606eb7132e339ce66bb8d7be0d Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 18 Dec 2014 11:32:52 +0800 Subject: [PATCH 077/162] ASoC: rt5670: set platform data by dmi This patch set specific data according to dmi data. Signed-off-by: Jin, Yao Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 8a0833de1665..cd47ef1f5561 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -2549,6 +2550,17 @@ static struct acpi_device_id rt5670_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5670_acpi_match); #endif +static const struct dmi_system_id dmi_platform_intel_braswell[] = { + { + .ident = "Intel Braswell", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "Braswell CRB"), + }, + }, + {} +}; + static int rt5670_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2568,6 +2580,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, if (pdata) rt5670->pdata = *pdata; + if (dmi_check_system(dmi_platform_intel_braswell)) { + rt5670->pdata.dmic_en = true; + rt5670->pdata.dmic1_data_pin = RT5670_DMIC_DATA_IN2P; + rt5670->pdata.jd_mode = 1; + } + rt5670->regmap = devm_regmap_init_i2c(i2c, &rt5670_regmap); if (IS_ERR(rt5670->regmap)) { ret = PTR_ERR(rt5670->regmap); From 64e89e5f55484d289c8b326521e5a12291e2283e Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 15 Dec 2014 15:42:33 +0800 Subject: [PATCH 078/162] ASoC: rt5670: Add runtime PM support This patch adds runtime PM support on rt5670 codec. Signed-off-by: Lin Mengdong Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index cd47ef1f5561..78d85de8af6f 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include @@ -2734,18 +2735,26 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5670, rt5670_dai, ARRAY_SIZE(rt5670_dai)); if (ret < 0) goto err; + pm_runtime_put(&i2c->dev); + return 0; err: + pm_runtime_disable(&i2c->dev); + return ret; } static int rt5670_i2c_remove(struct i2c_client *i2c) { + pm_runtime_disable(&i2c->dev); snd_soc_unregister_codec(&i2c->dev); return 0; From 77e3ea2801c8ca4700e6b17053b625b8a981ac77 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 15 Dec 2014 15:42:34 +0800 Subject: [PATCH 079/162] ASoC: rt5670: Keep sysclk on if JD func is used System clock is necessary for rt5670 JD function. We assume system clock source will be set in machine driver. So there are two things left we should do in codec driver. 1. Set sysclk to codec internal clock in probe since machine driver may not do that before JD function is registered. 2. Power up PLL once sysclk source is switched to PLL. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 78d85de8af6f..0a027bc94399 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2190,6 +2190,13 @@ static int rt5670_set_dai_sysclk(struct snd_soc_dai *dai, if (freq == rt5670->sysclk && clk_id == rt5670->sysclk_src) return 0; + if (rt5670->pdata.jd_mode) { + if (clk_id == RT5670_SCLK_S_PLL1) + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); + else + snd_soc_dapm_disable_pin(&codec->dapm, "PLL1"); + snd_soc_dapm_sync(&codec->dapm); + } switch (clk_id) { case RT5670_SCLK_S_MCLK: reg_val |= RT5670_SCLK_SRC_MCLK; @@ -2628,6 +2635,10 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, } if (rt5670->pdata.jd_mode) { + regmap_update_bits(rt5670->regmap, RT5670_GLB_CLK, + RT5670_SCLK_SRC_MASK, RT5670_SCLK_SRC_RCCLK); + rt5670->sysclk = 0; + rt5670->sysclk_src = RT5670_SCLK_S_RCCLK; regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG1, RT5670_PWR_MB, RT5670_PWR_MB); regmap_update_bits(rt5670->regmap, RT5670_PWR_ANLG2, From 3aebec3a701e70d6fe2816891e5abea066492779 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 7 Jan 2015 10:19:06 +0800 Subject: [PATCH 080/162] ASoC: rt5670: redefine ASRC control registers 0x84 and 0x85 The previous definition of registers 0x84 and 0x85 doesn't match the datasheet. So this patch removes the wrong definition and writes a new one for the two registers. Signed-off-by: Bard Liao Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.h | 65 +++++++++++++++------------------------ 1 file changed, 24 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 84857bdaa78b..82553b1726cd 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1023,50 +1023,33 @@ #define RT5670_DMIC_2_M_NOR (0x0 << 8) #define RT5670_DMIC_2_M_ASYN (0x1 << 8) +/* ASRC clock source selection (0x84, 0x85) */ +#define RT5670_CLK_SEL_SYS (0x0) +#define RT5670_CLK_SEL_I2S1_ASRC (0x1) +#define RT5670_CLK_SEL_I2S2_ASRC (0x2) +#define RT5670_CLK_SEL_I2S3_ASRC (0x3) +#define RT5670_CLK_SEL_SYS2 (0x5) +#define RT5670_CLK_SEL_SYS3 (0x6) + /* ASRC Control 2 (0x84) */ -#define RT5670_MDA_L_M_MASK (0x1 << 15) -#define RT5670_MDA_L_M_SFT 15 -#define RT5670_MDA_L_M_NOR (0x0 << 15) -#define RT5670_MDA_L_M_ASYN (0x1 << 15) -#define RT5670_MDA_R_M_MASK (0x1 << 14) -#define RT5670_MDA_R_M_SFT 14 -#define RT5670_MDA_R_M_NOR (0x0 << 14) -#define RT5670_MDA_R_M_ASYN (0x1 << 14) -#define RT5670_MAD_L_M_MASK (0x1 << 13) -#define RT5670_MAD_L_M_SFT 13 -#define RT5670_MAD_L_M_NOR (0x0 << 13) -#define RT5670_MAD_L_M_ASYN (0x1 << 13) -#define RT5670_MAD_R_M_MASK (0x1 << 12) -#define RT5670_MAD_R_M_SFT 12 -#define RT5670_MAD_R_M_NOR (0x0 << 12) -#define RT5670_MAD_R_M_ASYN (0x1 << 12) -#define RT5670_ADC_M_MASK (0x1 << 11) -#define RT5670_ADC_M_SFT 11 -#define RT5670_ADC_M_NOR (0x0 << 11) -#define RT5670_ADC_M_ASYN (0x1 << 11) -#define RT5670_STO_DAC_M_MASK (0x1 << 5) -#define RT5670_STO_DAC_M_SFT 5 -#define RT5670_STO_DAC_M_NOR (0x0 << 5) -#define RT5670_STO_DAC_M_ASYN (0x1 << 5) -#define RT5670_I2S1_R_D_MASK (0x1 << 4) -#define RT5670_I2S1_R_D_SFT 4 -#define RT5670_I2S1_R_D_DIS (0x0 << 4) -#define RT5670_I2S1_R_D_EN (0x1 << 4) -#define RT5670_I2S2_R_D_MASK (0x1 << 3) -#define RT5670_I2S2_R_D_SFT 3 -#define RT5670_I2S2_R_D_DIS (0x0 << 3) -#define RT5670_I2S2_R_D_EN (0x1 << 3) -#define RT5670_PRE_SCLK_MASK (0x3) -#define RT5670_PRE_SCLK_SFT 0 -#define RT5670_PRE_SCLK_512 (0x0) -#define RT5670_PRE_SCLK_1024 (0x1) -#define RT5670_PRE_SCLK_2048 (0x2) +#define RT5670_DA_STO_CLK_SEL_MASK (0xf << 12) +#define RT5670_DA_STO_CLK_SEL_SFT 12 +#define RT5670_DA_MONOL_CLK_SEL_MASK (0xf << 8) +#define RT5670_DA_MONOL_CLK_SEL_SFT 8 +#define RT5670_DA_MONOR_CLK_SEL_MASK (0xf << 4) +#define RT5670_DA_MONOR_CLK_SEL_SFT 4 +#define RT5670_AD_STO1_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_STO1_CLK_SEL_SFT 0 /* ASRC Control 3 (0x85) */ -#define RT5670_I2S1_RATE_MASK (0xf << 12) -#define RT5670_I2S1_RATE_SFT 12 -#define RT5670_I2S2_RATE_MASK (0xf << 8) -#define RT5670_I2S2_RATE_SFT 8 +#define RT5670_UP_CLK_SEL_MASK (0xf << 12) +#define RT5670_UP_CLK_SEL_SFT 12 +#define RT5670_DOWN_CLK_SEL_MASK (0xf << 8) +#define RT5670_DOWN_CLK_SEL_SFT 8 +#define RT5670_AD_MONOL_CLK_SEL_MASK (0xf << 4) +#define RT5670_AD_MONOL_CLK_SEL_SFT 4 +#define RT5670_AD_MONOR_CLK_SEL_MASK (0xf << 0) +#define RT5670_AD_MONOR_CLK_SEL_SFT 0 /* ASRC Control 4 (0x89) */ #define RT5670_I2S1_PD_MASK (0x7 << 12) From ea232b3f7233f9242e5a1287377fedb6972dfea4 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Wed, 7 Jan 2015 10:19:12 +0800 Subject: [PATCH 081/162] ASoC: rt5670: add API to select ASRC clock source When codec is in slave mode, ASRC can suppress noise for asynchronous MCLK and LRCLK or special I2S format. This patch defines an API to select the clock source for specified filters. And the codec driver will turn on ASRC for these filters if ASRC is selected as their clock source. Signed-off-by: Bard Liao Signed-off-by: Mengdong Lin Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 83 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5670.h | 15 +++++++ 2 files changed, 98 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0a027bc94399..0632b7458a53 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -590,6 +590,89 @@ static int can_use_asrc(struct snd_soc_dapm_widget *source, return 0; } + +/** + * rt5670_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5670 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + unsigned int asrc2_mask = 0, asrc2_value = 0; + unsigned int asrc3_mask = 0, asrc3_value = 0; + + if (clk_src > RT5670_CLK_SEL_SYS3) + return -EINVAL; + + if (filter_mask & RT5670_DA_STEREO_FILTER) { + asrc2_mask |= RT5670_DA_STO_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_STO_CLK_SEL_MASK) + | (clk_src << RT5670_DA_STO_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DA_MONO_L_FILTER) { + asrc2_mask |= RT5670_DA_MONOL_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_MONOL_CLK_SEL_MASK) + | (clk_src << RT5670_DA_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DA_MONO_R_FILTER) { + asrc2_mask |= RT5670_DA_MONOR_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_DA_MONOR_CLK_SEL_MASK) + | (clk_src << RT5670_DA_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_STEREO_FILTER) { + asrc2_mask |= RT5670_AD_STO1_CLK_SEL_MASK; + asrc2_value = (asrc2_value & ~RT5670_AD_STO1_CLK_SEL_MASK) + | (clk_src << RT5670_AD_STO1_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_MONO_L_FILTER) { + asrc3_mask |= RT5670_AD_MONOL_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_AD_MONOL_CLK_SEL_MASK) + | (clk_src << RT5670_AD_MONOL_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_AD_MONO_R_FILTER) { + asrc3_mask |= RT5670_AD_MONOR_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_AD_MONOR_CLK_SEL_MASK) + | (clk_src << RT5670_AD_MONOR_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_UP_RATE_FILTER) { + asrc3_mask |= RT5670_UP_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_UP_CLK_SEL_MASK) + | (clk_src << RT5670_UP_CLK_SEL_SFT); + } + + if (filter_mask & RT5670_DOWN_RATE_FILTER) { + asrc3_mask |= RT5670_DOWN_CLK_SEL_MASK; + asrc3_value = (asrc3_value & ~RT5670_DOWN_CLK_SEL_MASK) + | (clk_src << RT5670_DOWN_CLK_SEL_SFT); + } + + if (asrc2_mask) + snd_soc_update_bits(codec, RT5670_ASRC_2, + asrc2_mask, asrc2_value); + + if (asrc3_mask) + snd_soc_update_bits(codec, RT5670_ASRC_3, + asrc3_mask, asrc3_value); + return 0; +} +EXPORT_SYMBOL_GPL(rt5670_sel_asrc_clk_src); + /* Digital Mixer */ static const struct snd_kcontrol_new rt5670_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5670_STO1_ADC_MIXER, diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 82553b1726cd..0a67adbcfbc3 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1969,6 +1969,21 @@ enum { RT5670_DMIC_DATA_GPIO5, }; +/* filter mask */ +enum { + RT5670_DA_STEREO_FILTER = 0x1, + RT5670_DA_MONO_L_FILTER = (0x1 << 1), + RT5670_DA_MONO_R_FILTER = (0x1 << 2), + RT5670_AD_STEREO_FILTER = (0x1 << 3), + RT5670_AD_MONO_L_FILTER = (0x1 << 4), + RT5670_AD_MONO_R_FILTER = (0x1 << 5), + RT5670_UP_RATE_FILTER = (0x1 << 6), + RT5670_DOWN_RATE_FILTER = (0x1 << 7), +}; + +int rt5670_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + struct rt5670_priv { struct snd_soc_codec *codec; struct rt5670_platform_data pdata; From ab1f70952f61504f60805f13660c8740adcbe14f Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 11 Feb 2015 19:18:51 +0800 Subject: [PATCH 082/162] ASoC: rt5677: Add the chip type to distinguish the setting of the clock source There is only one clock source in the rt5676, so the chip type is added to distinguish the setting of the clock source in the VAD function. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 28 ++++++++++++++++++++++------ sound/soc/codecs/rt5677.h | 6 ++++++ 2 files changed, 28 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 5d0bb8748dd1..ab62777dbd33 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -718,11 +718,24 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) RT5677_LDO1_SEL_MASK, 0x0); regmap_update_bits(rt5677->regmap, RT5677_PWR_ANLG2, RT5677_PWR_LDO1, RT5677_PWR_LDO1); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, - RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); - regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, - RT5677_PLL2_PR_SRC_MASK | RT5677_DSP_CLK_SRC_MASK, - RT5677_PLL2_PR_SRC_MCLK2 | RT5677_DSP_CLK_SRC_BYPASS); + switch (rt5677->type) { + case RT5677: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK1, + RT5677_MCLK_SRC_MASK, RT5677_MCLK2_SRC); + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_PLL2_PR_SRC_MASK | + RT5677_DSP_CLK_SRC_MASK, + RT5677_PLL2_PR_SRC_MCLK2 | + RT5677_DSP_CLK_SRC_BYPASS); + break; + case RT5676: + regmap_update_bits(rt5677->regmap, RT5677_GLB_CLK2, + RT5677_DSP_CLK_SRC_MASK, + RT5677_DSP_CLK_SRC_BYPASS); + break; + default: + break; + } regmap_write(rt5677->regmap, RT5677_PWR_DSP2, 0x07ff); regmap_write(rt5677->regmap, RT5677_PWR_DSP1, 0x07fd); rt5677_set_dsp_mode(codec, true); @@ -4733,7 +4746,8 @@ static const struct regmap_config rt5677_regmap = { }; static const struct i2c_device_id rt5677_i2c_id[] = { - { "rt5677", 0 }, + { "rt5677", RT5677 }, + { "rt5676", RT5676 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5677_i2c_id); @@ -4850,6 +4864,8 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5677); + rt5677->type = id->driver_data; + if (pdata) rt5677->pdata = *pdata; diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index c0a625f290cc..07df96b43f59 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1665,6 +1665,11 @@ enum { RT5677_IRQ_JD3, }; +enum rt5677_type { + RT5677, + RT5676, +}; + struct rt5677_priv { struct snd_soc_codec *codec; struct rt5677_platform_data pdata; @@ -1681,6 +1686,7 @@ struct rt5677_priv { int pll_in; int pll_out; int pow_ldo2; /* POW_LDO2 pin */ + enum rt5677_type type; #ifdef CONFIG_GPIOLIB struct gpio_chip gpio_chip; #endif From cbca4076d156c93cedadabb0e463ba0db16bb166 Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Wed, 25 Feb 2015 17:36:14 +0800 Subject: [PATCH 083/162] ASoC: rt5677: Keep the LDO2 powered while used in the suspend mode The patch keeps the ldo2 power while the DSP function of "Voice Wake Up" used in the suspend mode. Signed-off-by: Oder Chiou Signed-off-by: Mark Brown --- sound/soc/codecs/rt5677.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index ab62777dbd33..5ff7ffaec5cc 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4513,10 +4513,10 @@ static int rt5677_suspend(struct snd_soc_codec *codec) if (!rt5677->dsp_vad_en) { regcache_cache_only(rt5677->regmap, true); regcache_mark_dirty(rt5677->regmap); - } - if (gpio_is_valid(rt5677->pow_ldo2)) - gpio_set_value_cansleep(rt5677->pow_ldo2, 0); + if (gpio_is_valid(rt5677->pow_ldo2)) + gpio_set_value_cansleep(rt5677->pow_ldo2, 0); + } return 0; } @@ -4525,12 +4525,12 @@ static int rt5677_resume(struct snd_soc_codec *codec) { struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - if (gpio_is_valid(rt5677->pow_ldo2)) { - gpio_set_value_cansleep(rt5677->pow_ldo2, 1); - msleep(10); - } - if (!rt5677->dsp_vad_en) { + if (gpio_is_valid(rt5677->pow_ldo2)) { + gpio_set_value_cansleep(rt5677->pow_ldo2, 1); + msleep(10); + } + regcache_cache_only(rt5677->regmap, false); regcache_sync(rt5677->regmap); } From a0cf43e2f0f391dad7882febbf04423e73e3ff99 Mon Sep 17 00:00:00 2001 From: Tomeu Vizoso Date: Thu, 12 Feb 2015 09:41:55 +0100 Subject: [PATCH 084/162] ASoC: tegra: Expose Headphones pin to userspace So userspace can enable or disable it based on the current policy. Signed-off-by: Tomeu Vizoso Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_max98090.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index af3fb997b752..8df71a436f11 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -136,6 +136,7 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { }; static const struct snd_kcontrol_new tegra_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), }; From 3a4562f756617b4b210fc487bfe23853a450d3c1 Mon Sep 17 00:00:00 2001 From: Tomeu Vizoso Date: Thu, 12 Feb 2015 09:41:56 +0100 Subject: [PATCH 085/162] ASoC: tegra: Add sink for the internal mic to tegra_max98090 Also adds a control for the pin of the internal mic, so userspace can apply policy when the state of the external mic jack changes. Signed-off-by: Tomeu Vizoso Acked-by: Stephen Warren Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + sound/soc/tegra/tegra_max98090.c | 2 ++ 2 files changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt index c949abc2992f..c3495beba358 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt @@ -18,6 +18,7 @@ Required properties: * Headphones * Speakers * Mic Jack + * Int Mic - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 8df71a436f11..29ea87cd852e 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -133,11 +133,13 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), }; static const struct snd_kcontrol_new tegra_max98090_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Int Mic"), }; static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) From dd3001490834e10615d9eb229b3e9bbcc0070541 Mon Sep 17 00:00:00 2001 From: Tomeu Vizoso Date: Thu, 12 Feb 2015 09:41:57 +0100 Subject: [PATCH 086/162] ASoC: tegra: Add control for the Mic Jack pin So userspace can enable and disable the external microphone. Signed-off-by: Tomeu Vizoso Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_max98090.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 29ea87cd852e..1f20c2c40a5a 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -139,6 +139,7 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { static const struct snd_kcontrol_new tegra_max98090_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), SOC_DAPM_PIN_SWITCH("Int Mic"), }; From 1a4ba30cced3002add8459eadcd65b8d3cd1515e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 24 Feb 2015 11:50:11 +0100 Subject: [PATCH 087/162] ALSA: hda - Split snd_hda_build_pcms() snd_hda_build_pcms() does actually three things: let the codec driver build up hda_pcm list, set the PCM default values, and call the attach_pcm bus ops for each hda_pcm instance. The former two are basically independent from the bus implementation, so it'd make the code a bit more readable. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 102 ++++++++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 1 + 2 files changed, 60 insertions(+), 43 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index db86b446743c..40300fcc5308 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4569,71 +4569,87 @@ static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) return -EAGAIN; } -/* - * attach a new PCM stream - */ -static int snd_hda_attach_pcm(struct hda_codec *codec, struct hda_pcm *pcm) +/* call build_pcms ops of the given codec and set up the default parameters */ +int snd_hda_codec_parse_pcms(struct hda_codec *codec) { - struct hda_bus *bus = codec->bus; - struct hda_pcm_stream *info; - int stream, err; + unsigned int pcm; + int err; - if (snd_BUG_ON(!pcm->name)) - return -EINVAL; - for (stream = 0; stream < 2; stream++) { - info = &pcm->stream[stream]; - if (info->substreams) { + if (codec->num_pcms) + return 0; /* already parsed */ + + if (!codec->patch_ops.build_pcms) + return 0; + + err = codec->patch_ops.build_pcms(codec); + if (err < 0) { + codec_err(codec, "cannot build PCMs for #%d (error %d)\n", + codec->addr, err); + return err; + } + + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + int stream; + + for (stream = 0; stream < 2; stream++) { + struct hda_pcm_stream *info = &cpcm->stream[stream]; + + if (!info->substreams) + continue; + if (snd_BUG_ON(!cpcm->name)) + return -EINVAL; err = set_pcm_default_values(codec, info); - if (err < 0) + if (err < 0) { + codec_warn(codec, + "fail to setup default for PCM %s\n", + cpcm->name); return err; + } } } - return bus->ops.attach_pcm(bus, codec, pcm); + + return 0; } /* assign all PCMs of the given codec */ int snd_hda_codec_build_pcms(struct hda_codec *codec) { + struct hda_bus *bus = codec->bus; unsigned int pcm; - int err; + int dev, err; - if (!codec->num_pcms) { - if (!codec->patch_ops.build_pcms) - return 0; - err = codec->patch_ops.build_pcms(codec); - if (err < 0) { - codec_err(codec, - "cannot build PCMs for #%d (error %d)\n", - codec->addr, err); - err = snd_hda_codec_reset(codec); - if (err < 0) { - codec_err(codec, - "cannot revert codec\n"); - return err; - } - } + if (snd_BUG_ON(!bus->ops.attach_pcm)) + return -EINVAL; + + err = snd_hda_codec_parse_pcms(codec); + if (err < 0) { + snd_hda_codec_reset(codec); + return err; } + + /* attach a new PCM streams */ for (pcm = 0; pcm < codec->num_pcms; pcm++) { struct hda_pcm *cpcm = &codec->pcm_info[pcm]; - int dev; + if (cpcm->pcm) + continue; /* already attached */ if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) continue; /* no substreams assigned */ - if (!cpcm->pcm) { - dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type); - if (dev < 0) - continue; /* no fatal error */ - cpcm->device = dev; - err = snd_hda_attach_pcm(codec, cpcm); - if (err < 0) { - codec_err(codec, - "cannot attach PCM stream %d for codec #%d\n", - dev, codec->addr); - continue; /* no fatal error */ - } + dev = get_empty_pcm_device(bus, cpcm->pcm_type); + if (dev < 0) + continue; /* no fatal error */ + cpcm->device = dev; + err = bus->ops.attach_pcm(bus, codec, cpcm); + if (err < 0) { + codec_err(codec, + "cannot attach PCM stream %d for codec #%d\n", + dev, codec->addr); + continue; /* no fatal error */ } } + return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 457fc589eb46..8cf70369cd10 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -517,6 +517,7 @@ int snd_hda_codec_build_controls(struct hda_codec *codec); * PCM */ int snd_hda_build_pcms(struct hda_bus *bus); +int snd_hda_codec_parse_pcms(struct hda_codec *codec); int snd_hda_codec_build_pcms(struct hda_codec *codec); int snd_hda_codec_prepare(struct hda_codec *codec, From 6efdd8513f182492c21fb7238592d4539d5c751a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 16:09:22 +0100 Subject: [PATCH 088/162] ALSA: hda - Add card field to hda_codec struct Allow the codec object to have an individual card pointer. Not only this simplifies the redirections in many places, also this will allow us to make each codec assigned to a different card object. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 4 ++-- sound/pci/hda/hda_bind.c | 6 +++--- sound/pci/hda/hda_codec.c | 30 +++++++++++++++--------------- sound/pci/hda/hda_codec.h | 5 +++-- sound/pci/hda/hda_controller.c | 2 +- sound/pci/hda/hda_hwdep.c | 2 +- sound/pci/hda/hda_jack.c | 8 ++++---- sound/pci/hda/hda_proc.c | 2 +- sound/pci/hda/hda_sysfs.c | 2 +- sound/pci/hda/hda_trace.h | 4 ++-- sound/pci/hda/patch_ca0132.c | 2 +- sound/pci/hda/patch_hdmi.c | 6 +++--- sound/pci/hda/patch_via.c | 8 ++++---- 13 files changed, 41 insertions(+), 40 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index e98438e95e79..581b7fdef0e3 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -160,7 +160,7 @@ static int snd_hda_do_attach(struct hda_beep *beep) input_dev->name = "HDA Digital PCBeep"; input_dev->phys = beep->phys; input_dev->id.bustype = BUS_PCI; - input_dev->dev.parent = &codec->bus->card->card_dev; + input_dev->dev.parent = &codec->card->card_dev; input_dev->id.vendor = codec->vendor_id >> 16; input_dev->id.product = codec->vendor_id & 0xffff; @@ -224,7 +224,7 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) if (beep == NULL) return -ENOMEM; snprintf(beep->phys, sizeof(beep->phys), - "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr); + "card%d/codec#%d/beep0", codec->card->number, codec->addr); /* enable linear scale */ snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, 0x01); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index ce2dd7b0dc07..2d00417494e2 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -287,9 +287,9 @@ int snd_hda_codec_configure(struct hda_codec *codec) } /* audio codec should override the mixer name */ - if (codec->afg || !*codec->bus->card->mixername) - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), + if (codec->afg || !*codec->card->mixername) + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", codec->vendor_name, codec->chip_name); return 0; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 40300fcc5308..0533c862b539 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1196,9 +1196,8 @@ static void snd_hda_codec_dev_release(struct device *dev) * * Returns 0 if successful, or a negative error code. */ -int snd_hda_codec_new(struct hda_bus *bus, - unsigned int codec_addr, - struct hda_codec **codecp) +int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec **codecp) { struct hda_codec *codec; struct device *dev; @@ -1217,7 +1216,7 @@ int snd_hda_codec_new(struct hda_bus *bus, return -EINVAL; if (bus->caddr_tbl[codec_addr]) { - dev_err(bus->card->dev, + dev_err(card->dev, "address 0x%x is already occupied\n", codec_addr); return -EBUSY; @@ -1225,21 +1224,22 @@ int snd_hda_codec_new(struct hda_bus *bus, codec = kzalloc(sizeof(*codec), GFP_KERNEL); if (codec == NULL) { - dev_err(bus->card->dev, "can't allocate struct hda_codec\n"); + dev_err(card->dev, "can't allocate struct hda_codec\n"); return -ENOMEM; } dev = hda_codec_dev(codec); device_initialize(dev); - dev->parent = bus->card->dev; + dev->parent = card->dev; dev->bus = &snd_hda_bus_type; dev->release = snd_hda_codec_dev_release; dev->groups = snd_hda_dev_attr_groups; - dev_set_name(dev, "hdaudioC%dD%d", bus->card->number, codec_addr); + dev_set_name(dev, "hdaudioC%dD%d", card->number, codec_addr); dev_set_drvdata(dev, codec); /* for sysfs */ device_enable_async_suspend(dev); codec->bus = bus; + codec->card = card; codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); @@ -1300,7 +1300,7 @@ int snd_hda_codec_new(struct hda_bus *bus, setup_fg_nodes(codec); if (!codec->afg && !codec->mfg) { - dev_err(bus->card->dev, "no AFG or MFG node found\n"); + dev_err(card->dev, "no AFG or MFG node found\n"); err = -ENODEV; goto error; } @@ -1308,7 +1308,7 @@ int snd_hda_codec_new(struct hda_bus *bus, fg = codec->afg ? codec->afg : codec->mfg; err = read_widget_caps(codec, fg); if (err < 0) { - dev_err(bus->card->dev, "cannot malloc\n"); + dev_err(card->dev, "cannot malloc\n"); goto error; } err = read_pin_defaults(codec); @@ -1337,9 +1337,9 @@ int snd_hda_codec_new(struct hda_bus *bus, sprintf(component, "HDA:%08x,%08x,%08x", codec->vendor_id, codec->subsystem_id, codec->revision_id); - snd_component_add(codec->bus->card, component); + snd_component_add(card, component); - err = snd_device_new(bus->card, SNDRV_DEV_CODEC, codec, &dev_ops); + err = snd_device_new(card, SNDRV_DEV_CODEC, codec, &dev_ops); if (err < 0) goto error; @@ -2237,7 +2237,7 @@ find_mixer_ctl(struct hda_codec *codec, const char *name, int dev, int idx) if (snd_BUG_ON(strlen(name) >= sizeof(id.name))) return NULL; strcpy(id.name, name); - return snd_ctl_find_id(codec->bus->card, &id); + return snd_ctl_find_id(codec->card, &id); } /** @@ -2301,7 +2301,7 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, nid = kctl->id.subdevice & 0xffff; if (kctl->id.subdevice & (HDA_SUBDEV_NID_FLAG|HDA_SUBDEV_AMP_FLAG)) kctl->id.subdevice = 0; - err = snd_ctl_add(codec->bus->card, kctl); + err = snd_ctl_add(codec->card, kctl); if (err < 0) return err; item = snd_array_new(&codec->mixers); @@ -2354,7 +2354,7 @@ void snd_hda_ctls_clear(struct hda_codec *codec) int i; struct hda_nid_item *items = codec->mixers.list; for (i = 0; i < codec->mixers.used; i++) - snd_ctl_remove(codec->bus->card, items[i].kctl); + snd_ctl_remove(codec->card, items[i].kctl); snd_array_free(&codec->mixers); snd_array_free(&codec->nids); } @@ -2427,7 +2427,7 @@ EXPORT_SYMBOL_GPL(snd_hda_unlock_devices); int snd_hda_codec_reset(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; - struct snd_card *card = bus->card; + struct snd_card *card = codec->card; int i; if (snd_hda_lock_devices(bus) < 0) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 8cf70369cd10..8908a0768736 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -274,6 +274,7 @@ struct hda_pcm { struct hda_codec { struct device dev; struct hda_bus *bus; + struct snd_card *card; unsigned int addr; /* codec addr*/ struct list_head list; /* list point */ @@ -420,8 +421,8 @@ enum { * constructors */ int snd_hda_bus_new(struct snd_card *card, struct hda_bus **busp); -int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, - struct hda_codec **codecp); +int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, + unsigned int codec_addr, struct hda_codec **codecp); int snd_hda_codec_configure(struct hda_codec *codec); int snd_hda_codec_update_widgets(struct hda_codec *codec); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 1695f0e2bd9d..f50863a5159d 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1898,7 +1898,7 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots) for (c = 0; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; - err = snd_hda_codec_new(bus, c, &codec); + err = snd_hda_codec_new(bus, bus->card, c, &codec); if (err < 0) continue; codec->jackpoll_interval = get_jackpoll_interval(chip); diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 125f3420fa6a..57df06e76968 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -101,7 +101,7 @@ int snd_hda_create_hwdep(struct hda_codec *codec) int err; sprintf(hwname, "HDA Codec %d", codec->addr); - err = snd_hwdep_new(codec->bus->card, hwname, codec->addr, &hwdep); + err = snd_hwdep_new(codec->card, hwname, codec->addr, &hwdep); if (err < 0) return err; codec->hwdep = hwdep; diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index e664307617bd..d7cfe7b8c32b 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -135,7 +135,7 @@ void snd_hda_jack_tbl_clear(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_INPUT_JACK /* free jack instances manually when clearing/reconfiguring */ if (!codec->bus->shutdown && jack->jack) - snd_device_free(codec->bus->card, jack->jack); + snd_device_free(codec->card, jack->jack); #endif for (cb = jack->callback; cb; cb = next) { next = cb->next; @@ -340,7 +340,7 @@ void snd_hda_jack_report_sync(struct hda_codec *codec) if (!jack->kctl || jack->block_report) continue; state = get_jack_plug_state(jack->pin_sense); - snd_kctl_jack_report(codec->bus->card, jack->kctl, state); + snd_kctl_jack_report(codec->card, jack->kctl, state); #ifdef CONFIG_SND_HDA_INPUT_JACK if (jack->jack) snd_jack_report(jack->jack, @@ -412,11 +412,11 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, jack->phantom_jack = !!phantom_jack; state = snd_hda_jack_detect(codec, nid); - snd_kctl_jack_report(codec->bus->card, kctl, state); + snd_kctl_jack_report(codec->card, kctl, state); #ifdef CONFIG_SND_HDA_INPUT_JACK if (!phantom_jack) { jack->type = get_input_jack_type(codec, nid); - err = snd_jack_new(codec->bus->card, name, jack->type, + err = snd_jack_new(codec->card, name, jack->type, &jack->jack); if (err < 0) return err; diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index ce5a6da83419..cc32b878ae2e 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -839,7 +839,7 @@ int snd_hda_codec_proc_new(struct hda_codec *codec) int err; snprintf(name, sizeof(name), "codec#%d", codec->addr); - err = snd_card_proc_new(codec->bus->card, name, &entry); + err = snd_card_proc_new(codec->card, name, &entry); if (err < 0) return err; diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index ccc962a1699f..e13c75d67847 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -149,7 +149,7 @@ static int reconfig_codec(struct hda_codec *codec) err = snd_hda_codec_build_controls(codec); if (err < 0) goto error; - err = snd_card_register(codec->bus->card); + err = snd_card_register(codec->card); error: snd_hda_power_down(codec); return err; diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h index c0e1c7d24dbe..7fedfa862419 100644 --- a/sound/pci/hda/hda_trace.h +++ b/sound/pci/hda/hda_trace.h @@ -23,7 +23,7 @@ DECLARE_EVENT_CLASS(hda_cmd, ), TP_fast_assign( - __entry->card = (codec)->bus->card->number; + __entry->card = (codec)->card->number; __entry->addr = (codec)->addr; __entry->val = (val); ), @@ -71,7 +71,7 @@ DECLARE_EVENT_CLASS(hda_power, ), TP_fast_assign( - __entry->card = (codec)->bus->card->number; + __entry->card = (codec)->card->number; __entry->addr = (codec)->addr; ), diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 81991b4134cd..ced3e82d9e23 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4352,7 +4352,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec) const struct dsp_image_seg *dsp_os_image; const struct firmware *fw_entry; - if (request_firmware(&fw_entry, EFX_FILE, codec->bus->card->dev) != 0) + if (request_firmware(&fw_entry, EFX_FILE, codec->card->dev) != 0) return false; dsp_os_image = (struct dsp_image_seg *)(fw_entry->data); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f1812aabd63e..0f8354cbc7a7 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -579,7 +579,7 @@ static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) int err; snprintf(name, sizeof(name), "eld#%d.%d", codec->addr, index); - err = snd_card_proc_new(codec->bus->card, name, &entry); + err = snd_card_proc_new(codec->card, name, &entry); if (err < 0) return err; @@ -594,7 +594,7 @@ static int eld_proc_new(struct hdmi_spec_per_pin *per_pin, int index) static void eld_proc_free(struct hdmi_spec_per_pin *per_pin) { if (!per_pin->codec->bus->shutdown && per_pin->proc_entry) { - snd_device_free(per_pin->codec->bus->card, per_pin->proc_entry); + snd_device_free(per_pin->codec->card, per_pin->proc_entry); per_pin->proc_entry = NULL; } } @@ -1624,7 +1624,7 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) } if (eld_changed) - snd_ctl_notify(codec->bus->card, + snd_ctl_notify(codec->card, SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &per_pin->eld_ctl->id); unlock: diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 2045f33b1ace..57ad503ff940 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -907,16 +907,16 @@ static int patch_vt1708S(struct hda_codec *codec) if (get_codec_type(codec) == VT1708BCE) { kfree(codec->chip_name); codec->chip_name = kstrdup("VT1708BCE", GFP_KERNEL); - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", codec->vendor_name, codec->chip_name); } /* correct names for VT1705 */ if (codec->vendor_id == 0x11064397) { kfree(codec->chip_name); codec->chip_name = kstrdup("VT1705", GFP_KERNEL); - snprintf(codec->bus->card->mixername, - sizeof(codec->bus->card->mixername), + snprintf(codec->card->mixername, + sizeof(codec->card->mixername), "%s %s", codec->vendor_name, codec->chip_name); } From f4de8fe6cffb449a779dff61f071bd1af9e18e0f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 16:17:18 +0100 Subject: [PATCH 089/162] ALSA: hda - Remove superfluous memory allocation error messages The memory allocators should have already given the kernel warning messages, thus we don't have to annoy again. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 16 ++++------------ sound/pci/hda/hda_controller.c | 18 ++++-------------- sound/pci/hda/hda_intel.c | 5 +---- 3 files changed, 9 insertions(+), 30 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0533c862b539..262c41a8f96e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -776,10 +776,8 @@ int snd_hda_bus_new(struct snd_card *card, *busp = NULL; bus = kzalloc(sizeof(*bus), GFP_KERNEL); - if (bus == NULL) { - dev_err(card->dev, "can't allocate struct hda_bus\n"); + if (!bus) return -ENOMEM; - } bus->card = card; mutex_init(&bus->cmd_mutex); @@ -1223,10 +1221,8 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, } codec = kzalloc(sizeof(*codec), GFP_KERNEL); - if (codec == NULL) { - dev_err(card->dev, "can't allocate struct hda_codec\n"); + if (!codec) return -ENOMEM; - } dev = hda_codec_dev(codec); device_initialize(dev); @@ -1307,10 +1303,8 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, fg = codec->afg ? codec->afg : codec->mfg; err = read_widget_caps(codec, fg); - if (err < 0) { - dev_err(card->dev, "cannot malloc\n"); + if (err < 0) goto error; - } err = read_pin_defaults(codec); if (err < 0) goto error; @@ -1371,10 +1365,8 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) kfree(codec->wcaps); fg = codec->afg ? codec->afg : codec->mfg; err = read_widget_caps(codec, fg); - if (err < 0) { - codec_err(codec, "cannot malloc\n"); + if (err < 0) return err; - } snd_array_free(&codec->init_pins); err = read_pin_defaults(codec); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index f50863a5159d..be02bca6f7e6 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -974,14 +974,9 @@ static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, */ static int azx_alloc_cmd_io(struct azx *chip) { - int err; - /* single page (at least 4096 bytes) must suffice for both ringbuffes */ - err = chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, - PAGE_SIZE, &chip->rb); - if (err < 0) - dev_err(chip->card->dev, "cannot allocate CORB/RIRB\n"); - return err; + return chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, + PAGE_SIZE, &chip->rb); } EXPORT_SYMBOL_GPL(azx_alloc_cmd_io); @@ -1472,7 +1467,6 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, int azx_alloc_stream_pages(struct azx *chip) { int i, err; - struct snd_card *card = chip->card; for (i = 0; i < chip->num_streams; i++) { dsp_lock_init(&chip->azx_dev[i]); @@ -1480,18 +1474,14 @@ int azx_alloc_stream_pages(struct azx *chip) err = chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, BDL_SIZE, &chip->azx_dev[i].bdl); - if (err < 0) { - dev_err(card->dev, "cannot allocate BDL\n"); + if (err < 0) return -ENOMEM; - } } /* allocate memory for the position buffer */ err = chip->ops->dma_alloc_pages(chip, SNDRV_DMA_TYPE_DEV, chip->num_streams * 8, &chip->posbuf); - if (err < 0) { - dev_err(card->dev, "cannot allocate posbuf\n"); + if (err < 0) return -ENOMEM; - } /* allocate CORB/RIRB */ err = azx_alloc_cmd_io(chip); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e75e8137e296..f7fb1b51d446 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1383,7 +1383,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, hda = kzalloc(sizeof(*hda), GFP_KERNEL); if (!hda) { - dev_err(card->dev, "Cannot allocate hda\n"); pci_disable_device(pci); return -ENOMEM; } @@ -1564,10 +1563,8 @@ static int azx_first_init(struct azx *chip) chip->num_streams = chip->playback_streams + chip->capture_streams; chip->azx_dev = kcalloc(chip->num_streams, sizeof(*chip->azx_dev), GFP_KERNEL); - if (!chip->azx_dev) { - dev_err(card->dev, "cannot malloc azx_dev\n"); + if (!chip->azx_dev) return -ENOMEM; - } err = azx_alloc_stream_pages(chip); if (err < 0) From bbbc7e8502c95237dbd86cc4d0a12ca9a6b18c8f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 17:43:19 +0100 Subject: [PATCH 090/162] ALSA: hda - Allocate hda_pcm objects dynamically So far, the hda_codec object kept the hda_pcm list in an array, and the codec driver was expected to assign the array. However, this makes the object life cycle management harder, because the assigned array is freed at the codec driver detach while it might be still accessed by the opened streams. In this patch, we allocate each hda_pcm object dynamically and manage it as a linked list. Each object has a kref refcount, and both the codec driver binder and the PCM open/close touches it, so that the object won't be freed while in use. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 116 ++++++++++++++++++++++------------ sound/pci/hda/hda_codec.h | 18 +++++- sound/pci/hda/hda_generic.c | 28 ++++---- sound/pci/hda/hda_generic.h | 2 +- sound/pci/hda/hda_proc.c | 6 +- sound/pci/hda/patch_ca0132.c | 29 ++++----- sound/pci/hda/patch_hdmi.c | 30 +++------ sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_si3054.c | 11 ++-- sound/pci/hda/patch_via.c | 6 +- 10 files changed, 145 insertions(+), 103 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 262c41a8f96e..20283bead10a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1116,6 +1116,60 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) return p; } +/* + * PCM device + */ +static void release_pcm(struct kref *kref) +{ + struct hda_pcm *pcm = container_of(kref, struct hda_pcm, kref); + + if (pcm->pcm) + snd_device_free(pcm->codec->card, pcm->pcm); + clear_bit(pcm->device, pcm->codec->bus->pcm_dev_bits); + kfree(pcm->name); + kfree(pcm); +} + +void snd_hda_codec_pcm_put(struct hda_pcm *pcm) +{ + kref_put(&pcm->kref, release_pcm); +} +EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_put); + +struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, + const char *fmt, ...) +{ + struct hda_pcm *pcm; + va_list args; + + va_start(args, fmt); + pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return NULL; + + pcm->codec = codec; + kref_init(&pcm->kref); + pcm->name = kvasprintf(GFP_KERNEL, fmt, args); + if (!pcm->name) { + kfree(pcm); + return NULL; + } + + list_add_tail(&pcm->list, &codec->pcm_list_head); + return pcm; +} +EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_new); + +static void codec_release_pcms(struct hda_codec *codec) +{ + struct hda_pcm *pcm, *n; + + list_for_each_entry_safe(pcm, n, &codec->pcm_list_head, list) { + list_del_init(&pcm->list); + snd_hda_codec_pcm_put(pcm); + } +} + /* * codec destructor */ @@ -1124,6 +1178,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) if (!codec) return; cancel_delayed_work_sync(&codec->jackpoll_work); + codec_release_pcms(codec); if (device_is_registered(hda_codec_dev(codec))) device_del(hda_codec_dev(codec)); snd_hda_jack_tbl_clear(codec); @@ -1251,6 +1306,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); INIT_LIST_HEAD(&codec->conn_list); + INIT_LIST_HEAD(&codec->pcm_list_head); INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; @@ -2370,9 +2426,8 @@ int snd_hda_lock_devices(struct hda_bus *bus) goto err_clear; list_for_each_entry(codec, &bus->codec_list, list) { - int pcm; - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + struct hda_pcm *cpcm; + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { if (!cpcm->pcm) continue; if (cpcm->pcm->streams[0].substream_opened || @@ -2419,8 +2474,6 @@ EXPORT_SYMBOL_GPL(snd_hda_unlock_devices); int snd_hda_codec_reset(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; - struct snd_card *card = codec->card; - int i; if (snd_hda_lock_devices(bus) < 0) return -EBUSY; @@ -2429,14 +2482,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) cancel_delayed_work_sync(&codec->jackpoll_work); flush_workqueue(bus->workq); snd_hda_ctls_clear(codec); - /* release PCMs */ - for (i = 0; i < codec->num_pcms; i++) { - if (codec->pcm_info[i].pcm) { - snd_device_free(card, codec->pcm_info[i].pcm); - clear_bit(codec->pcm_info[i].device, - bus->pcm_dev_bits); - } - } + codec_release_pcms(codec); snd_hda_detach_beep_device(codec); if (device_is_registered(hda_codec_dev(codec))) device_del(hda_codec_dev(codec)); @@ -2454,8 +2500,6 @@ int snd_hda_codec_reset(struct hda_codec *codec) snd_array_free(&codec->cvt_setups); snd_array_free(&codec->spdif_out); snd_array_free(&codec->verbs); - codec->num_pcms = 0; - codec->pcm_info = NULL; codec->preset = NULL; codec->slave_dig_outs = NULL; codec->spdif_status_reset = 0; @@ -3952,12 +3996,12 @@ static void hda_call_codec_resume(struct hda_codec *codec) static int hda_codec_runtime_suspend(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); + struct hda_pcm *pcm; unsigned int state; - int i; cancel_delayed_work_sync(&codec->jackpoll_work); - for (i = 0; i < codec->num_pcms; i++) - snd_pcm_suspend_all(codec->pcm_info[i].pcm); + list_for_each_entry(pcm, &codec->pcm_list_head, list) + snd_pcm_suspend_all(pcm->pcm); state = hda_call_codec_suspend(codec); if (codec->d3_stop_clk && codec->epss && (state & AC_PWRST_CLK_STOP_OK)) clear_bit(codec->addr, &codec->bus->codec_powered); @@ -4018,22 +4062,21 @@ EXPORT_SYMBOL_GPL(snd_hda_build_controls); */ static int add_std_chmaps(struct hda_codec *codec) { - int i, str, err; + struct hda_pcm *pcm; + int str, err; - for (i = 0; i < codec->num_pcms; i++) { + list_for_each_entry(pcm, &codec->pcm_list_head, list) { for (str = 0; str < 2; str++) { - struct snd_pcm *pcm = codec->pcm_info[i].pcm; - struct hda_pcm_stream *hinfo = - &codec->pcm_info[i].stream[str]; + struct hda_pcm_stream *hinfo = &pcm->stream[str]; struct snd_pcm_chmap *chmap; const struct snd_pcm_chmap_elem *elem; - if (codec->pcm_info[i].own_chmap) + if (pcm->own_chmap) continue; if (!pcm || !hinfo->substreams) continue; elem = hinfo->chmap ? hinfo->chmap : snd_pcm_std_chmaps; - err = snd_pcm_add_chmap_ctls(pcm, str, elem, + err = snd_pcm_add_chmap_ctls(pcm->pcm, str, elem, hinfo->channels_max, 0, &chmap); if (err < 0) @@ -4564,10 +4607,10 @@ static int get_empty_pcm_device(struct hda_bus *bus, unsigned int type) /* call build_pcms ops of the given codec and set up the default parameters */ int snd_hda_codec_parse_pcms(struct hda_codec *codec) { - unsigned int pcm; + struct hda_pcm *cpcm; int err; - if (codec->num_pcms) + if (!list_empty(&codec->pcm_list_head)) return 0; /* already parsed */ if (!codec->patch_ops.build_pcms) @@ -4580,8 +4623,7 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) return err; } - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { int stream; for (stream = 0; stream < 2; stream++) { @@ -4589,8 +4631,6 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) if (!info->substreams) continue; - if (snd_BUG_ON(!cpcm->name)) - return -EINVAL; err = set_pcm_default_values(codec, info); if (err < 0) { codec_warn(codec, @@ -4608,7 +4648,7 @@ int snd_hda_codec_parse_pcms(struct hda_codec *codec) int snd_hda_codec_build_pcms(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; - unsigned int pcm; + struct hda_pcm *cpcm; int dev, err; if (snd_BUG_ON(!bus->ops.attach_pcm)) @@ -4621,9 +4661,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) } /* attach a new PCM streams */ - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - struct hda_pcm *cpcm = &codec->pcm_info[pcm]; - + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { if (cpcm->pcm) continue; /* already attached */ if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) @@ -4651,11 +4689,9 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) * * Create PCM information for each codec included in the bus. * - * The build_pcms codec patch is requested to set up codec->num_pcms and - * codec->pcm_info properly. The array is referred by the top-level driver - * to create its PCM instances. - * The allocated codec->pcm_info should be released in codec->patch_ops.free - * callback. + * The build_pcms codec patch is requested to create and assign new + * hda_pcm objects. The codec is responsible to call snd_hda_codec_pcm_new() + * and fills the fields. Later they are instantiated by this function. * * At least, substreams, channels_min and channels_max must be filled for * each stream. substreams = 0 indicates that the stream doesn't exist. diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 8908a0768736..2ccd6f9a91fe 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -21,6 +21,7 @@ #ifndef __SOUND_HDA_CODEC_H #define __SOUND_HDA_CODEC_H +#include #include #include #include @@ -268,6 +269,10 @@ struct hda_pcm { int device; /* device number to assign */ struct snd_pcm *pcm; /* assigned PCM instance */ bool own_chmap; /* codec driver provides own channel maps */ + /* private: */ + struct hda_codec *codec; + struct kref kref; + struct list_head list; }; /* codec information */ @@ -301,8 +306,7 @@ struct hda_codec { struct hda_codec_ops patch_ops; /* PCM to create, set by patch_ops.build_pcms callback */ - unsigned int num_pcms; - struct hda_pcm *pcm_info; + struct list_head pcm_list_head; /* codec specific info */ void *spec; @@ -521,6 +525,16 @@ int snd_hda_build_pcms(struct hda_bus *bus); int snd_hda_codec_parse_pcms(struct hda_codec *codec); int snd_hda_codec_build_pcms(struct hda_codec *codec); +__printf(2, 3) +struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, + const char *fmt, ...); + +static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm) +{ + kref_get(&pcm->kref); +} +void snd_hda_codec_pcm_put(struct hda_pcm *pcm); + int snd_hda_codec_prepare(struct hda_codec *codec, struct hda_pcm_stream *hinfo, unsigned int stream, diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 947d1a50f384..092f06fd21bd 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4644,7 +4644,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec) err = snd_hda_create_dig_out_ctls(codec, spec->multiout.dig_out_nid, spec->multiout.dig_out_nid, - spec->pcm_rec[1].pcm_type); + spec->pcm_rec[1]->pcm_type); if (err < 0) return err; if (!spec->no_analog) { @@ -5115,20 +5115,20 @@ static void fill_pcm_stream_name(char *str, size_t len, const char *sfx, int snd_hda_gen_build_pcms(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; + struct hda_pcm *info; const struct hda_pcm_stream *p; bool have_multi_adcs; - codec->num_pcms = 1; - codec->pcm_info = info; - if (spec->no_analog) goto skip_analog; fill_pcm_stream_name(spec->stream_name_analog, sizeof(spec->stream_name_analog), " Analog", codec->chip_name); - info->name = spec->stream_name_analog; + info = snd_hda_codec_pcm_new(codec, "%s", spec->stream_name_analog); + if (!info) + return -ENOMEM; + spec->pcm_rec[0] = info; if (spec->multiout.num_dacs > 0) { p = spec->stream_analog_playback; @@ -5161,10 +5161,12 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) fill_pcm_stream_name(spec->stream_name_digital, sizeof(spec->stream_name_digital), " Digital", codec->chip_name); - codec->num_pcms = 2; + info = snd_hda_codec_pcm_new(codec, "%s", + spec->stream_name_digital); + if (!info) + return -ENOMEM; codec->slave_dig_outs = spec->multiout.slave_dig_outs; - info = spec->pcm_rec + 1; - info->name = spec->stream_name_digital; + spec->pcm_rec[1] = info; if (spec->dig_out_type) info->pcm_type = spec->dig_out_type; else @@ -5198,9 +5200,11 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) fill_pcm_stream_name(spec->stream_name_alt_analog, sizeof(spec->stream_name_alt_analog), " Alt Analog", codec->chip_name); - codec->num_pcms = 3; - info = spec->pcm_rec + 2; - info->name = spec->stream_name_alt_analog; + info = snd_hda_codec_pcm_new(codec, "%s", + spec->stream_name_alt_analog); + if (!info) + return -ENOMEM; + spec->pcm_rec[2] = info; if (spec->alt_dac_nid) { p = spec->stream_analog_alt_playback; if (!p) diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 3d852660443a..b211f889b335 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -144,7 +144,7 @@ struct hda_gen_spec { int const_channel_count; /* channel count for all */ /* PCM information */ - struct hda_pcm pcm_rec[3]; /* used in build_pcms() */ + struct hda_pcm *pcm_rec[3]; /* used in build_pcms() */ /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index cc32b878ae2e..aeb983e7506d 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -99,10 +99,10 @@ static void print_nid_array(struct snd_info_buffer *buffer, static void print_nid_pcms(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { - int pcm, type; + int type; struct hda_pcm *cpcm; - for (pcm = 0; pcm < codec->num_pcms; pcm++) { - cpcm = &codec->pcm_info[pcm]; + + list_for_each_entry(cpcm, &codec->pcm_list_head, list) { for (type = 0; type < 2; type++) { if (cpcm->stream[type].nid != nid || cpcm->pcm == NULL) continue; diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index ced3e82d9e23..555781fad26f 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -719,7 +719,6 @@ struct ca0132_spec { unsigned int num_inputs; hda_nid_t shared_mic_nid; hda_nid_t shared_out_nid; - struct hda_pcm pcm_rec[5]; /* PCM information */ /* chip access */ struct mutex chipio_mutex; /* chip access mutex */ @@ -4036,12 +4035,11 @@ static struct hda_pcm_stream ca0132_pcm_digital_capture = { static int ca0132_build_pcms(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; + struct hda_pcm *info; - codec->pcm_info = info; - codec->num_pcms = 0; - - info->name = "CA0132 Analog"; + info = snd_hda_codec_pcm_new(codec, "CA0132 Analog"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0]; info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = @@ -4049,27 +4047,27 @@ static int ca0132_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0]; - codec->num_pcms++; - info++; - info->name = "CA0132 Analog Mic-In2"; + info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1]; - codec->num_pcms++; - info++; - info->name = "CA0132 What U Hear"; + info = snd_hda_codec_pcm_new(codec, "CA0132 What U Hear"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[2]; - codec->num_pcms++; if (!spec->dig_out && !spec->dig_in) return 0; - info++; - info->name = "CA0132 Digital"; + info = snd_hda_codec_pcm_new(codec, "CA0132 Digital"); + if (!info) + return -ENOMEM; info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->dig_out) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = @@ -4081,7 +4079,6 @@ static int ca0132_build_pcms(struct hda_codec *codec) ca0132_pcm_digital_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in; } - codec->num_pcms++; return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0f8354cbc7a7..708bbed15ea3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -86,7 +86,6 @@ struct hdmi_spec_per_pin { bool non_pcm; bool chmap_set; /* channel-map override by ALSA API? */ unsigned char chmap[8]; /* ALSA API channel-map */ - char pcm_name[8]; /* filled in build_pcm callbacks */ #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif @@ -132,7 +131,7 @@ struct hdmi_spec { int num_pins; struct snd_array pins; /* struct hdmi_spec_per_pin */ - struct snd_array pcm_rec; /* struct hda_pcm */ + struct hda_pcm *pcm_rec[16]; unsigned int channels_max; /* max over all cvts */ struct hdmi_eld temp_eld; @@ -355,8 +354,7 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { ((struct hdmi_spec_per_pin *)snd_array_elem(&spec->pins, idx)) #define get_cvt(spec, idx) \ ((struct hdmi_spec_per_cvt *)snd_array_elem(&spec->cvts, idx)) -#define get_pcm_rec(spec, idx) \ - ((struct hda_pcm *)snd_array_elem(&spec->pcm_rec, idx)) +#define get_pcm_rec(spec, idx) ((spec)->pcm_rec[idx]) static int pin_nid_to_pin_index(struct hda_codec *codec, hda_nid_t pin_nid) { @@ -2056,11 +2054,10 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) struct hdmi_spec_per_pin *per_pin; per_pin = get_pin(spec, pin_idx); - sprintf(per_pin->pcm_name, "HDMI %d", pin_idx); - info = snd_array_new(&spec->pcm_rec); + info = snd_hda_codec_pcm_new(codec, "HDMI %d", pin_idx); if (!info) return -ENOMEM; - info->name = per_pin->pcm_name; + spec->pcm_rec[pin_idx] = info; info->pcm_type = HDA_PCM_TYPE_HDMI; info->own_chmap = true; @@ -2070,9 +2067,6 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) /* other pstr fields are set in open */ } - codec->num_pcms = spec->num_pins; - codec->pcm_info = spec->pcm_rec.list; - return 0; } @@ -2125,13 +2119,15 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) /* add channel maps */ for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hda_pcm *pcm; struct snd_pcm_chmap *chmap; struct snd_kcontrol *kctl; int i; - if (!codec->pcm_info[pin_idx].pcm) + pcm = spec->pcm_rec[pin_idx]; + if (!pcm || !pcm->pcm) break; - err = snd_pcm_add_chmap_ctls(codec->pcm_info[pin_idx].pcm, + err = snd_pcm_add_chmap_ctls(pcm->pcm, SNDRV_PCM_STREAM_PLAYBACK, NULL, 0, pin_idx, &chmap); if (err < 0) @@ -2186,14 +2182,12 @@ static void hdmi_array_init(struct hdmi_spec *spec, int nums) { snd_array_init(&spec->pins, sizeof(struct hdmi_spec_per_pin), nums); snd_array_init(&spec->cvts, sizeof(struct hdmi_spec_per_cvt), nums); - snd_array_init(&spec->pcm_rec, sizeof(struct hda_pcm), nums); } static void hdmi_array_free(struct hdmi_spec *spec) { snd_array_free(&spec->pins); snd_array_free(&spec->cvts); - snd_array_free(&spec->pcm_rec); } static void generic_hdmi_free(struct hda_codec *codec) @@ -2381,11 +2375,10 @@ static int simple_playback_build_pcms(struct hda_codec *codec) chans = get_wcaps(codec, per_cvt->cvt_nid); chans = get_wcaps_channels(chans); - info = snd_array_new(&spec->pcm_rec); + info = snd_hda_codec_pcm_new(codec, "HDMI 0"); if (!info) return -ENOMEM; - info->name = get_pin(spec, 0)->pcm_name; - sprintf(info->name, "HDMI 0"); + spec->pcm_rec[0] = info; info->pcm_type = HDA_PCM_TYPE_HDMI; pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK]; *pstr = spec->pcm_playback; @@ -2393,9 +2386,6 @@ static int simple_playback_build_pcms(struct hda_codec *codec) if (pstr->channels_max <= 2 && chans && chans <= 16) pstr->channels_max = chans; - codec->num_pcms = 1; - codec->pcm_info = info; - return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 70808f92276a..0a5a2246e57a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5843,7 +5843,7 @@ static void alc_fixup_bass_chmap(struct hda_codec *codec, { if (action == HDA_FIXUP_ACT_BUILD) { struct alc_spec *spec = codec->spec; - spec->gen.pcm_rec[0].stream[0].chmap = asus_pcm_2_1_chmaps; + spec->gen.pcm_rec[0]->stream[0].chmap = asus_pcm_2_1_chmaps; } } diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 38a477333321..df243134baa8 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -83,7 +83,6 @@ struct si3054_spec { unsigned international; - struct hda_pcm pcm; }; @@ -199,11 +198,11 @@ static const struct hda_pcm_stream si3054_pcm = { static int si3054_build_pcms(struct hda_codec *codec) { - struct si3054_spec *spec = codec->spec; - struct hda_pcm *info = &spec->pcm; - codec->num_pcms = 1; - codec->pcm_info = info; - info->name = "Si3054 Modem"; + struct hda_pcm *info; + + info = snd_hda_codec_pcm_new(codec, "Si3054 Modem"); + if (!info) + return -ENOMEM; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = si3054_pcm; info->stream[SNDRV_PCM_STREAM_CAPTURE] = si3054_pcm; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = codec->mfg; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 57ad503ff940..11a05638e03b 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -683,8 +683,10 @@ static int vt1708_build_pcms(struct hda_codec *codec) * 24bit samples are used. Until any workaround is found, * disable the 24bit format, so far. */ - for (i = 0; i < codec->num_pcms; i++) { - struct hda_pcm *info = &spec->gen.pcm_rec[i]; + for (i = 0; i < ARRAY_SIZE(spec->gen.pcm_rec); i++) { + struct hda_pcm *info = spec->gen.pcm_rec[i]; + if (!info) + continue; if (!info->stream[SNDRV_PCM_STREAM_PLAYBACK].substreams || info->pcm_type != HDA_PCM_TYPE_AUDIO) continue; From 61ca4107a16828559e2463e49b87002990da0b98 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 17:57:55 +0100 Subject: [PATCH 091/162] ALSA: hda - Don't assume non-NULL PCM ops The PCM ops might be set NULL, or cleared to NULL when the driver is unbound. Give a proper NULL check at each place to be more robust. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 9 +++++++-- sound/pci/hda/hda_controller.c | 30 +++++++++++++++++++----------- 2 files changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 20283bead10a..3bd9158addc2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4525,7 +4525,11 @@ int snd_hda_codec_prepare(struct hda_codec *codec, { int ret; mutex_lock(&codec->bus->prepare_mutex); - ret = hinfo->ops.prepare(hinfo, codec, stream, format, substream); + if (hinfo->ops.prepare) + ret = hinfo->ops.prepare(hinfo, codec, stream, format, + substream); + else + ret = -ENODEV; if (ret >= 0) purify_inactive_streams(codec); mutex_unlock(&codec->bus->prepare_mutex); @@ -4546,7 +4550,8 @@ void snd_hda_codec_cleanup(struct hda_codec *codec, struct snd_pcm_substream *substream) { mutex_lock(&codec->bus->prepare_mutex); - hinfo->ops.cleanup(hinfo, codec, substream); + if (hinfo->ops.cleanup) + hinfo->ops.cleanup(hinfo, codec, substream); mutex_unlock(&codec->bus->prepare_mutex); } EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index be02bca6f7e6..ad85f9bfaf57 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -416,7 +416,8 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) azx_dev->running = 0; spin_unlock_irqrestore(&chip->reg_lock, flags); azx_release_device(azx_dev); - hinfo->ops.close(hinfo, apcm->codec, substream); + if (hinfo->ops.close) + hinfo->ops.close(hinfo, apcm->codec, substream); snd_hda_power_down(apcm->codec); mutex_unlock(&chip->open_mutex); return 0; @@ -808,8 +809,8 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) mutex_lock(&chip->open_mutex); azx_dev = azx_assign_device(chip, substream); if (azx_dev == NULL) { - mutex_unlock(&chip->open_mutex); - return -EBUSY; + err = -EBUSY; + goto unlock; } runtime->hw = azx_pcm_hw; runtime->hw.channels_min = hinfo->channels_min; @@ -844,12 +845,13 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); snd_hda_power_up(apcm->codec); - err = hinfo->ops.open(hinfo, apcm->codec, substream); + if (hinfo->ops.open) + err = hinfo->ops.open(hinfo, apcm->codec, substream); + else + err = -ENODEV; if (err < 0) { azx_release_device(azx_dev); - snd_hda_power_down(apcm->codec); - mutex_unlock(&chip->open_mutex); - return err; + goto powerdown; } snd_pcm_limit_hw_rates(runtime); /* sanity check */ @@ -858,10 +860,10 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_BUG_ON(!runtime->hw.formats) || snd_BUG_ON(!runtime->hw.rates)) { azx_release_device(azx_dev); - hinfo->ops.close(hinfo, apcm->codec, substream); - snd_hda_power_down(apcm->codec); - mutex_unlock(&chip->open_mutex); - return -EINVAL; + if (hinfo->ops.close) + hinfo->ops.close(hinfo, apcm->codec, substream); + err = -EINVAL; + goto powerdown; } /* disable LINK_ATIME timestamps for capture streams @@ -880,6 +882,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); return 0; + + powerdown: + snd_hda_power_down(apcm->codec); + unlock: + mutex_unlock(&chip->open_mutex); + return err; } static int azx_pcm_mmap(struct snd_pcm_substream *substream, From e086e3035e0691b362755d1b5e24df631eee335a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 18:01:22 +0100 Subject: [PATCH 092/162] ALSA: core: Re-add snd_device_disconnect() Revive snd_device_disconnect() again so that it can be called from the individual driver. This time, HD-audio will need it. Signed-off-by: Takashi Iwai --- include/sound/core.h | 3 ++- sound/core/device.c | 43 +++++++++++++++++++++++++++++++++---------- sound/core/init.c | 5 +---- 3 files changed, 36 insertions(+), 15 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index da5748289968..b12931f513f4 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -278,7 +278,8 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, void *device_data, struct snd_device_ops *ops); int snd_device_register(struct snd_card *card, void *device_data); int snd_device_register_all(struct snd_card *card); -int snd_device_disconnect_all(struct snd_card *card); +void snd_device_disconnect(struct snd_card *card, void *device_data); +void snd_device_disconnect_all(struct snd_card *card); void snd_device_free(struct snd_card *card, void *device_data); void snd_device_free_all(struct snd_card *card); diff --git a/sound/core/device.c b/sound/core/device.c index 41bec3075ae5..446dc452654e 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -73,7 +73,7 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, } EXPORT_SYMBOL(snd_device_new); -static int __snd_device_disconnect(struct snd_device *dev) +static void __snd_device_disconnect(struct snd_device *dev) { if (dev->state == SNDRV_DEV_REGISTERED) { if (dev->ops->dev_disconnect && @@ -81,7 +81,6 @@ static int __snd_device_disconnect(struct snd_device *dev) dev_err(dev->card->dev, "device disconnect failure\n"); dev->state = SNDRV_DEV_DISCONNECTED; } - return 0; } static void __snd_device_free(struct snd_device *dev) @@ -108,6 +107,34 @@ static struct snd_device *look_for_dev(struct snd_card *card, void *device_data) return NULL; } +/** + * snd_device_disconnect - disconnect the device + * @card: the card instance + * @device_data: the data pointer to disconnect + * + * Turns the device into the disconnection state, invoking + * dev_disconnect callback, if the device was already registered. + * + * Usually called from snd_card_disconnect(). + * + * Return: Zero if successful, or a negative error code on failure or if the + * device not found. + */ +void snd_device_disconnect(struct snd_card *card, void *device_data) +{ + struct snd_device *dev; + + if (snd_BUG_ON(!card || !device_data)) + return; + dev = look_for_dev(card, device_data); + if (dev) + __snd_device_disconnect(dev); + else + dev_dbg(card->dev, "device disconnect %p (from %pF), not found\n", + device_data, __builtin_return_address(0)); +} +EXPORT_SYMBOL_GPL(snd_device_disconnect); + /** * snd_device_free - release the device from the card * @card: the card instance @@ -195,18 +222,14 @@ int snd_device_register_all(struct snd_card *card) * disconnect all the devices on the card. * called from init.c */ -int snd_device_disconnect_all(struct snd_card *card) +void snd_device_disconnect_all(struct snd_card *card) { struct snd_device *dev; - int err = 0; if (snd_BUG_ON(!card)) - return -ENXIO; - list_for_each_entry_reverse(dev, &card->devices, list) { - if (__snd_device_disconnect(dev) < 0) - err = -ENXIO; - } - return err; + return; + list_for_each_entry_reverse(dev, &card->devices, list) + __snd_device_disconnect(dev); } /* diff --git a/sound/core/init.c b/sound/core/init.c index 35419054821c..04734e047bfe 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -400,7 +400,6 @@ static const struct file_operations snd_shutdown_f_ops = int snd_card_disconnect(struct snd_card *card) { struct snd_monitor_file *mfile; - int err; if (!card) return -EINVAL; @@ -445,9 +444,7 @@ int snd_card_disconnect(struct snd_card *card) #endif /* notify all devices that we are disconnected */ - err = snd_device_disconnect_all(card); - if (err < 0) - dev_err(card->dev, "not all devices for card %i can be disconnected\n", card->number); + snd_device_disconnect_all(card); snd_info_card_disconnect(card); if (card->registered) { From 9a6246ff78ac33af78f82704cde6fec361597eea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 18:17:28 +0100 Subject: [PATCH 093/162] ALSA: hda - Implement unbind more safely Now we have all pieces ready, and put them into places: - add the hda_pcm refcount to azx_pcm_open() and azx_pcm_close(), - call the most of cleanup code in hda_codec_reset() from the codec driver remove, - call the same code also from the hda_codec object free. Then the codec driver can be unbound more safely now. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 3 +- sound/pci/hda/hda_codec.c | 70 ++++++++++++++++++---------------- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_controller.c | 3 ++ sound/pci/hda/hda_local.h | 1 + 5 files changed, 43 insertions(+), 35 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 2d00417494e2..311896a23cd1 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -125,8 +125,7 @@ static int hda_codec_driver_remove(struct device *dev) if (codec->patch_ops.free) codec->patch_ops.free(codec); - codec->preset = NULL; - memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + snd_hda_codec_cleanup_for_unbind(codec); module_put(dev->driver->owner); return 0; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3bd9158addc2..2c7e481a9171 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1160,36 +1160,62 @@ struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_new); +/* + * codec destructor + */ static void codec_release_pcms(struct hda_codec *codec) { struct hda_pcm *pcm, *n; list_for_each_entry_safe(pcm, n, &codec->pcm_list_head, list) { list_del_init(&pcm->list); + if (pcm->pcm) + snd_device_disconnect(codec->card, pcm->pcm); snd_hda_codec_pcm_put(pcm); } } -/* - * codec destructor - */ +void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) +{ + cancel_delayed_work_sync(&codec->jackpoll_work); + flush_workqueue(codec->bus->workq); + if (!codec->in_freeing) + snd_hda_ctls_clear(codec); + codec_release_pcms(codec); + snd_hda_detach_beep_device(codec); + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + snd_hda_jack_tbl_clear(codec); + codec->proc_widget_hook = NULL; + codec->spec = NULL; + + free_hda_cache(&codec->amp_cache); + free_hda_cache(&codec->cmd_cache); + init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); + init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); + + /* free only driver_pins so that init_pins + user_pins are restored */ + snd_array_free(&codec->driver_pins); + snd_array_free(&codec->cvt_setups); + snd_array_free(&codec->spdif_out); + snd_array_free(&codec->verbs); + codec->preset = NULL; + codec->slave_dig_outs = NULL; + codec->spdif_status_reset = 0; + snd_array_free(&codec->mixers); + snd_array_free(&codec->nids); + remove_conn_list(codec); +} + static void snd_hda_codec_free(struct hda_codec *codec) { if (!codec) return; - cancel_delayed_work_sync(&codec->jackpoll_work); - codec_release_pcms(codec); + codec->in_freeing = 1; if (device_is_registered(hda_codec_dev(codec))) device_del(hda_codec_dev(codec)); - snd_hda_jack_tbl_clear(codec); free_init_pincfgs(codec); flush_workqueue(codec->bus->workq); list_del(&codec->list); - snd_array_free(&codec->mixers); - snd_array_free(&codec->nids); - snd_array_free(&codec->cvt_setups); - snd_array_free(&codec->spdif_out); - remove_conn_list(codec); codec->bus->caddr_tbl[codec->addr] = NULL; clear_bit(codec->addr, &codec->bus->codec_powered); snd_hda_sysfs_clear(codec); @@ -2479,31 +2505,9 @@ int snd_hda_codec_reset(struct hda_codec *codec) return -EBUSY; /* OK, let it free */ - cancel_delayed_work_sync(&codec->jackpoll_work); - flush_workqueue(bus->workq); - snd_hda_ctls_clear(codec); - codec_release_pcms(codec); - snd_hda_detach_beep_device(codec); if (device_is_registered(hda_codec_dev(codec))) device_del(hda_codec_dev(codec)); - memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); - snd_hda_jack_tbl_clear(codec); - codec->proc_widget_hook = NULL; - codec->spec = NULL; - free_hda_cache(&codec->amp_cache); - free_hda_cache(&codec->cmd_cache); - init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); - init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); - /* free only driver_pins so that init_pins + user_pins are restored */ - snd_array_free(&codec->driver_pins); - snd_array_free(&codec->cvt_setups); - snd_array_free(&codec->spdif_out); - snd_array_free(&codec->verbs); - codec->preset = NULL; - codec->slave_dig_outs = NULL; - codec->spdif_status_reset = 0; - /* allow device access again */ snd_hda_unlock_devices(bus); return 0; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 2ccd6f9a91fe..fc62ca51fd35 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -350,6 +350,7 @@ struct hda_codec { #endif /* misc flags */ + unsigned int in_freeing:1; /* being released */ unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each * status change * (e.g. Realtek codecs) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ad85f9bfaf57..cae50d5ffb81 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -420,6 +420,7 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) hinfo->ops.close(hinfo, apcm->codec, substream); snd_hda_power_down(apcm->codec); mutex_unlock(&chip->open_mutex); + snd_hda_codec_pcm_put(apcm->info); return 0; } @@ -806,6 +807,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) int err; int buff_step; + snd_hda_codec_pcm_get(apcm->info); mutex_lock(&chip->open_mutex); azx_dev = azx_assign_device(chip, substream); if (azx_dev == NULL) { @@ -887,6 +889,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_hda_power_down(apcm->codec); unlock: mutex_unlock(&chip->open_mutex); + snd_hda_codec_pcm_put(apcm->info); return err; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 8588813163e3..1d001647fc47 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -150,6 +150,7 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, #define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \ __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL) int snd_hda_codec_reset(struct hda_codec *codec); +void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec); enum { HDA_VMUTE_OFF, From bcd96557bd0ab1129fcdde073d5700aed8fcb942 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 20:44:54 +0100 Subject: [PATCH 094/162] ALSA: hda - Build PCMs and controls at codec driver probe This makes the code flow easier -- instead of the controller driver calling snd_hda_build_pcms() and snd_hda_build_controls() explicitly, the codec driver itself builds PCMs and controls at probe time. Then the controller driver only needs to call snd_card_register(). Also, this allows us the full bind/unbind control, too. Even when a codec driver is bound later, it automatically registers the new PCM and controls by itself. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 22 ++++++++++--- sound/pci/hda/hda_codec.c | 67 --------------------------------------- sound/pci/hda/hda_codec.h | 2 -- sound/pci/hda/hda_intel.c | 10 ------ sound/pci/hda/hda_tegra.c | 10 ------ 5 files changed, 17 insertions(+), 94 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 311896a23cd1..a49bc45c2ea5 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -106,16 +106,28 @@ static int hda_codec_driver_probe(struct device *dev) } err = codec->preset->patch(codec); - if (err < 0) { - module_put(owner); - goto error; + if (err < 0) + goto error_module; + + err = snd_hda_codec_build_pcms(codec); + if (err < 0) + goto error_module; + err = snd_hda_codec_build_controls(codec); + if (err < 0) + goto error_module; + if (codec->card->registered) { + err = snd_card_register(codec->card); + if (err < 0) + goto error_module; } return 0; + error_module: + module_put(owner); + error: - codec->preset = NULL; - memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + snd_hda_codec_cleanup_for_unbind(codec); return err; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 2c7e481a9171..7085d3733d0d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4031,36 +4031,6 @@ const struct dev_pm_ops hda_codec_driver_pm = { NULL) }; -/** - * snd_hda_build_controls - build mixer controls - * @bus: the BUS - * - * Creates mixer controls for each codec included in the bus. - * - * Returns 0 if successful, otherwise a negative error code. - */ -int snd_hda_build_controls(struct hda_bus *bus) -{ - struct hda_codec *codec; - - list_for_each_entry(codec, &bus->codec_list, list) { - int err = snd_hda_codec_build_controls(codec); - if (err < 0) { - codec_err(codec, - "cannot build controls for #%d (error %d)\n", - codec->addr, err); - err = snd_hda_codec_reset(codec); - if (err < 0) { - codec_err(codec, - "cannot revert codec\n"); - return err; - } - } - } - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_build_controls); - /* * add standard channel maps if not specified */ @@ -4692,43 +4662,6 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) return 0; } -/** - * snd_hda_build_pcms - build PCM information - * @bus: the BUS - * - * Create PCM information for each codec included in the bus. - * - * The build_pcms codec patch is requested to create and assign new - * hda_pcm objects. The codec is responsible to call snd_hda_codec_pcm_new() - * and fills the fields. Later they are instantiated by this function. - * - * At least, substreams, channels_min and channels_max must be filled for - * each stream. substreams = 0 indicates that the stream doesn't exist. - * When rates and/or formats are zero, the supported values are queried - * from the given nid. The nid is used also by the default ops.prepare - * and ops.cleanup callbacks. - * - * The driver needs to call ops.open in its open callback. Similarly, - * ops.close is supposed to be called in the close callback. - * ops.prepare should be called in the prepare or hw_params callback - * with the proper parameters for set up. - * ops.cleanup should be called in hw_free for clean up of streams. - * - * This function returns 0 if successful, or a negative error code. - */ -int snd_hda_build_pcms(struct hda_bus *bus) -{ - struct hda_codec *codec; - - list_for_each_entry(codec, &bus->codec_list, list) { - int err = snd_hda_codec_build_pcms(codec); - if (err < 0) - return err; - } - return 0; -} -EXPORT_SYMBOL_GPL(snd_hda_build_pcms); - /** * snd_hda_add_new_ctls - create controls from the array * @codec: the HDA codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index fc62ca51fd35..46d253e2f266 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -516,13 +516,11 @@ void snd_hda_spdif_ctls_assign(struct hda_codec *codec, int idx, hda_nid_t nid); /* * Mixer */ -int snd_hda_build_controls(struct hda_bus *bus); int snd_hda_codec_build_controls(struct hda_codec *codec); /* * PCM */ -int snd_hda_build_pcms(struct hda_bus *bus); int snd_hda_codec_parse_pcms(struct hda_codec *codec); int snd_hda_codec_build_pcms(struct hda_codec *codec); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f7fb1b51d446..e81461a413b8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1895,16 +1895,6 @@ static int azx_probe_continue(struct azx *chip) goto out_free; } - /* create PCM streams */ - err = snd_hda_build_pcms(chip->bus); - if (err < 0) - goto out_free; - - /* create mixer controls */ - err = snd_hda_build_controls(chip->bus); - if (err < 0) - goto out_free; - err = snd_card_register(chip->card); if (err < 0) goto out_free; diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 1359fdd20f02..7586abe91dfb 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -497,16 +497,6 @@ static int hda_tegra_probe(struct platform_device *pdev) if (err < 0) goto out_free; - /* create PCM streams */ - err = snd_hda_build_pcms(chip->bus); - if (err < 0) - goto out_free; - - /* create mixer controls */ - err = snd_hda_build_controls(chip->bus); - if (err < 0) - goto out_free; - err = snd_card_register(chip->card); if (err < 0) goto out_free; From 2f35c630f7d49efdef29b58d81ed2531ddd916d9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 22:43:26 +0100 Subject: [PATCH 095/162] ALSA: hda - Use standard workqueue for unsol and jack events The events that are handled by HD-audio drivers are no frequent and urgent ones, so we can use the standard workqueue without any problem nowadays. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 26 +++++--------------------- sound/pci/hda/hda_codec.h | 2 -- sound/pci/hda/hda_intel.c | 8 ++++---- sound/pci/hda/patch_ca0132.c | 3 +-- sound/pci/hda/patch_hdmi.c | 8 +++----- sound/pci/hda/patch_via.c | 3 +-- 6 files changed, 14 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7085d3733d0d..f2ccb39a3788 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -681,7 +681,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) struct hda_bus_unsolicited *unsol; unsigned int wp; - if (!bus || !bus->workq) + if (!bus) return 0; trace_hda_unsol_event(bus, res, res_ex); @@ -693,7 +693,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) unsol->queue[wp] = res; unsol->queue[wp + 1] = res_ex; - queue_work(bus->workq, &unsol->work); + schedule_work(&unsol->work); return 0; } @@ -732,13 +732,9 @@ static void snd_hda_bus_free(struct hda_bus *bus) return; WARN_ON(!list_empty(&bus->codec_list)); - if (bus->workq) - flush_workqueue(bus->workq); + cancel_work_sync(&bus->unsol.work); if (bus->ops.private_free) bus->ops.private_free(bus); - if (bus->workq) - destroy_workqueue(bus->workq); - kfree(bus); } @@ -785,16 +781,6 @@ int snd_hda_bus_new(struct snd_card *card, INIT_LIST_HEAD(&bus->codec_list); INIT_WORK(&bus->unsol.work, process_unsol_events); - snprintf(bus->workq_name, sizeof(bus->workq_name), - "hd-audio%d", card->number); - bus->workq = create_singlethread_workqueue(bus->workq_name); - if (!bus->workq) { - dev_err(card->dev, "cannot create workqueue %s\n", - bus->workq_name); - kfree(bus); - return -ENOMEM; - } - err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops); if (err < 0) { snd_hda_bus_free(bus); @@ -1068,8 +1054,8 @@ static void hda_jackpoll_work(struct work_struct *work) if (!codec->jackpoll_interval) return; - queue_delayed_work(codec->bus->workq, &codec->jackpoll_work, - codec->jackpoll_interval); + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); } static void init_hda_cache(struct hda_cache_rec *cache, @@ -1178,7 +1164,6 @@ static void codec_release_pcms(struct hda_codec *codec) void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) { cancel_delayed_work_sync(&codec->jackpoll_work); - flush_workqueue(codec->bus->workq); if (!codec->in_freeing) snd_hda_ctls_clear(codec); codec_release_pcms(codec); @@ -1214,7 +1199,6 @@ static void snd_hda_codec_free(struct hda_codec *codec) if (device_is_registered(hda_codec_dev(codec))) device_del(hda_codec_dev(codec)); free_init_pincfgs(codec); - flush_workqueue(codec->bus->workq); list_del(&codec->list); codec->bus->caddr_tbl[codec->addr] = NULL; clear_bit(codec->addr, &codec->bus->codec_powered); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 46d253e2f266..bf9efb7e1b9a 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -132,8 +132,6 @@ struct hda_bus { /* unsolicited event queue */ struct hda_bus_unsolicited unsol; - char workq_name[16]; - struct workqueue_struct *workq; /* common workqueue for codecs */ /* assigned PCMs */ DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e81461a413b8..dbc5a593da46 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -528,10 +528,10 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) if (ok == 1) { azx_dev->irq_pending = 0; return ok; - } else if (ok == 0 && chip->bus && chip->bus->workq) { + } else if (ok == 0) { /* bogus IRQ, process it later */ azx_dev->irq_pending = 1; - queue_work(chip->bus->workq, &hda->irq_pending_work); + schedule_work(&hda->irq_pending_work); } return 0; } @@ -893,8 +893,8 @@ static int azx_runtime_resume(struct device *dev) if (status && bus) { list_for_each_entry(codec, &bus->codec_list, list) if (status & (1 << codec->addr)) - queue_delayed_work(codec->bus->workq, - &codec->jackpoll_work, codec->jackpoll_interval); + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); } /* disable controller Wake Up event*/ diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 555781fad26f..72d20652df50 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4410,8 +4410,7 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) * state machine run. */ cancel_delayed_work_sync(&spec->unsol_hp_work); - queue_delayed_work(codec->bus->workq, &spec->unsol_hp_work, - msecs_to_jiffies(500)); + schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); cb->tbl->block_report = 1; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 708bbed15ea3..7e9ff7b16e56 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1576,9 +1576,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) update_eld = true; } else if (repoll) { - queue_delayed_work(codec->bus->workq, - &per_pin->work, - msecs_to_jiffies(300)); + schedule_delayed_work(&per_pin->work, + msecs_to_jiffies(300)); goto unlock; } } @@ -2198,11 +2197,10 @@ static void generic_hdmi_free(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx); - cancel_delayed_work(&per_pin->work); + cancel_delayed_work_sync(&per_pin->work); eld_proc_free(per_pin); } - flush_workqueue(codec->bus->workq); hdmi_array_free(spec); kfree(spec); } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 11a05638e03b..2112fbe9e577 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -222,8 +222,7 @@ static void vt1708_update_hp_work(struct hda_codec *codec) if (!spec->hp_work_active) { codec->jackpoll_interval = msecs_to_jiffies(100); snd_hda_codec_write(codec, 0x1, 0, 0xf81, 0); - queue_delayed_work(codec->bus->workq, - &codec->jackpoll_work, 0); + schedule_delayed_work(&codec->jackpoll_work, 0); spec->hp_work_active = true; } } else if (!hp_detect_with_aa(codec)) From d56db741b8e688a0b9d4d5bb9caa11dfcb7c0b08 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 27 Feb 2015 23:25:35 +0100 Subject: [PATCH 096/162] ALSA: hda - Release resources in device release callback Move the destructor code to device release callback for the codec object instead. This is a safer place to release the resources than dev_free callback in general. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 51 ++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f2ccb39a3788..6fecf57c8d7c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1191,28 +1191,6 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) remove_conn_list(codec); } -static void snd_hda_codec_free(struct hda_codec *codec) -{ - if (!codec) - return; - codec->in_freeing = 1; - if (device_is_registered(hda_codec_dev(codec))) - device_del(hda_codec_dev(codec)); - free_init_pincfgs(codec); - list_del(&codec->list); - codec->bus->caddr_tbl[codec->addr] = NULL; - clear_bit(codec->addr, &codec->bus->codec_powered); - snd_hda_sysfs_clear(codec); - free_hda_cache(&codec->amp_cache); - free_hda_cache(&codec->cmd_cache); - kfree(codec->vendor_name); - kfree(codec->chip_name); - kfree(codec->modelname); - kfree(codec->wcaps); - codec->bus->num_codecs--; - put_device(hda_codec_dev(codec)); -} - static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1241,14 +1219,32 @@ static int snd_hda_codec_dev_disconnect(struct snd_device *device) static int snd_hda_codec_dev_free(struct snd_device *device) { - snd_hda_codec_free(device->device_data); + struct hda_codec *codec = device->device_data; + + codec->in_freeing = 1; + if (device_is_registered(hda_codec_dev(codec))) + device_del(hda_codec_dev(codec)); + put_device(hda_codec_dev(codec)); return 0; } -/* just free the container */ static void snd_hda_codec_dev_release(struct device *dev) { - kfree(dev_to_hda_codec(dev)); + struct hda_codec *codec = dev_to_hda_codec(dev); + + free_init_pincfgs(codec); + list_del(&codec->list); + codec->bus->caddr_tbl[codec->addr] = NULL; + clear_bit(codec->addr, &codec->bus->codec_powered); + snd_hda_sysfs_clear(codec); + free_hda_cache(&codec->amp_cache); + free_hda_cache(&codec->cmd_cache); + kfree(codec->vendor_name); + kfree(codec->chip_name); + kfree(codec->modelname); + kfree(codec->wcaps); + codec->bus->num_codecs--; + kfree(codec); } /** @@ -1362,7 +1358,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, setup_fg_nodes(codec); if (!codec->afg && !codec->mfg) { - dev_err(card->dev, "no AFG or MFG node found\n"); + codec_err(codec, "no AFG or MFG node found\n"); err = -ENODEV; goto error; } @@ -1408,7 +1404,7 @@ int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card, return 0; error: - snd_hda_codec_free(codec); + put_device(hda_codec_dev(codec)); return err; } EXPORT_SYMBOL_GPL(snd_hda_codec_new); @@ -2464,7 +2460,6 @@ void snd_hda_unlock_devices(struct hda_bus *bus) { struct snd_card *card = bus->card; - card = bus->card; spin_lock(&card->files_lock); card->shutdown = 0; spin_unlock(&card->files_lock); From 0004defd4e44d81966b0c4164c2ee01f20ab357b Mon Sep 17 00:00:00 2001 From: Vishal Thanki Date: Tue, 3 Mar 2015 18:59:00 +0530 Subject: [PATCH 097/162] ASoC: simple-card: Add a NULL pointer check in asoc_simple_card_dai_link_of Make sure devm_kzalloc() succeeds. Signed-off-by: Vishal Thanki Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f7c6734bd5da..fb550b5869d2 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -372,6 +372,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, strlen(dai_link->cpu_dai_name) + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); + if (!name) { + ret = -ENOMEM; + goto dai_link_of_err; + } + sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); dai_link->name = dai_link->stream_name = name; From 1a6ab46fa9c2bc9399694b4856ab7ea19c036485 Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Wed, 4 Mar 2015 10:56:13 +0900 Subject: [PATCH 098/162] ALSA: Fix spelling typo in Documentation/DocBook/alsa-driver-api.xml This patch fix spelling typo found in alsa-driver-api.xml. It is because this file is generated from comments in source files, I have to fix source files. Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai --- include/sound/compress_driver.h | 4 ++-- include/sound/control.h | 2 +- include/sound/soc.h | 2 +- include/uapi/sound/compress_offload.h | 2 +- sound/core/pcm_dmaengine.c | 4 ++-- 5 files changed, 7 insertions(+), 7 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index f48089d364c5..fa1d05512c09 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -70,7 +70,7 @@ struct snd_compr_runtime { * @device: device pointer * @direction: stream direction, playback/recording * @metadata_set: metadata set flag, true when set - * @next_track: has userspace signall next track transistion, true when set + * @next_track: has userspace signal next track transition, true when set * @private_data: pointer to DSP private data */ struct snd_compr_stream { @@ -95,7 +95,7 @@ struct snd_compr_stream { * and the stream properties * @get_params: retrieve the codec parameters, mandatory * @set_metadata: Set the metadata values for a stream - * @get_metadata: retreives the requested metadata values from stream + * @get_metadata: retrieves the requested metadata values from stream * @trigger: Trigger operations like start, pause, resume, drain, stop. * This callback is mandatory * @pointer: Retrieve current h/w pointer information. Mandatory diff --git a/include/sound/control.h b/include/sound/control.h index 75f3054023f7..95aad6d3fd1a 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -227,7 +227,7 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) * Add a virtual slave control to the given master. * Unlike snd_ctl_add_slave(), the element added via this function * is supposed to have volatile values, and get callback is called - * at each time quried from the master. + * at each time queried from the master. * * When the control peeks the hardware values directly and the value * can be changed by other means than the put callback of the element, diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade195628..cf0bb156d6da 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1469,7 +1469,7 @@ static inline struct snd_soc_codec *snd_soc_kcontrol_codec( } /** - * snd_soc_kcontrol_platform() - Returns the platform that registerd the control + * snd_soc_kcontrol_platform() - Returns the platform that registered the control * @kcontrol: The control for which to get the platform * * Note: This function will only work correctly if the control has been diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 22ed8cb7800b..e00d8cbfc628 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -75,7 +75,7 @@ struct snd_compr_tstamp { /** * struct snd_compr_avail - avail descriptor * @avail: Number of bytes available in ring buffer for writing/reading - * @tstamp: timestamp infomation + * @tstamp: timestamp information */ struct snd_compr_avail { __u64 avail; diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 6542c4083594..fba365a78390 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -289,7 +289,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_request_channel); * * The function should usually be called from the pcm open callback. Note that * this function will use private_data field of the substream's runtime. So it - * is not availabe to your pcm driver implementation. + * is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, struct dma_chan *chan) @@ -328,7 +328,7 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); * This function will request a DMA channel using the passed filter function and * data. The function should usually be called from the pcm open callback. Note * that this function will use private_data field of the substream's runtime. So - * it is not availabe to your pcm driver implementation. + * it is not available to your pcm driver implementation. */ int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) From 8b28c93fe5a55873ce22b7126e84eb59290f8603 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Mar 2015 15:37:01 +0100 Subject: [PATCH 099/162] ALSA: usb-audio: Check Marantz/Denon USB DACs in a single place There are three places doing the same check. Let's make them together. Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 40 +++++++++++++++++----------------------- 1 file changed, 17 insertions(+), 23 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 753a47de8459..353532b8aee4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1120,17 +1120,24 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) /* Marantz/Denon USB DACs need a vendor cmd to switch * between PCM and native DSD mode */ +static bool is_marantz_denon_dac(unsigned int id) +{ + switch (id) { + case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ + case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ + case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + return true; + } + return false; +} + int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, struct audioformat *fmt) { struct usb_device *dev = subs->dev; int err; - switch (subs->stream->chip->usb_id) { - case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ - case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ - case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ - + if (is_marantz_denon_dac(subs->stream->chip->usb_id)) { /* First switch to alt set 0, otherwise the mode switch cmd * will not be accepted by the DAC */ @@ -1203,17 +1210,10 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, /* Marantz/Denon devices with USB DAC functionality need a delay * after each class compliant request */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x154e) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { - - switch (le16_to_cpu(dev->descriptor.idProduct)) { - case 0x1003: /* Denon DA300-USB */ - case 0x3005: /* Marantz HD-DAC1 */ - case 0x3006: /* Marantz SA-14S1 */ - mdelay(20); - break; - } - } + if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), + le16_to_cpu(dev->descriptor.idProduct))) + && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + mdelay(20); /* Zoom R16/24 needs a tiny delay here, otherwise requests like * get/set frequency return as failed despite actually succeeding. @@ -1268,15 +1268,9 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, } /* Denon/Marantz devices with USB DAC functionality */ - switch (chip->usb_id) { - case USB_ID(0x154e, 0x1003): /* Denon DA300-USB */ - case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ - case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + if (is_marantz_denon_dac(chip->usb_id)) { if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; - break; - default: - break; } return 0; From c472b93990e02c31f02322ddf0fdd9d571169310 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:16 +0100 Subject: [PATCH 100/162] ASoC: sn95031: Pass CODEC to sn95031_jack_detection() The sn95031 driver currently gets the CODEC implicitly from the jack that is passed to sn95031_jack_detection(). But the codec field is going to be removed from the snd_soc_jack struct, so refactor things to pass the CODEC explicitly. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 14 ++++++++------ sound/soc/codecs/sn95031.h | 3 ++- sound/soc/intel/mfld_machine.c | 13 +++++++------ 3 files changed, 17 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 47b257e41809..1e5d2643c286 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -783,19 +783,21 @@ static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_BTNCTRL2, 0x01); } -static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack) +static int sn95031_get_headset_state(struct snd_soc_codec *codec, + struct snd_soc_jack *mfld_jack) { - int micbias = sn95031_get_mic_bias(mfld_jack->codec); + int micbias = sn95031_get_mic_bias(codec); int jack_type = snd_soc_jack_get_type(mfld_jack, micbias); pr_debug("jack type detected = %d\n", jack_type); if (jack_type == SND_JACK_HEADSET) - sn95031_enable_jack_btn(mfld_jack->codec); + sn95031_enable_jack_btn(codec); return jack_type; } -void sn95031_jack_detection(struct mfld_jack_data *jack_data) +void sn95031_jack_detection(struct snd_soc_codec *codec, + struct mfld_jack_data *jack_data) { unsigned int status; unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET; @@ -809,11 +811,11 @@ void sn95031_jack_detection(struct mfld_jack_data *jack_data) status = SND_JACK_HEADSET | SND_JACK_BTN_1; } else if (jack_data->intr_id & 0x4) { pr_debug("headset or headphones inserted\n"); - status = sn95031_get_headset_state(jack_data->mfld_jack); + status = sn95031_get_headset_state(codec, jack_data->mfld_jack); } else if (jack_data->intr_id & 0x8) { pr_debug("headset or headphones removed\n"); status = 0; - sn95031_disable_jack_btn(jack_data->mfld_jack->codec); + sn95031_disable_jack_btn(codec); } else { pr_err("unidentified interrupt\n"); return; diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h index 20376d234fb8..7651fe4e6a45 100644 --- a/sound/soc/codecs/sn95031.h +++ b/sound/soc/codecs/sn95031.h @@ -127,6 +127,7 @@ struct mfld_jack_data { struct snd_soc_jack *mfld_jack; }; -extern void sn95031_jack_detection(struct mfld_jack_data *jack_data); +extern void sn95031_jack_detection(struct snd_soc_codec *codec, + struct mfld_jack_data *jack_data); #endif diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index 90b7a57713a0..d22b44db824e 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -228,10 +228,13 @@ static void mfld_jack_check(unsigned int intr_status) { struct mfld_jack_data jack_data; + if (!mfld_codec) + return; + jack_data.mfld_jack = &mfld_jack; jack_data.intr_id = intr_status; - sn95031_jack_detection(&jack_data); + sn95031_jack_detection(mfld_codec, &jack_data); /* TODO: add american headset detection post gpiolib support */ } @@ -240,8 +243,6 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) struct snd_soc_dapm_context *dapm = &runtime->card->dapm; int ret_val; - mfld_codec = runtime->codec; - /* default is earpiece pin, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "Headphones"); /* default is lineout NC, userspace sets it explcitly */ @@ -254,7 +255,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(mfld_codec, "Intel(R) MID Audio Jack", + ret_val = snd_soc_jack_new(runtime->codec, "Intel(R) MID Audio Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack); if (ret_val) { @@ -275,6 +276,8 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) return ret_val; } + mfld_codec = runtime->codec; + /* we want to check if anything is inserted at boot, * so send a fake event to codec and it will read adc * to find if anything is there or not */ @@ -359,8 +362,6 @@ static irqreturn_t snd_mfld_jack_detection(int irq, void *data) { struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data; - if (mfld_jack.codec == NULL) - return IRQ_HANDLED; mfld_jack_check(mc_drv_ctx->interrupt_status); return IRQ_HANDLED; From 970939964c26db4643f58d4e84487821962e0b65 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:17 +0100 Subject: [PATCH 101/162] ASoC: Allow to register jacks at the card level Jacks are typically card level elements, but are currently registered with a CODEC. When it was originally introduced snd_soc_jack_new() took a snd_soc_card as its parameter, but at that time DAPM was only implemented at the CODEC level and there was only one CODEC per card. This made it clear which CODEC to use for the jack DAPM operations. But the multi-component patchset added support for having multiple CODECs per card and with it the API was updated to register jacks with a specific CODEC instance instead. Subsequently DAPM support at the card level has been introduced, but the snd_soc_jack_new() API has so remained unchanged. This leaves us with the issue that the DAPM pins that are managed by the jack detection logic usually are part of the card DAPM context but are accessed through a CODEC DAPM context. Currently this works fine, but might break in the future if we take a more hierarchical approach to DAPM contexts. Furthermore with componentization progressing systems that do not register a snd_soc_codec might appear, while these system may still want to able to register a jack. This patch addresses these issues by adding a new function called snd_soc_card_jack_new() that can be used to register jacks with the card rather than a CODEC. This new function is mostly identical to snd_soc_jack_new() except that it additionally allows to directly specify the DAPM pins associated with the jack. This was done since most users of snd_soc_jack_new() typically call snd_soc_jack_add_pins() right after it, which is not necessary with the new API and allows to reduce the amount of boiler plate code. The old snd_soc_jack_new() is re-implemented as a wrapper around snd_soc_card_jack_new(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 28 +++++++++++++++++++++++++--- sound/soc/soc-jack.c | 42 ++++++++++++++++++++++++++---------------- 2 files changed, 51 insertions(+), 19 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0d1ade195628..99d9ce272209 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -450,8 +450,10 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); /* Jack reporting */ -int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, - struct snd_soc_jack *jack); +int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, + unsigned int num_pins); + void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask); int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, struct snd_soc_jack_pin *pins); @@ -659,7 +661,7 @@ struct snd_soc_jack_gpio { struct snd_soc_jack { struct mutex mutex; struct snd_jack *jack; - struct snd_soc_codec *codec; + struct snd_soc_card *card; struct list_head pins; int status; struct blocking_notifier_head notifier; @@ -1482,6 +1484,26 @@ static inline struct snd_soc_platform *snd_soc_kcontrol_platform( return snd_soc_component_to_platform(snd_soc_kcontrol_component(kcontrol)); } +/** + * snd_soc_jack_new - Create a new jack + * @codec: ASoC CODEC + * @id: an identifying string for this jack + * @type: a bitmask of enum snd_jack_type values that can be detected by + * this jack + * @jack: structure to use for the jack + * + * Creates a new jack object. + * + * Returns zero if successful, or a negative error code on failure. + * On success jack will be initialised. + */ +static inline int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, + int type, struct snd_soc_jack *jack) +{ + return snd_soc_card_jack_new(codec->component.card, id, type, jack, + NULL, 0); +} + int snd_soc_util_init(void); void snd_soc_util_exit(void); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 4380dcc064a5..9f60c25c4568 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -22,30 +22,42 @@ #include /** - * snd_soc_jack_new - Create a new jack - * @codec: ASoC codec + * snd_soc_card_jack_new - Create a new jack + * @card: ASoC card * @id: an identifying string for this jack * @type: a bitmask of enum snd_jack_type values that can be detected by * this jack * @jack: structure to use for the jack + * @pins: Array of jack pins to be added to the jack or NULL + * @num_pins: Number of elements in the @pins array * * Creates a new jack object. * * Returns zero if successful, or a negative error code on failure. * On success jack will be initialised. */ -int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, - struct snd_soc_jack *jack) +int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, + struct snd_soc_jack *jack, struct snd_soc_jack_pin *pins, + unsigned int num_pins) { + int ret; + mutex_init(&jack->mutex); - jack->codec = codec; + jack->card = card; INIT_LIST_HEAD(&jack->pins); INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); - return snd_jack_new(codec->component.card->snd_card, id, type, &jack->jack); + ret = snd_jack_new(card->snd_card, id, type, &jack->jack); + if (ret) + return ret; + + if (num_pins) + return snd_soc_jack_add_pins(jack, num_pins, pins); + + return 0; } -EXPORT_SYMBOL_GPL(snd_soc_jack_new); +EXPORT_SYMBOL_GPL(snd_soc_card_jack_new); /** * snd_soc_jack_report - Report the current status for a jack @@ -63,7 +75,6 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new); */ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) { - struct snd_soc_codec *codec; struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; unsigned int sync = 0; @@ -74,8 +85,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) if (!jack) return; - codec = jack->codec; - dapm = &codec->dapm; + dapm = &jack->card->dapm; mutex_lock(&jack->mutex); @@ -175,12 +185,12 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { if (!pins[i].pin) { - dev_err(jack->codec->dev, "ASoC: No name for pin %d\n", + dev_err(jack->card->dev, "ASoC: No name for pin %d\n", i); return -EINVAL; } if (!pins[i].mask) { - dev_err(jack->codec->dev, "ASoC: No mask for pin %d" + dev_err(jack->card->dev, "ASoC: No mask for pin %d" " (%s)\n", i, pins[i].pin); return -EINVAL; } @@ -260,7 +270,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) static irqreturn_t gpio_handler(int irq, void *data) { struct snd_soc_jack_gpio *gpio = data; - struct device *dev = gpio->jack->codec->component.card->dev; + struct device *dev = gpio->jack->card->dev; trace_snd_soc_jack_irq(gpio->name); @@ -299,7 +309,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, for (i = 0; i < count; i++) { if (!gpios[i].name) { - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: No name for gpio at index %d\n", i); ret = -EINVAL; goto undo; @@ -320,7 +330,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, } else { /* legacy GPIO number */ if (!gpio_is_valid(gpios[i].gpio)) { - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: Invalid gpio %d\n", gpios[i].gpio); ret = -EINVAL; @@ -350,7 +360,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, if (gpios[i].wake) { ret = irq_set_irq_wake(gpiod_to_irq(gpios[i].desc), 1); if (ret != 0) - dev_err(jack->codec->dev, + dev_err(jack->card->dev, "ASoC: Failed to mark GPIO at index %d as wake source: %d\n", i, ret); } From 386669fcec85a16cb81cd19236abe76abe0f1fc1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:18 +0100 Subject: [PATCH 102/162] ASoC: simple-card: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/generic/simple-card.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index f7c6734bd5da..b8ee47b7ba9c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -176,11 +176,11 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) return ret; if (gpio_is_valid(priv->gpio_hp_det)) { - snd_soc_jack_new(codec->codec, "Headphones", SND_JACK_HEADPHONE, - &simple_card_hp_jack); - snd_soc_jack_add_pins(&simple_card_hp_jack, - ARRAY_SIZE(simple_card_hp_jack_pins), - simple_card_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", + SND_JACK_HEADPHONE, + &simple_card_hp_jack, + simple_card_hp_jack_pins, + ARRAY_SIZE(simple_card_hp_jack_pins)); simple_card_hp_jack_gpio.gpio = priv->gpio_hp_det; simple_card_hp_jack_gpio.invert = priv->gpio_hp_det_invert; @@ -189,11 +189,11 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) } if (gpio_is_valid(priv->gpio_mic_det)) { - snd_soc_jack_new(codec->codec, "Mic Jack", SND_JACK_MICROPHONE, - &simple_card_mic_jack); - snd_soc_jack_add_pins(&simple_card_mic_jack, - ARRAY_SIZE(simple_card_mic_jack_pins), - simple_card_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, + &simple_card_mic_jack, + simple_card_mic_jack_pins, + ARRAY_SIZE(simple_card_mic_jack_pins)); simple_card_mic_jack_gpio.gpio = priv->gpio_mic_det; simple_card_mic_jack_gpio.invert = priv->gpio_mic_det_invert; snd_soc_jack_add_gpios(&simple_card_mic_jack, 1, From 27cb64b474516421001932d966ca3184795d4e29 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:19 +0100 Subject: [PATCH 103/162] ASoC: imx-es8328: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-es8328.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index f8cf10e16ce9..20e7400e2611 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -53,9 +53,9 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) /* Headphone jack detection */ if (gpio_is_valid(data->jack_gpio)) { - ret = snd_soc_jack_new(rtd->codec, "Headphone", - SND_JACK_HEADPHONE | SND_JACK_BTN_0, - &headset_jack); + ret = snd_soc_card_jack_new(rtd->card, "Headphone", + SND_JACK_HEADPHONE | SND_JACK_BTN_0, + &headset_jack, NULL, 0); if (ret) return ret; From 47ec96d4ca7e4a7b9b8b115a10d59e89f794ef95 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:20 +0100 Subject: [PATCH 104/162] ASoC: wm1133-ev: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/fsl/wm1133-ev1.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index a958937ab405..0653aa83c927 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -205,16 +205,14 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; /* Headphone jack detection */ - snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack); - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE); /* Microphone jack detection */ - snd_soc_jack_new(codec, "Microphone", - SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Microphone", + SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack, + mic_jack_pins, ARRAY_SIZE(mic_jack_pins)); wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE, SND_JACK_BTN_0); From 85c85e5d6d579a5ff8b5471c4e753946eedbf788 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:21 +0100 Subject: [PATCH 105/162] ASoC: broadwell: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/broadwell.c | 12 +++--------- 1 file changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c index 9cf7d01479ad..9effa3da982f 100644 --- a/sound/soc/intel/broadwell.c +++ b/sound/soc/intel/broadwell.c @@ -80,15 +80,9 @@ static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; int ret = 0; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset); - - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&broadwell_headset, - ARRAY_SIZE(broadwell_headset_pins), - broadwell_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset, + broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins)); if (ret) return ret; From e0f7dd9d88f4c151aeca45d290e171d907249888 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:22 +0100 Subject: [PATCH 106/162] ASoC: byt-max98090: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/byt-max98090.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 9832afe7d22c..d8b1f038da1c 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -84,7 +84,6 @@ static struct snd_soc_jack_gpio hs_jack_gpios[] = { static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) { int ret; - struct snd_soc_codec *codec = runtime->codec; struct snd_soc_card *card = runtime->card; struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card); struct snd_soc_jack *jack = &drv->jack; @@ -100,13 +99,9 @@ static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime) } /* Enable jack detection */ - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET, jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_LINEOUT | SND_JACK_HEADSET, jack, + hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; From fb1edb4b68a829619bcd50a0c23c557000d0d638 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:23 +0100 Subject: [PATCH 107/162] ASoC: cht_bsw_rt5645: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/intel/cht_bsw_rt5645.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/cht_bsw_rt5645.c b/sound/soc/intel/cht_bsw_rt5645.c index bd29617a9ab9..0bfca2192ca0 100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/cht_bsw_rt5645.c @@ -169,17 +169,17 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) return ret; } - ret = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, - &ctx->hp_jack); + ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack", + SND_JACK_HEADPHONE, &ctx->hp_jack, + NULL, 0); if (ret) { dev_err(runtime->dev, "HP jack creation failed %d\n", ret); return ret; } - ret = snd_soc_jack_new(codec, "Mic Jack", - SND_JACK_MICROPHONE, - &ctx->mic_jack); + ret = snd_soc_card_jack_new(runtime->card, "Mic Jack", + SND_JACK_MICROPHONE, &ctx->mic_jack, + NULL, 0); if (ret) { dev_err(runtime->dev, "Mic jack creation failed %d\n", ret); return ret; From af13cbc1a288d3921f1af739da84371e6c53aea3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:24 +0100 Subject: [PATCH 108/162] ASoC: mfld_machine: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/mfld_machine.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index d22b44db824e..49c09a0add79 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -255,20 +255,15 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(runtime->codec, "Intel(R) MID Audio Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1, &mfld_jack); + ret_val = snd_soc_card_jack_new(runtime->card, + "Intel(R) MID Audio Jack", SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack, + mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins)); if (ret_val) { pr_err("jack creation failed\n"); return ret_val; } - ret_val = snd_soc_jack_add_pins(&mfld_jack, - ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins); - if (ret_val) { - pr_err("adding jack pins failed\n"); - return ret_val; - } ret_val = snd_soc_jack_add_zones(&mfld_jack, ARRAY_SIZE(mfld_zones), mfld_zones); if (ret_val) { From df8c66189dd42f719c75800a526bdc901f300f41 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:25 +0100 Subject: [PATCH 109/162] ASoC: ams-deltea: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/ams-delta.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 706613077c15..16cc95fa4573 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -479,8 +479,8 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ - ret = snd_soc_jack_new(rtd->codec, "hook_switch", - SND_JACK_HEADSET, &ams_delta_hook_switch); + ret = snd_soc_card_jack_new(card, "hook_switch", SND_JACK_HEADSET, + &ams_delta_hook_switch, NULL, 0); if (ret) dev_warn(card->dev, "Failed to allocate resources for hook switch, " From 25649592cfa6c210c9f86670472b864782c8d677 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:26 +0100 Subject: [PATCH 110/162] ASoC: omap-abe-twl6040: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index b9c65f1ad5a8..0843a68f277c 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -182,17 +182,17 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* Headset jack detection only if it is supported */ if (priv->jack_detection) { - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET, &hs_jack); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, &hs_jack, + hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - return ret; + return 0; } static const struct snd_soc_dapm_route dmic_audio_map[] = { From da21cf6d65283680247da74c3d03f7e5cdfb40d1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:27 +0100 Subject: [PATCH 111/162] ASoC: omap-twl4030: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-twl4030.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index fb1f6bb87cd4..3673ada43bfb 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -170,14 +170,10 @@ static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) if (priv->jack_detect > 0) { hs_jack_gpios[0].gpio = priv->jack_detect; - ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &priv->hs_jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&priv->hs_jack, - ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET, &priv->hs_jack, + hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) return ret; From 753d45e6b886c93a2a8a88eddaca345643a87f4e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:28 +0100 Subject: [PATCH 112/162] ASoC: rx51: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/rx51.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 7f299357c2d2..c2ddf0fbfa28 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -311,9 +311,9 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) } /* AV jack detection */ - err = snd_soc_jack_new(codec, "AV Jack", - SND_JACK_HEADSET | SND_JACK_VIDEOOUT, - &rx51_av_jack); + err = snd_soc_card_jack_new(rtd->card, "AV Jack", + SND_JACK_HEADSET | SND_JACK_VIDEOOUT, + &rx51_av_jack, NULL, 0); if (err) { dev_err(card->dev, "Failed to add AV Jack\n"); return err; From f7a4433b498384f0e300c51b654910f3e03b5ca6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:29 +0100 Subject: [PATCH 113/162] ASoC: hx4700: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/hx4700.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 73eb5ddf9753..9f8be7cd567e 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -126,17 +126,12 @@ static const struct snd_soc_dapm_route hx4700_audio_map[] = { */ static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; int err; /* Jack detection API stuff */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pin), - hs_jack_pin); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack, hs_jack_pin, + ARRAY_SIZE(hs_jack_pin)); if (err) return err; From bc1e2e06a07ad4c0c021165b34fa8259bdf4d8c6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:30 +0100 Subject: [PATCH 114/162] ASoC: palm27x: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/palm27x.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 910336c5ebeb..c20bbc042425 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -75,17 +75,12 @@ static struct snd_soc_card palm27x_asoc; static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; int err; /* Jack detection API stuff */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &hs_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &hs_jack, hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (err) return err; From 3b14125bc553a0fe091a5d43a22be41cdc43b156 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:31 +0100 Subject: [PATCH 115/162] ASoC: ttc-dkb: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/ttc-dkb.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index 5001dbb9b257..1753c7d9e760 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -78,15 +78,12 @@ static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; /* Headset jack detection */ - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE - | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, - &hs_jack); - snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, - &mic_jack); - snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), - mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack, hs_jack_pins, ARRAY_SIZE(hs_jack_pins)); + snd_soc_card_jack_new(rtd->card, "Microphone Jack", SND_JACK_MICROPHONE, + &mic_jack, mic_jack_pins, + ARRAY_SIZE(mic_jack_pins)); /* headphone, microphone detection & headset short detection */ pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, From d30d141f9cb7eb9fb3f03af11146dc0d2b393ff2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:32 +0100 Subject: [PATCH 116/162] ASoC: z2: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/pxa/z2.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 76ccb172d0a7..bcbfbe8303f7 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -143,13 +143,9 @@ static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "MONO1"); /* Jack detection API stuff */ - ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &hs_jack); - if (ret) - goto err; - - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, + &hs_jack, hs_jack_pins, + ARRAY_SIZE(hs_jack_pins)); if (ret) goto err; From dfe11f282c61808f7140d9dd741f7e54cf97cda6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:33 +0100 Subject: [PATCH 117/162] ASoC: h1980_uda1380: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/h1940_uda1380.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 59b044255b78..c72e9fb26658 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -162,13 +162,8 @@ static struct platform_device *s3c24xx_snd_device; static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &hp_jack); - - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); From 39ec5109d6089e1acd04b51b9df5349f5b8a7f5c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:34 +0100 Subject: [PATCH 118/162] ASoC: littlemill: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/littlemill.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 141519c21e21..31a820eb0ac3 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -260,12 +260,12 @@ static int littlemill_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_MECHANICAL | - SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | - SND_JACK_BTN_4 | SND_JACK_BTN_5, - &littlemill_headset); + ret = snd_soc_card_jack_new(card, "Headset", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | + SND_JACK_BTN_4 | SND_JACK_BTN_5, + &littlemill_headset, NULL, 0); if (ret) return ret; From f97e0eacf2b5d9c1a470e53df60519d555ac5a75 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:35 +0100 Subject: [PATCH 119/162] ASoC: lowland: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/lowland.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 243dea7ba38f..5f156093101e 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -56,16 +56,10 @@ static int lowland_wm5100_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET | - SND_JACK_BTN_0, - &lowland_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&lowland_headset, - ARRAY_SIZE(lowland_headset_pins), - lowland_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0, + &lowland_headset, lowland_headset_pins, + ARRAY_SIZE(lowland_headset_pins)); if (ret) return ret; From e9c9a723eea5102fa6adedf454e02fff6201a3c3 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:36 +0100 Subject: [PATCH 120/162] ASoC: rx1950_uda1380: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/rx1950_uda1380.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 873f2cb4bebe..35e37c457f1f 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -211,13 +211,8 @@ static int rx1950_hw_params(struct snd_pcm_substream *substream, static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &hp_jack); - - snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), - hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); From 55b2ed2d9dd8c611837f34ca29df881eb0a1de8d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:37 +0100 Subject: [PATCH 121/162] ASoC: smartq: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/smartq_wm8987.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 8291d2a5f152..dfbe2db1c407 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -151,13 +151,10 @@ static int smartq_wm8987_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); /* Headphone jack detection */ - err = snd_soc_jack_new(codec, "Headphone Jack", - SND_JACK_HEADPHONE, &smartq_jack); - if (err) - return err; - - err = snd_soc_jack_add_pins(&smartq_jack, ARRAY_SIZE(smartq_jack_pins), - smartq_jack_pins); + err = snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &smartq_jack, + smartq_jack_pins, + ARRAY_SIZE(smartq_jack_pins)); if (err) return err; From 663976ad478b50664353fdf19a5a3dcad3cb4e22 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:38 +0100 Subject: [PATCH 122/162] ASoC: speyside: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/speyside.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 5ec7c52282f2..2dcb988bdff2 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -153,16 +153,10 @@ static int speyside_wm8996_init(struct snd_soc_pcm_runtime *rtd) pr_err("Failed to request HP_SEL GPIO: %d\n", ret); gpio_direction_output(WM8996_HPSEL_GPIO, speyside_jack_polarity); - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_LINEOUT | SND_JACK_HEADSET | - SND_JACK_BTN_0, - &speyside_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&speyside_headset, - ARRAY_SIZE(speyside_headset_pins), - speyside_headset_pins); + ret = snd_soc_card_jack_new(rtd->card, "Headset", SND_JACK_LINEOUT | + SND_JACK_HEADSET | SND_JACK_BTN_0, + &speyside_headset, speyside_headset_pins, + ARRAY_SIZE(speyside_headset_pins)); if (ret) return ret; From 3fd94f37da000a2b562a3f4e6c553b7ab1ad9e19 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:39 +0100 Subject: [PATCH 123/162] ASoC: tobermory: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/samsung/tobermory.c | 13 ++++--------- 1 file changed, 4 insertions(+), 9 deletions(-) diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 9c80506527c4..85ccfb7188cb 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -179,15 +179,10 @@ static int tobermory_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; - ret = snd_soc_jack_new(codec, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, - &tobermory_headset); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&tobermory_headset, - ARRAY_SIZE(tobermory_headset_pins), - tobermory_headset_pins); + ret = snd_soc_card_jack_new(card, "Headset", SND_JACK_HEADSET | + SND_JACK_BTN_0, &tobermory_headset, + tobermory_headset_pins, + ARRAY_SIZE(tobermory_headset_pins)); if (ret) return ret; From 12cc6d1dca4d3a9e929090cb0cf9ef452f414518 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:40 +0100 Subject: [PATCH 124/162] ASoC: tegra_alc5632: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 769aca2fc5f5..6dcd06a966c7 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -106,11 +106,10 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, - &tegra_alc5632_hs_jack); - snd_soc_jack_add_pins(&tegra_alc5632_hs_jack, - ARRAY_SIZE(tegra_alc5632_hs_jack_pins), - tegra_alc5632_hs_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headset Jack", SND_JACK_HEADSET, + &tegra_alc5632_hs_jack, + tegra_alc5632_hs_jack_pins, + ARRAY_SIZE(tegra_alc5632_hs_jack_pins)); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; From d020e77c61b8a9d563d205cfcec7e71090d1377d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:41 +0100 Subject: [PATCH 125/162] ASoC: tegra_max98090: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_max98090.c | 22 ++++++++++------------ 1 file changed, 10 insertions(+), 12 deletions(-) diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index af3fb997b752..6760f0ebc133 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -141,16 +141,14 @@ static const struct snd_kcontrol_new tegra_max98090_controls[] = { static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; struct tegra_max98090 *machine = snd_soc_card_get_drvdata(rtd->card); if (gpio_is_valid(machine->gpio_hp_det)) { - snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, - &tegra_max98090_hp_jack); - snd_soc_jack_add_pins(&tegra_max98090_hp_jack, - ARRAY_SIZE(tegra_max98090_hp_jack_pins), - tegra_max98090_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", + SND_JACK_HEADPHONE, + &tegra_max98090_hp_jack, + tegra_max98090_hp_jack_pins, + ARRAY_SIZE(tegra_max98090_hp_jack_pins)); tegra_max98090_hp_jack_gpio.gpio = machine->gpio_hp_det; snd_soc_jack_add_gpios(&tegra_max98090_hp_jack, @@ -159,11 +157,11 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) } if (gpio_is_valid(machine->gpio_mic_det)) { - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_max98090_mic_jack); - snd_soc_jack_add_pins(&tegra_max98090_mic_jack, - ARRAY_SIZE(tegra_max98090_mic_jack_pins), - tegra_max98090_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", + SND_JACK_MICROPHONE, + &tegra_max98090_mic_jack, + tegra_max98090_mic_jack_pins, + ARRAY_SIZE(tegra_max98090_mic_jack_pins)); tegra_max98090_mic_jack_gpio.gpio = machine->gpio_mic_det; snd_soc_jack_add_gpios(&tegra_max98090_mic_jack, From 00eafe3b1b191c9b2611b74c03e1b573ae257b1e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:42 +0100 Subject: [PATCH 126/162] ASoC: tegra_rt5640: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_rt5640.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index ed759a3076b8..773daecaa5e8 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -108,15 +108,11 @@ static const struct snd_kcontrol_new tegra_rt5640_controls[] = { static int tegra_rt5640_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; struct tegra_rt5640 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headphones", SND_JACK_HEADPHONE, - &tegra_rt5640_hp_jack); - snd_soc_jack_add_pins(&tegra_rt5640_hp_jack, - ARRAY_SIZE(tegra_rt5640_hp_jack_pins), - tegra_rt5640_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphones", SND_JACK_HEADPHONE, + &tegra_rt5640_hp_jack, tegra_rt5640_hp_jack_pins, + ARRAY_SIZE(tegra_rt5640_hp_jack_pins)); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5640_hp_jack_gpio.gpio = machine->gpio_hp_det; From 783b1e7948010ded40eba784b558d86d72ae2ef4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:43 +0100 Subject: [PATCH 127/162] ASoC: tegra_rt5677: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_rt5677.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index e4cf978a6e3a..68d8b67e79c1 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -146,10 +146,9 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_rt5677 *machine = snd_soc_card_get_drvdata(rtd->card); - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &tegra_rt5677_hp_jack); - snd_soc_jack_add_pins(&tegra_rt5677_hp_jack, 1, - &tegra_rt5677_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, + &tegra_rt5677_hp_jack, + &tegra_rt5677_hp_jack_pins, 1); if (gpio_is_valid(machine->gpio_hp_det)) { tegra_rt5677_hp_jack_gpio.gpio = machine->gpio_hp_det; @@ -158,10 +157,9 @@ static int tegra_rt5677_asoc_init(struct snd_soc_pcm_runtime *rtd) } - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_rt5677_mic_jack); - snd_soc_jack_add_pins(&tegra_rt5677_mic_jack, 1, - &tegra_rt5677_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_rt5677_mic_jack, + &tegra_rt5677_mic_jack_pins, 1); if (gpio_is_valid(machine->gpio_mic_present)) { tegra_rt5677_mic_jack_gpio.gpio = machine->gpio_mic_present; From 7ba8cbb2f0fd9ff232fa19159e2646bf64135126 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:44 +0100 Subject: [PATCH 128/162] ASoC: tegra_wm8903: Register jacks at the card level The jacks are card level elements so use snd_soc_card_jack_new() instead of snd_soc_jack_new() to register them. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_wm8903.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index e52420dae2b4..4a95b70f0cf0 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -177,21 +177,19 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) if (gpio_is_valid(machine->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det; - snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, - &tegra_wm8903_hp_jack); - snd_soc_jack_add_pins(&tegra_wm8903_hp_jack, - ARRAY_SIZE(tegra_wm8903_hp_jack_pins), - tegra_wm8903_hp_jack_pins); + snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &tegra_wm8903_hp_jack, + tegra_wm8903_hp_jack_pins, + ARRAY_SIZE(tegra_wm8903_hp_jack_pins)); snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, 1, &tegra_wm8903_hp_jack_gpio); } - snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, - &tegra_wm8903_mic_jack); - snd_soc_jack_add_pins(&tegra_wm8903_mic_jack, - ARRAY_SIZE(tegra_wm8903_mic_jack_pins), - tegra_wm8903_mic_jack_pins); + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_wm8903_mic_jack, + tegra_wm8903_mic_jack_pins, + ARRAY_SIZE(tegra_wm8903_mic_jack_pins)); wm8903_mic_detect(codec, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, 0); From 77c71765ef78f87dd901fcb4c751908e835a3b2e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 4 Mar 2015 10:33:45 +0100 Subject: [PATCH 129/162] ASoC: Remove snd_soc_jack_new() There are no users of snd_soc_jack_new() left and new users should use snd_soc_card_jack_new() instead. So remove the function. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 20 -------------------- 1 file changed, 20 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 99d9ce272209..40b3ee96f317 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1484,26 +1484,6 @@ static inline struct snd_soc_platform *snd_soc_kcontrol_platform( return snd_soc_component_to_platform(snd_soc_kcontrol_component(kcontrol)); } -/** - * snd_soc_jack_new - Create a new jack - * @codec: ASoC CODEC - * @id: an identifying string for this jack - * @type: a bitmask of enum snd_jack_type values that can be detected by - * this jack - * @jack: structure to use for the jack - * - * Creates a new jack object. - * - * Returns zero if successful, or a negative error code on failure. - * On success jack will be initialised. - */ -static inline int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, - int type, struct snd_soc_jack *jack) -{ - return snd_soc_card_jack_new(codec->component.card, id, type, jack, - NULL, 0); -} - int snd_soc_util_init(void); void snd_soc_util_exit(void); From 4c03a5ebc7f75e98b32591d1d2c6758c811dcbef Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:17 +0200 Subject: [PATCH 130/162] ASoC: davinci: Select SND_EDMA_SOC when SND_DAVINCI_SOC is enabled edma-pcm going to replace davinci-pcm as platform driver for daVinci platform. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 2b81ca418d2a..eae4e229f341 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,10 +1,11 @@ config SND_DAVINCI_SOC tristate "SoC Audio for TI DAVINCI" depends on ARCH_DAVINCI + select SND_EDMA_SOC config SND_EDMA_SOC tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)" - depends on SOC_AM33XX || SOC_AM43XX + depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. From 257ade78b6019cf1570c1239894a7a6a549768e1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:18 +0200 Subject: [PATCH 131/162] ASoC: davinci-i2s: Convert to use edma-pcm The edma-pcm can replace the old davinci-pcm as platform driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-i2s.c | 67 +++++++++++++-------------------- 1 file changed, 26 insertions(+), 41 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 15fb28fc8e1b..56cb4d95637d 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -23,8 +23,9 @@ #include #include #include +#include -#include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-i2s.h" @@ -122,7 +123,8 @@ static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { struct davinci_mcbsp_dev { struct device *dev; - struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; + int dma_request[2]; void __iomem *base; #define MOD_DSP_A 0 #define MOD_DSP_B 1 @@ -419,8 +421,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - struct davinci_pcm_dma_params *dma_params = - &dev->dma_params[substream->stream]; struct snd_interval *i = NULL; int mcbsp_word_length, master; unsigned int rcr, xcr, srgr, clk_div, freq, framesize; @@ -532,8 +532,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } - dma_params->acnt = dma_params->data_type = data_type[fmt]; - dma_params->fifo_level = 0; mcbsp_word_length = asp_word_length[fmt]; switch (master) { @@ -600,15 +598,6 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); - - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); - return 0; -} - static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -620,7 +609,6 @@ static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { - .startup = davinci_i2s_startup, .shutdown = davinci_i2s_shutdown, .prepare = davinci_i2s_prepare, .trigger = davinci_i2s_trigger, @@ -630,7 +618,18 @@ static const struct snd_soc_dai_ops davinci_i2s_dai_ops = { }; +static int davinci_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + return 0; +} + static struct snd_soc_dai_driver davinci_i2s_dai = { + .probe = davinci_i2s_dai_probe, .playback = { .channels_min = 2, .channels_max = 2, @@ -651,11 +650,9 @@ static const struct snd_soc_component_driver davinci_i2s_component = { static int davinci_i2s_probe(struct platform_device *pdev) { - struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea, *res; - enum dma_event_q asp_chan_q = EVENTQ_0; - enum dma_event_q ram_chan_q = EVENTQ_1; + int *dma; int ret; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -676,22 +673,6 @@ static int davinci_i2s_probe(struct platform_device *pdev) GFP_KERNEL); if (!dev) return -ENOMEM; - if (pdata) { - dev->enable_channel_combine = pdata->enable_channel_combine; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = - pdata->sram_size_playback; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = - pdata->sram_size_capture; - dev->clk_input_pin = pdata->clk_input_pin; - dev->i2s_accurate_sck = pdata->i2s_accurate_sck; - asp_chan_q = pdata->asp_chan_q; - ram_chan_q = pdata->ram_chan_q; - } - - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].asp_chan_q = asp_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].ram_chan_q = ram_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].asp_chan_q = asp_chan_q; - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].ram_chan_q = ram_chan_q; dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) @@ -705,10 +686,10 @@ static int davinci_i2s_probe(struct platform_device *pdev) goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); /* first TX, then RX */ @@ -718,7 +699,9 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start; + dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; + *dma = res->start; + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (!res) { @@ -726,9 +709,11 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENXIO; goto err_release_clk; } - dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; - dev->dev = &pdev->dev; + dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE]; + *dma = res->start; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma; + dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &davinci_i2s_component, @@ -736,7 +721,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (ret != 0) goto err_release_clk; - ret = davinci_soc_platform_register(&pdev->dev); + ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); goto err_unregister_component; From 62731d33c41d95914a0a796f319924e22e7ea411 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:19 +0200 Subject: [PATCH 132/162] ASoC: davinci-vcif: Convert to use edma-pcm The edma-pcm can replace the old davinci-pcm as platform driver. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-vcif.c | 55 ++++++++++++++------------------ 1 file changed, 24 insertions(+), 31 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 5bee04279ebe..fabd05f24aeb 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -33,8 +33,9 @@ #include #include #include +#include -#include "davinci-pcm.h" +#include "edma-pcm.h" #include "davinci-i2s.h" #define MOD_REG_BIT(val, mask, set) do { \ @@ -47,7 +48,8 @@ struct davinci_vcif_dev { struct davinci_vc *davinci_vc; - struct davinci_pcm_dma_params dma_params[2]; + struct snd_dmaengine_dai_dma_data dma_data[2]; + int dma_request[2]; }; static void davinci_vcif_start(struct snd_pcm_substream *substream) @@ -93,8 +95,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, { struct davinci_vcif_dev *davinci_vcif_dev = snd_soc_dai_get_drvdata(dai); struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc; - struct davinci_pcm_dma_params *dma_params = - &davinci_vcif_dev->dma_params[substream->stream]; u32 w; /* Restart the codec before setup */ @@ -113,16 +113,12 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, /* Determine xfer data type */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: - dma_params->data_type = 0; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_RD_UNSIGNED | DAVINCI_VC_CTRL_WD_BITS_8 | DAVINCI_VC_CTRL_WD_UNSIGNED, 1); break; case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_WD_BITS_8, 1); @@ -130,8 +126,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, DAVINCI_VC_CTRL_WD_UNSIGNED, 0); break; case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 | DAVINCI_VC_CTRL_RD_UNSIGNED | DAVINCI_VC_CTRL_WD_BITS_8 | @@ -142,8 +136,6 @@ static int davinci_vcif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - dma_params->acnt = dma_params->data_type; - writel(w, davinci_vc->base + DAVINCI_VC_CTRL); return 0; @@ -172,24 +164,25 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } -static int davinci_vcif_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); - - snd_soc_dai_set_dma_data(dai, substream, dev->dma_params); - return 0; -} - #define DAVINCI_VCIF_RATES SNDRV_PCM_RATE_8000_48000 static const struct snd_soc_dai_ops davinci_vcif_dai_ops = { - .startup = davinci_vcif_startup, .trigger = davinci_vcif_trigger, .hw_params = davinci_vcif_hw_params, }; +static int davinci_vcif_dai_probe(struct snd_soc_dai *dai) +{ + struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai); + + dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + + return 0; +} + static struct snd_soc_dai_driver davinci_vcif_dai = { + .probe = davinci_vcif_dai_probe, .playback = { .channels_min = 1, .channels_max = 2, @@ -225,16 +218,16 @@ static int davinci_vcif_probe(struct platform_device *pdev) /* DMA tx params */ davinci_vcif_dev->davinci_vc = davinci_vc; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = - davinci_vc->davinci_vcif.dma_tx_channel; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr = - davinci_vc->davinci_vcif.dma_tx_addr; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = + &davinci_vc->davinci_vcif.dma_tx_channel; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = + davinci_vc->davinci_vcif.dma_tx_addr; /* DMA rx params */ - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = - davinci_vc->davinci_vcif.dma_rx_channel; - davinci_vcif_dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr = - davinci_vc->davinci_vcif.dma_rx_addr; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = + &davinci_vc->davinci_vcif.dma_rx_channel; + davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = + davinci_vc->davinci_vcif.dma_rx_addr; dev_set_drvdata(&pdev->dev, davinci_vcif_dev); @@ -245,7 +238,7 @@ static int davinci_vcif_probe(struct platform_device *pdev) return ret; } - ret = davinci_soc_platform_register(&pdev->dev); + ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); snd_soc_unregister_component(&pdev->dev); From 9759e7ef53138c5ab46ea516ad08977eb5770393 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:20 +0200 Subject: [PATCH 133/162] ASoC: davinci-mcasp: Deprecate the use of davinci-pcm in favor of edma-pcm The edma-pcm performs as good as the old davinci-pcm and it's use does not require the 'ping-pong' mode of davinci-pcm, which was introduced to overcome under/over flow issues when using davinci-pcm. Keep the SND_DAVINCI_SOC config option to select the SND_EDMA_SOC to avoid regression in audio support. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Kconfig | 15 +++--- sound/soc/davinci/davinci-mcasp.c | 87 +++++++------------------------ 2 files changed, 27 insertions(+), 75 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index eae4e229f341..3736d9aabc56 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,15 +1,16 @@ config SND_DAVINCI_SOC - tristate "SoC Audio for TI DAVINCI" + tristate depends on ARCH_DAVINCI select SND_EDMA_SOC config SND_EDMA_SOC - tristate "SoC Audio for Texas Instruments chips using eDMA (AM33XX/43XX)" + tristate "SoC Audio for Texas Instruments chips using eDMA" depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. The following line of SoCs are supported by this platform driver: + - daVinci devices - AM335x - AM437x/AM438x @@ -18,7 +19,7 @@ config SND_DAVINCI_SOC_I2S config SND_DAVINCI_SOC_MCASP tristate "Multichannel Audio Serial Port (McASP) support" - depends on SND_DAVINCI_SOC || SND_OMAP_SOC || SND_EDMA_SOC + depends on SND_OMAP_SOC || SND_EDMA_SOC help Say Y or M here if you want to have support for McASP IP found in various Texas Instruments SoCs like: @@ -46,7 +47,7 @@ config SND_AM33XX_SOC_EVM config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" - depends on SND_DAVINCI_SOC && I2C + depends on SND_EDMA_SOC && I2C depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_GENERIC_EVM help @@ -74,7 +75,7 @@ endchoice config SND_DM6467_SOC_EVM tristate "SoC Audio support for DaVinci DM6467 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DM6467_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DM6467_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM select SND_SOC_SPDIF @@ -83,7 +84,7 @@ config SND_DM6467_SOC_EVM config SND_DA830_SOC_EVM tristate "SoC Audio support for DA830/OMAP-L137 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DA830_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM help @@ -92,7 +93,7 @@ config SND_DA830_SOC_EVM config SND_DA850_SOC_EVM tristate "SoC Audio support for DA850/OMAP-L138 EVM" - depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM && I2C + depends on SND_EDMA_SOC && MACH_DAVINCI_DA850_EVM && I2C select SND_DAVINCI_SOC_GENERIC_EVM help Say Y if you want to add support for SoC audio on TI diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 031c1fb44ae7..0c882995a357 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include @@ -36,7 +37,6 @@ #include #include -#include "davinci-pcm.h" #include "edma-pcm.h" #include "davinci-mcasp.h" @@ -65,7 +65,6 @@ struct davinci_mcasp_context { }; struct davinci_mcasp { - struct davinci_pcm_dma_params dma_params[2]; struct snd_dmaengine_dai_dma_data dma_data[2]; void __iomem *base; u32 fifo_base; @@ -82,6 +81,7 @@ struct davinci_mcasp { u16 bclk_lrclk_ratio; int streams; u32 irq_request[2]; + int dma_request[2]; int sysclk_freq; bool bclk_master; @@ -643,7 +643,6 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, int period_words, int channels) { - struct davinci_pcm_dma_params *dma_params = &mcasp->dma_params[stream]; struct snd_dmaengine_dai_dma_data *dma_data = &mcasp->dma_data[stream]; int i; u8 tx_ser = 0; @@ -711,10 +710,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, * For example if three serializers are enabled the DMA * need to transfer three words per DMA request. */ - dma_params->fifo_level = active_serializers; dma_data->maxburst = active_serializers; } else { - dma_params->fifo_level = 0; dma_data->maxburst = 0; } return 0; @@ -746,7 +743,6 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, /* Configure the burst size for platform drivers */ if (numevt == 1) numevt = 0; - dma_params->fifo_level = numevt; dma_data->maxburst = numevt; return 0; @@ -872,8 +868,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); - struct davinci_pcm_dma_params *dma_params = - &mcasp->dma_params[substream->stream]; int word_length; int channels = params_channels(params); int period_size = params_period_size(params); @@ -914,31 +908,26 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; word_length = 8; break; case SNDRV_PCM_FORMAT_U16_LE: case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; word_length = 16; break; case SNDRV_PCM_FORMAT_U24_3LE: case SNDRV_PCM_FORMAT_S24_3LE: - dma_params->data_type = 3; word_length = 24; break; case SNDRV_PCM_FORMAT_U24_LE: case SNDRV_PCM_FORMAT_S24_LE: - dma_params->data_type = 4; word_length = 24; break; case SNDRV_PCM_FORMAT_U32_LE: case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; word_length = 32; break; @@ -947,11 +936,6 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (mcasp->version == MCASP_VERSION_2 && !dma_params->fifo_level) - dma_params->acnt = 4; - else - dma_params->acnt = dma_params->data_type; - davinci_config_channel_size(mcasp, word_length); if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) @@ -1055,17 +1039,8 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - if (mcasp->version >= MCASP_VERSION_3) { - /* Using dmaengine PCM */ - dai->playback_dma_data = - &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dai->capture_dma_data = - &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - } else { - /* Using davinci-pcm */ - dai->playback_dma_data = mcasp->dma_params; - dai->capture_dma_data = mcasp->dma_params; - } + dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; return 0; } @@ -1184,28 +1159,24 @@ static const struct snd_soc_component_driver davinci_mcasp_component = { static struct davinci_mcasp_pdata dm646x_mcasp_pdata = { .tx_dma_offset = 0x400, .rx_dma_offset = 0x400, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_1, }; static struct davinci_mcasp_pdata da830_mcasp_pdata = { .tx_dma_offset = 0x2000, .rx_dma_offset = 0x2000, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_2, }; static struct davinci_mcasp_pdata am33xx_mcasp_pdata = { .tx_dma_offset = 0, .rx_dma_offset = 0, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_3, }; static struct davinci_mcasp_pdata dra7_mcasp_pdata = { .tx_dma_offset = 0x200, .rx_dma_offset = 0x284, - .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_4, }; @@ -1382,12 +1353,12 @@ nodata: static int davinci_mcasp_probe(struct platform_device *pdev) { - struct davinci_pcm_dma_params *dma_params; struct snd_dmaengine_dai_dma_data *dma_data; struct resource *mem, *ioarea, *res, *dat; struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; char *irq_name; + int *dma; int irq; int ret; @@ -1521,59 +1492,45 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) mcasp->dat_port = true; - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_playback; if (dat) - dma_params->dma_addr = dat->start; + dma_data->addr = dat->start; else - dma_params->dma_addr = mem->start + pdata->tx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + dma_data->addr = mem->start + pdata->tx_dma_offset; + dma = &mcasp->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) - dma_params->channel = res->start; + *dma = res->start; else - dma_params->channel = pdata->tx_dma_channel; + *dma = pdata->tx_dma_channel; /* dmaengine filter data for DT and non-DT boot */ if (pdev->dev.of_node) dma_data->filter_data = "tx"; else - dma_data->filter_data = &dma_params->channel; + dma_data->filter_data = dma; /* RX is not valid in DIT mode */ if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { - dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; - dma_params->asp_chan_q = pdata->asp_chan_q; - dma_params->ram_chan_q = pdata->ram_chan_q; - dma_params->sram_pool = pdata->sram_pool; - dma_params->sram_size = pdata->sram_size_capture; if (dat) - dma_params->dma_addr = dat->start; + dma_data->addr = dat->start; else - dma_params->dma_addr = mem->start + pdata->rx_dma_offset; - - /* Unconditional dmaengine stuff */ - dma_data->addr = dma_params->dma_addr; + dma_data->addr = mem->start + pdata->rx_dma_offset; + dma = &mcasp->dma_request[SNDRV_PCM_STREAM_CAPTURE]; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (res) - dma_params->channel = res->start; + *dma = res->start; else - dma_params->channel = pdata->rx_dma_channel; + *dma = pdata->rx_dma_channel; /* dmaengine filter data for DT and non-DT boot */ if (pdev->dev.of_node) dma_data->filter_data = "rx"; else - dma_data->filter_data = &dma_params->channel; + dma_data->filter_data = dma; } if (mcasp->version < MCASP_VERSION_3) { @@ -1596,17 +1553,11 @@ static int davinci_mcasp_probe(struct platform_device *pdev) goto err; switch (mcasp->version) { -#if IS_BUILTIN(CONFIG_SND_DAVINCI_SOC) || \ - (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ - IS_MODULE(CONFIG_SND_DAVINCI_SOC)) - case MCASP_VERSION_1: - case MCASP_VERSION_2: - ret = davinci_soc_platform_register(&pdev->dev); - break; -#endif #if IS_BUILTIN(CONFIG_SND_EDMA_SOC) || \ (IS_MODULE(CONFIG_SND_DAVINCI_SOC_MCASP) && \ IS_MODULE(CONFIG_SND_EDMA_SOC)) + case MCASP_VERSION_1: + case MCASP_VERSION_2: case MCASP_VERSION_3: ret = edma_pcm_platform_register(&pdev->dev); break; From 4da4608c91308d0d15dd022074724446c15710dc Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 3 Mar 2015 16:45:21 +0200 Subject: [PATCH 134/162] ASoC: davinci: Remove unused davinci-pcm platform driver All DAI drivers has been converted to use edma-pcm instead of davinci-pcm and the driver can be removed from the tree. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/davinci/Makefile | 2 - sound/soc/davinci/davinci-pcm.c | 861 -------------------------------- sound/soc/davinci/davinci-pcm.h | 41 -- 3 files changed, 904 deletions(-) delete mode 100644 sound/soc/davinci/davinci-pcm.c delete mode 100644 sound/soc/davinci/davinci-pcm.h diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 09bf2ba92d38..f883933c1a19 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -1,11 +1,9 @@ # DAVINCI Platform Support -snd-soc-davinci-objs := davinci-pcm.o snd-soc-edma-objs := edma-pcm.o snd-soc-davinci-i2s-objs := davinci-i2s.o snd-soc-davinci-mcasp-objs:= davinci-mcasp.o snd-soc-davinci-vcif-objs:= davinci-vcif.o -obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o obj-$(CONFIG_SND_EDMA_SOC) += snd-soc-edma.o obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o obj-$(CONFIG_SND_DAVINCI_SOC_MCASP) += snd-soc-davinci-mcasp.o diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c deleted file mode 100644 index 7809e9d935fc..000000000000 --- a/sound/soc/davinci/davinci-pcm.c +++ /dev/null @@ -1,861 +0,0 @@ -/* - * ALSA PCM interface for the TI DAVINCI processor - * - * Author: Vladimir Barinov, - * Copyright: (C) 2007 MontaVista Software, Inc., - * added SRAM ping/pong (C) 2008 Troy Kisky - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include -#include -#include -#include -#include -#include -#include -#include - -#include -#include -#include -#include - -#include - -#include "davinci-pcm.h" - -#ifdef DEBUG -static void print_buf_info(int slot, char *name) -{ - struct edmacc_param p; - if (slot < 0) - return; - edma_read_slot(slot, &p); - printk(KERN_DEBUG "%s: 0x%x, opt=%x, src=%x, a_b_cnt=%x dst=%x\n", - name, slot, p.opt, p.src, p.a_b_cnt, p.dst); - printk(KERN_DEBUG " src_dst_bidx=%x link_bcntrld=%x src_dst_cidx=%x ccnt=%x\n", - p.src_dst_bidx, p.link_bcntrld, p.src_dst_cidx, p.ccnt); -} -#else -static void print_buf_info(int slot, char *name) -{ -} -#endif - -static struct snd_pcm_hardware pcm_hardware_playback = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME| - SNDRV_PCM_INFO_BATCH), - .buffer_bytes_max = 128 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8 * 1024, - .periods_min = 16, - .periods_max = 255, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware pcm_hardware_capture = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_BATCH), - .buffer_bytes_max = 128 * 1024, - .period_bytes_min = 32, - .period_bytes_max = 8 * 1024, - .periods_min = 16, - .periods_max = 255, - .fifo_size = 0, -}; - -/* - * How ping/pong works.... - * - * Playback: - * ram_params - copys 2*ping_size from start of SDRAM to iram, - * links to ram_link2 - * ram_link2 - copys rest of SDRAM to iram in ping_size units, - * links to ram_link - * ram_link - copys entire SDRAM to iram in ping_size uints, - * links to self - * - * asp_params - same as asp_link[0] - * asp_link[0] - copys from lower half of iram to asp port - * links to asp_link[1], triggers iram copy event on completion - * asp_link[1] - copys from upper half of iram to asp port - * links to asp_link[0], triggers iram copy event on completion - * triggers interrupt only needed to let upper SOC levels update position - * in stream on completion - * - * When playback is started: - * ram_params started - * asp_params started - * - * Capture: - * ram_params - same as ram_link, - * links to ram_link - * ram_link - same as playback - * links to self - * - * asp_params - same as playback - * asp_link[0] - same as playback - * asp_link[1] - same as playback - * - * When capture is started: - * asp_params started - */ -struct davinci_runtime_data { - spinlock_t lock; - int period; /* current DMA period */ - int asp_channel; /* Master DMA channel */ - int asp_link[2]; /* asp parameter link channel, ping/pong */ - struct davinci_pcm_dma_params *params; /* DMA params */ - int ram_channel; - int ram_link; - int ram_link2; - struct edmacc_param asp_params; - struct edmacc_param ram_params; -}; - -static void davinci_pcm_period_elapsed(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - - prtd->period++; - if (unlikely(prtd->period >= runtime->periods)) - prtd->period = 0; -} - -static void davinci_pcm_period_reset(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - prtd->period = 0; -} -/* - * Not used with ping/pong - */ -static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int period_size; - unsigned int dma_offset; - dma_addr_t dma_pos; - dma_addr_t src, dst; - unsigned short src_bidx, dst_bidx; - unsigned short src_cidx, dst_cidx; - unsigned int data_type; - unsigned short acnt; - unsigned int count; - unsigned int fifo_level; - - period_size = snd_pcm_lib_period_bytes(substream); - dma_offset = prtd->period * period_size; - dma_pos = runtime->dma_addr + dma_offset; - fifo_level = prtd->params->fifo_level; - - pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos, - period_size); - - data_type = prtd->params->data_type; - count = period_size / data_type; - if (fifo_level) - count /= fifo_level; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = dma_pos; - dst = prtd->params->dma_addr; - src_bidx = data_type; - dst_bidx = 4; - src_cidx = data_type * fifo_level; - dst_cidx = 0; - } else { - src = prtd->params->dma_addr; - dst = dma_pos; - src_bidx = 0; - dst_bidx = data_type; - src_cidx = 0; - dst_cidx = data_type * fifo_level; - } - - acnt = prtd->params->acnt; - edma_set_src(prtd->asp_link[0], src, INCR, W8BIT); - edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT); - - edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx); - edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx); - - if (!fifo_level) - edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, - ASYNC); - else - edma_set_transfer_params(prtd->asp_link[0], acnt, - fifo_level, - count, fifo_level, - ABSYNC); -} - -static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) -{ - struct snd_pcm_substream *substream = data; - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - print_buf_info(prtd->ram_channel, "i ram_channel"); - pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); - - if (unlikely(ch_status != EDMA_DMA_COMPLETE)) - return; - - if (snd_pcm_running(substream)) { - spin_lock(&prtd->lock); - if (prtd->ram_channel < 0) { - /* No ping/pong must fix up link dma data*/ - davinci_pcm_enqueue_dma(substream); - } - davinci_pcm_period_elapsed(substream); - spin_unlock(&prtd->lock); - snd_pcm_period_elapsed(substream); - } -} - -#ifdef CONFIG_GENERIC_ALLOCATOR -static int allocate_sram(struct snd_pcm_substream *substream, - struct gen_pool *sram_pool, unsigned size, - struct snd_pcm_hardware *ppcm) -{ - struct snd_dma_buffer *buf = &substream->dma_buffer; - struct snd_dma_buffer *iram_dma = NULL; - dma_addr_t iram_phys = 0; - void *iram_virt = NULL; - - if (buf->private_data || !size) - return 0; - - ppcm->period_bytes_max = size; - iram_virt = gen_pool_dma_alloc(sram_pool, size, &iram_phys); - if (!iram_virt) - goto exit1; - iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); - if (!iram_dma) - goto exit2; - iram_dma->area = iram_virt; - iram_dma->addr = iram_phys; - memset(iram_dma->area, 0, size); - iram_dma->bytes = size; - buf->private_data = iram_dma; - return 0; -exit2: - if (iram_virt) - gen_pool_free(sram_pool, (unsigned)iram_virt, size); -exit1: - return -ENOMEM; -} - -static void davinci_free_sram(struct snd_pcm_substream *substream, - struct snd_dma_buffer *iram_dma) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct gen_pool *sram_pool = prtd->params->sram_pool; - - gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes); -} -#else -static int allocate_sram(struct snd_pcm_substream *substream, - struct gen_pool *sram_pool, unsigned size, - struct snd_pcm_hardware *ppcm) -{ - return 0; -} - -static void davinci_free_sram(struct snd_pcm_substream *substream, - struct snd_dma_buffer *iram_dma) -{ -} -#endif - -/* - * Only used with ping/pong. - * This is called after runtime->dma_addr, period_bytes and data_type are valid - */ -static int ping_pong_dma_setup(struct snd_pcm_substream *substream) -{ - unsigned short ram_src_cidx, ram_dst_cidx; - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - struct snd_dma_buffer *iram_dma = - (struct snd_dma_buffer *)substream->dma_buffer.private_data; - struct davinci_pcm_dma_params *params = prtd->params; - unsigned int data_type = params->data_type; - unsigned int acnt = params->acnt; - /* divide by 2 for ping/pong */ - unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; - unsigned int fifo_level = prtd->params->fifo_level; - unsigned int count; - if ((data_type == 0) || (data_type > 4)) { - printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type); - return -EINVAL; - } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dma_addr_t asp_src_pong = iram_dma->addr + ping_size; - ram_src_cidx = ping_size; - ram_dst_cidx = -ping_size; - edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT); - - edma_set_src_index(prtd->asp_link[0], data_type, - data_type * fifo_level); - edma_set_src_index(prtd->asp_link[1], data_type, - data_type * fifo_level); - - edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); - } else { - dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; - ram_src_cidx = -ping_size; - ram_dst_cidx = ping_size; - edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT); - - edma_set_dest_index(prtd->asp_link[0], data_type, - data_type * fifo_level); - edma_set_dest_index(prtd->asp_link[1], data_type, - data_type * fifo_level); - - edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); - } - - if (!fifo_level) { - count = ping_size / data_type; - edma_set_transfer_params(prtd->asp_link[0], acnt, count, - 1, 0, ASYNC); - edma_set_transfer_params(prtd->asp_link[1], acnt, count, - 1, 0, ASYNC); - } else { - count = ping_size / (data_type * fifo_level); - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, ABSYNC); - edma_set_transfer_params(prtd->asp_link[1], acnt, fifo_level, - count, fifo_level, ABSYNC); - } - - edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx); - edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx); - edma_set_transfer_params(prtd->ram_link, ping_size, 2, - runtime->periods, 2, ASYNC); - - /* init master params */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - edma_read_slot(prtd->ram_link, &prtd->ram_params); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - struct edmacc_param p_ram; - /* Copy entire iram buffer before playback started */ - prtd->ram_params.a_b_cnt = (1 << 16) | (ping_size << 1); - /* 0 dst_bidx */ - prtd->ram_params.src_dst_bidx = (ping_size << 1); - /* 0 dst_cidx */ - prtd->ram_params.src_dst_cidx = (ping_size << 1); - prtd->ram_params.ccnt = 1; - - /* Skip 1st period */ - edma_read_slot(prtd->ram_link, &p_ram); - p_ram.src += (ping_size << 1); - p_ram.ccnt -= 1; - edma_write_slot(prtd->ram_link2, &p_ram); - /* - * When 1st started, ram -> iram dma channel will fill the - * entire iram. Then, whenever a ping/pong asp buffer finishes, - * 1/2 iram will be filled. - */ - prtd->ram_params.link_bcntrld = - EDMA_CHAN_SLOT(prtd->ram_link2) << 5; - } - return 0; -} - -/* 1 asp tx or rx channel using 2 parameter channels - * 1 ram to/from iram channel using 1 parameter channel - * - * Playback - * ram copy channel kicks off first, - * 1st ram copy of entire iram buffer completion kicks off asp channel - * asp tcc always kicks off ram copy of 1/2 iram buffer - * - * Record - * asp channel starts, tcc kicks off ram copy - */ -static int request_ping_pong(struct snd_pcm_substream *substream, - struct davinci_runtime_data *prtd, - struct snd_dma_buffer *iram_dma) -{ - dma_addr_t asp_src_ping; - dma_addr_t asp_dst_ping; - int ret; - struct davinci_pcm_dma_params *params = prtd->params; - - /* Request ram master channel */ - ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, - davinci_pcm_dma_irq, substream, - prtd->params->ram_chan_q); - if (ret < 0) - goto exit1; - - /* Request ram link channel */ - ret = prtd->ram_link = edma_alloc_slot( - EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit2; - - ret = prtd->asp_link[1] = edma_alloc_slot( - EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit3; - - prtd->ram_link2 = -1; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = prtd->ram_link2 = edma_alloc_slot( - EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit4; - } - /* circle ping-pong buffers */ - edma_link(prtd->asp_link[0], prtd->asp_link[1]); - edma_link(prtd->asp_link[1], prtd->asp_link[0]); - /* circle ram buffers */ - edma_link(prtd->ram_link, prtd->ram_link); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - asp_src_ping = iram_dma->addr; - asp_dst_ping = params->dma_addr; /* fifo */ - } else { - asp_src_ping = params->dma_addr; /* fifo */ - asp_dst_ping = iram_dma->addr; - } - /* ping */ - edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT); - edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT); - edma_set_src_index(prtd->asp_link[0], 0, 0); - edma_set_dest_index(prtd->asp_link[0], 0, 0); - - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); - prtd->asp_params.opt |= TCCHEN | - EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(prtd->asp_link[0], &prtd->asp_params); - - /* pong */ - edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT); - edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT); - edma_set_src_index(prtd->asp_link[1], 0, 0); - edma_set_dest_index(prtd->asp_link[1], 0, 0); - - edma_read_slot(prtd->asp_link[1], &prtd->asp_params); - prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); - /* interrupt after every pong completion */ - prtd->asp_params.opt |= TCINTEN | TCCHEN | - EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(prtd->asp_link[1], &prtd->asp_params); - - /* ram */ - edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT); - edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT); - pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," - "for asp:%u %u %u\n", __func__, - prtd->ram_channel, prtd->ram_link, prtd->ram_link2, - prtd->asp_channel, prtd->asp_link[0], - prtd->asp_link[1]); - return 0; -exit4: - edma_free_channel(prtd->asp_link[1]); - prtd->asp_link[1] = -1; -exit3: - edma_free_channel(prtd->ram_link); - prtd->ram_link = -1; -exit2: - edma_free_channel(prtd->ram_channel); - prtd->ram_channel = -1; -exit1: - return ret; -} - -static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) -{ - struct snd_dma_buffer *iram_dma; - struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct davinci_pcm_dma_params *params = prtd->params; - int ret; - - if (!params) - return -ENODEV; - - /* Request asp master DMA channel */ - ret = prtd->asp_channel = edma_alloc_channel(params->channel, - davinci_pcm_dma_irq, substream, - prtd->params->asp_chan_q); - if (ret < 0) - goto exit1; - - /* Request asp link channels */ - ret = prtd->asp_link[0] = edma_alloc_slot( - EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (ret < 0) - goto exit2; - - iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; - if (iram_dma) { - if (request_ping_pong(substream, prtd, iram_dma) == 0) - return 0; - printk(KERN_WARNING "%s: dma channel allocation failed," - "not using sram\n", __func__); - } - - /* Issue transfer completion IRQ when the channel completes a - * transfer, then always reload from the same slot (by a kind - * of loopback link). The completion IRQ handler will update - * the reload slot with a new buffer. - * - * REVISIT save p_ram here after setting up everything except - * the buffer and its length (ccnt) ... use it as a template - * so davinci_pcm_enqueue_dma() takes less time in IRQ. - */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - prtd->asp_params.opt |= TCINTEN | - EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); - prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; - edma_write_slot(prtd->asp_link[0], &prtd->asp_params); - return 0; -exit2: - edma_free_channel(prtd->asp_channel); - prtd->asp_channel = -1; -exit1: - return ret; -} - -static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - int ret = 0; - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - edma_start(prtd->asp_channel); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - prtd->ram_channel >= 0) { - /* copy 1st iram buffer */ - edma_start(prtd->ram_channel); - } - break; - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - edma_resume(prtd->asp_channel); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - edma_pause(prtd->asp_channel); - break; - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static int davinci_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct davinci_runtime_data *prtd = substream->runtime->private_data; - - davinci_pcm_period_reset(substream); - if (prtd->ram_channel >= 0) { - int ret = ping_pong_dma_setup(substream); - if (ret < 0) - return ret; - - edma_write_slot(prtd->ram_channel, &prtd->ram_params); - edma_write_slot(prtd->asp_channel, &prtd->asp_params); - - print_buf_info(prtd->ram_channel, "ram_channel"); - print_buf_info(prtd->ram_link, "ram_link"); - print_buf_info(prtd->ram_link2, "ram_link2"); - print_buf_info(prtd->asp_channel, "asp_channel"); - print_buf_info(prtd->asp_link[0], "asp_link[0]"); - print_buf_info(prtd->asp_link[1], "asp_link[1]"); - - /* - * There is a phase offset of 2 periods between the position - * used by dma setup and the position reported in the pointer - * function. - * - * The phase offset, when not using ping-pong buffers, is due to - * the two consecutive calls to davinci_pcm_enqueue_dma() below. - * - * Whereas here, with ping-pong buffers, the phase is due to - * there being an entire buffer transfer complete before the - * first dma completion event triggers davinci_pcm_dma_irq(). - */ - davinci_pcm_period_elapsed(substream); - davinci_pcm_period_elapsed(substream); - - return 0; - } - davinci_pcm_enqueue_dma(substream); - davinci_pcm_period_elapsed(substream); - - /* Copy self-linked parameter RAM entry into master channel */ - edma_read_slot(prtd->asp_link[0], &prtd->asp_params); - edma_write_slot(prtd->asp_channel, &prtd->asp_params); - davinci_pcm_enqueue_dma(substream); - davinci_pcm_period_elapsed(substream); - - return 0; -} - -static snd_pcm_uframes_t -davinci_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - unsigned int offset; - int asp_count; - unsigned int period_size = snd_pcm_lib_period_bytes(substream); - - /* - * There is a phase offset of 2 periods between the position used by dma - * setup and the position reported in the pointer function. Either +2 in - * the dma setup or -2 here in the pointer function (with wrapping, - * both) accounts for this offset -- choose the latter since it makes - * the first-time setup clearer. - */ - spin_lock(&prtd->lock); - asp_count = prtd->period - 2; - spin_unlock(&prtd->lock); - - if (asp_count < 0) - asp_count += runtime->periods; - asp_count *= period_size; - - offset = bytes_to_frames(runtime, asp_count); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -static int davinci_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd; - struct snd_pcm_hardware *ppcm; - int ret = 0; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa; - struct davinci_pcm_dma_params *params; - - pa = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - if (!pa) - return -ENODEV; - params = &pa[substream->stream]; - - ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - &pcm_hardware_playback : &pcm_hardware_capture; - allocate_sram(substream, params->sram_pool, params->sram_size, ppcm); - snd_soc_set_runtime_hwparams(substream, ppcm); - /* ensure that buffer size is a multiple of period size */ - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - - prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - spin_lock_init(&prtd->lock); - prtd->params = params; - prtd->asp_channel = -1; - prtd->asp_link[0] = prtd->asp_link[1] = -1; - prtd->ram_channel = -1; - prtd->ram_link = -1; - prtd->ram_link2 = -1; - - runtime->private_data = prtd; - - ret = davinci_pcm_dma_request(substream); - if (ret) { - printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n"); - kfree(prtd); - } - - return ret; -} - -static int davinci_pcm_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct davinci_runtime_data *prtd = runtime->private_data; - - if (prtd->ram_channel >= 0) - edma_stop(prtd->ram_channel); - if (prtd->asp_channel >= 0) - edma_stop(prtd->asp_channel); - if (prtd->asp_link[0] >= 0) - edma_unlink(prtd->asp_link[0]); - if (prtd->asp_link[1] >= 0) - edma_unlink(prtd->asp_link[1]); - if (prtd->ram_link >= 0) - edma_unlink(prtd->ram_link); - - if (prtd->asp_link[0] >= 0) - edma_free_slot(prtd->asp_link[0]); - if (prtd->asp_link[1] >= 0) - edma_free_slot(prtd->asp_link[1]); - if (prtd->asp_channel >= 0) - edma_free_channel(prtd->asp_channel); - if (prtd->ram_link >= 0) - edma_free_slot(prtd->ram_link); - if (prtd->ram_link2 >= 0) - edma_free_slot(prtd->ram_link2); - if (prtd->ram_channel >= 0) - edma_free_channel(prtd->ram_channel); - - kfree(prtd); - - return 0; -} - -static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - return snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); -} - -static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -static int davinci_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops davinci_pcm_ops = { - .open = davinci_pcm_open, - .close = davinci_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = davinci_pcm_hw_params, - .hw_free = davinci_pcm_hw_free, - .prepare = davinci_pcm_prepare, - .trigger = davinci_pcm_trigger, - .pointer = davinci_pcm_pointer, - .mmap = davinci_pcm_mmap, -}; - -static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, - size_t size) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - - pr_debug("davinci_pcm: preallocate_dma_buffer: area=%p, addr=%p, " - "size=%d\n", (void *) buf->area, (void *) buf->addr, size); - - if (!buf->area) - return -ENOMEM; - - buf->bytes = size; - return 0; -} - -static void davinci_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - struct snd_dma_buffer *iram_dma; - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - iram_dma = buf->private_data; - if (iram_dma) { - davinci_free_sram(substream, iram_dma); - kfree(iram_dma); - } - } -} - -static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK, - pcm_hardware_playback.buffer_bytes_max); - if (ret) - return ret; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE, - pcm_hardware_capture.buffer_bytes_max); - if (ret) - return ret; - } - - return 0; -} - -static struct snd_soc_platform_driver davinci_soc_platform = { - .ops = &davinci_pcm_ops, - .pcm_new = davinci_pcm_new, - .pcm_free = davinci_pcm_free, -}; - -int davinci_soc_platform_register(struct device *dev) -{ - return devm_snd_soc_register_platform(dev, &davinci_soc_platform); -} -EXPORT_SYMBOL_GPL(davinci_soc_platform_register); - -MODULE_AUTHOR("Vladimir Barinov"); -MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h deleted file mode 100644 index 0fe2346a9aa2..000000000000 --- a/sound/soc/davinci/davinci-pcm.h +++ /dev/null @@ -1,41 +0,0 @@ -/* - * ALSA PCM interface for the TI DAVINCI processor - * - * Author: Vladimir Barinov, - * Copyright: (C) 2007 MontaVista Software, Inc., - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _DAVINCI_PCM_H -#define _DAVINCI_PCM_H - -#include -#include -#include - -struct davinci_pcm_dma_params { - int channel; /* sync dma channel ID */ - unsigned short acnt; - dma_addr_t dma_addr; /* device physical address for DMA */ - unsigned sram_size; - struct gen_pool *sram_pool; /* SRAM gen_pool for ping pong */ - enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ - enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ - unsigned char data_type; /* xfer data type */ - unsigned char convert_mono_stereo; - unsigned int fifo_level; -}; - -#if IS_ENABLED(CONFIG_SND_DAVINCI_SOC) -int davinci_soc_platform_register(struct device *dev); -#else -static inline int davinci_soc_platform_register(struct device *dev) -{ - return 0; -} -#endif /* CONFIG_SND_DAVINCI_SOC */ - -#endif From 6742e15cf92a8dc3065843a627952ed518e08267 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 3 Mar 2015 13:28:53 +0200 Subject: [PATCH 135/162] ASoC: omap-pcm: Allow only formats with 1, 2, and 4 byte physical size sDMA support only transfer elements with 1, 2, and 4 byte physical size. Initialize the pcm driver accordingly. Signed-off-by: Jyri Sarha Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index f4b05bc23e4b..e49ee2383a88 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -39,7 +39,7 @@ #define pcm_omap1510() 0 #endif -static const struct snd_pcm_hardware omap_pcm_hardware = { +static struct snd_pcm_hardware omap_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -53,6 +53,24 @@ static const struct snd_pcm_hardware omap_pcm_hardware = { .buffer_bytes_max = 128 * 1024, }; +/* sDMA supports only 1, 2, and 4 byte transfer elements. */ +static void omap_pcm_limit_supported_formats(void) +{ + int i; + + for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { + switch (snd_pcm_format_physical_width(i)) { + case 8: + case 16: + case 32: + omap_pcm_hardware.formats |= (1LL << i); + break; + default: + break; + } + } +} + /* this may get called several times by oss emulation */ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -235,6 +253,7 @@ static struct snd_soc_platform_driver omap_soc_platform = { int omap_pcm_platform_register(struct device *dev) { + omap_pcm_limit_supported_formats(); return devm_snd_soc_register_platform(dev, &omap_soc_platform); } EXPORT_SYMBOL_GPL(omap_pcm_platform_register); From 2bf9eba14340a53776a742f2c8a0bfbd9c86d259 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 3 Mar 2015 18:31:29 +0800 Subject: [PATCH 136/162] ASoC: rt5670: Fix the speaker mono output issue We need to set left/right control for the speaker amp to get stereo output on speaker. Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5670.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0632b7458a53..592f961b5de5 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2700,6 +2700,12 @@ static int rt5670_i2c_probe(struct i2c_client *i2c, regmap_write(rt5670->regmap, RT5670_RESET, 0); + regmap_read(rt5670->regmap, RT5670_VENDOR_ID, &val); + if (val >= 4) + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0980); + else + regmap_write(rt5670->regmap, RT5670_GPIO_CTRL3, 0x0d00); + ret = regmap_register_patch(rt5670->regmap, init_list, ARRAY_SIZE(init_list)); if (ret != 0) From bbed297d373471c8e4c3183bf67472a768576664 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Sun, 22 Feb 2015 16:43:21 +0000 Subject: [PATCH 137/162] ASoC: wm8804: Split out bus drivers Simplify dependencies by using new style split out bus interfaces. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 18 ++++- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wm8804-i2c.c | 64 ++++++++++++++++ sound/soc/codecs/wm8804-spi.c | 56 ++++++++++++++ sound/soc/codecs/wm8804.c | 139 ++++------------------------------ sound/soc/codecs/wm8804.h | 7 ++ 6 files changed, 162 insertions(+), 126 deletions(-) create mode 100644 sound/soc/codecs/wm8804-i2c.c create mode 100644 sound/soc/codecs/wm8804-spi.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 064e6c18e109..1d17988df796 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -141,7 +141,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8770 if SPI_MASTER select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8782 - select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8804_I2C if I2C + select SND_SOC_WM8804_SPI if SPI_MASTER select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C @@ -744,8 +745,19 @@ config SND_SOC_WM8782 tristate config SND_SOC_WM8804 - tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver" - depends on SND_SOC_I2C_AND_SPI + tristate + +config SND_SOC_WM8804_I2C + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver I2C" + depends on I2C + select SND_SOC_WM8804 + select REGMAP_I2C + +config SND_SOC_WM8804_SPI + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver SPI" + depends on SPI_MASTER + select SND_SOC_WM8804 + select REGMAP_SPI config SND_SOC_WM8900 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 69b8666d187a..7acb6c174cb4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -145,6 +145,8 @@ snd-soc-wm8770-objs := wm8770.o snd-soc-wm8776-objs := wm8776.o snd-soc-wm8782-objs := wm8782.o snd-soc-wm8804-objs := wm8804.o +snd-soc-wm8804-i2c-objs := wm8804-i2c.o +snd-soc-wm8804-spi-objs := wm8804-spi.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8904-objs := wm8904.o @@ -323,6 +325,8 @@ obj-$(CONFIG_SND_SOC_WM8770) += snd-soc-wm8770.o obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8782) += snd-soc-wm8782.o obj-$(CONFIG_SND_SOC_WM8804) += snd-soc-wm8804.o +obj-$(CONFIG_SND_SOC_WM8804_I2C) += snd-soc-wm8804-i2c.o +obj-$(CONFIG_SND_SOC_WM8804_SPI) += snd-soc-wm8804-spi.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8904) += snd-soc-wm8904.o diff --git a/sound/soc/codecs/wm8804-i2c.c b/sound/soc/codecs/wm8804-i2c.c new file mode 100644 index 000000000000..5bd4af2b4059 --- /dev/null +++ b/sound/soc/codecs/wm8804-i2c.c @@ -0,0 +1,64 @@ +/* + * wm8804-i2c.c -- WM8804 S/PDIF transceiver driver - I2C + * + * Copyright 2015 Cirrus Logic Inc + * + * Author: Charles Keepax + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include "wm8804.h" + +static int wm8804_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return wm8804_probe(&i2c->dev, regmap); +} + +static int wm8804_i2c_remove(struct i2c_client *i2c) +{ + wm8804_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id wm8804_i2c_id[] = { + { "wm8804", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); + +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + +static struct i2c_driver wm8804_i2c_driver = { + .driver = { + .name = "wm8804", + .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, + }, + .probe = wm8804_i2c_probe, + .remove = wm8804_i2c_remove, + .id_table = wm8804_i2c_id +}; + +module_i2c_driver(wm8804_i2c_driver); + +MODULE_DESCRIPTION("ASoC WM8804 driver - I2C"); +MODULE_AUTHOR("Charles Keepax "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8804-spi.c b/sound/soc/codecs/wm8804-spi.c new file mode 100644 index 000000000000..287e11e90794 --- /dev/null +++ b/sound/soc/codecs/wm8804-spi.c @@ -0,0 +1,56 @@ +/* + * wm8804-spi.c -- WM8804 S/PDIF transceiver driver - SPI + * + * Copyright 2015 Cirrus Logic Inc + * + * Author: Charles Keepax + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include + +#include "wm8804.h" + +static int wm8804_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return wm8804_probe(&spi->dev, regmap); +} + +static int wm8804_spi_remove(struct spi_device *spi) +{ + wm8804_remove(&spi->dev); + return 0; +} + +static const struct of_device_id wm8804_of_match[] = { + { .compatible = "wlf,wm8804", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8804_of_match); + +static struct spi_driver wm8804_spi_driver = { + .driver = { + .name = "wm8804", + .owner = THIS_MODULE, + .of_match_table = wm8804_of_match, + }, + .probe = wm8804_spi_probe, + .remove = wm8804_spi_remove +}; + +module_spi_driver(wm8804_spi_driver); + +MODULE_DESCRIPTION("ASoC WM8804 driver - SPI"); +MODULE_AUTHOR("Charles Keepax "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index b2b0e68f707e..b5a04fc5060f 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -15,10 +15,7 @@ #include #include #include -#include #include -#include -#include #include #include #include @@ -518,7 +515,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8804_remove(struct snd_soc_codec *codec) +static int wm8804_codec_remove(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; int i; @@ -531,7 +528,7 @@ static int wm8804_remove(struct snd_soc_codec *codec) return 0; } -static int wm8804_probe(struct snd_soc_codec *codec) +static int wm8804_codec_probe(struct snd_soc_codec *codec) { struct wm8804_priv *wm8804; int i, id1, id2, ret; @@ -649,8 +646,8 @@ static struct snd_soc_dai_driver wm8804_dai = { }; static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { - .probe = wm8804_probe, - .remove = wm8804_remove, + .probe = wm8804_codec_probe, + .remove = wm8804_codec_remove, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, @@ -658,13 +655,7 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .num_controls = ARRAY_SIZE(wm8804_snd_controls), }; -static const struct of_device_id wm8804_of_match[] = { - { .compatible = "wlf,wm8804", }, - { } -}; -MODULE_DEVICE_TABLE(of, wm8804_of_match); - -static const struct regmap_config wm8804_regmap_config = { +const struct regmap_config wm8804_regmap_config = { .reg_bits = 8, .val_bits = 8, @@ -675,128 +666,30 @@ static const struct regmap_config wm8804_regmap_config = { .reg_defaults = wm8804_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8804_reg_defaults), }; +EXPORT_SYMBOL_GPL(wm8804_regmap_config); -#if defined(CONFIG_SPI_MASTER) -static int wm8804_spi_probe(struct spi_device *spi) +int wm8804_probe(struct device *dev, struct regmap *regmap) { struct wm8804_priv *wm8804; - int ret; - wm8804 = devm_kzalloc(&spi->dev, sizeof *wm8804, GFP_KERNEL); + wm8804 = devm_kzalloc(dev, sizeof(*wm8804), GFP_KERNEL); if (!wm8804) return -ENOMEM; - wm8804->regmap = devm_regmap_init_spi(spi, &wm8804_regmap_config); - if (IS_ERR(wm8804->regmap)) { - ret = PTR_ERR(wm8804->regmap); - return ret; - } + dev_set_drvdata(dev, wm8804); - spi_set_drvdata(spi, wm8804); + wm8804->regmap = regmap; - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_wm8804, &wm8804_dai, 1); - - return ret; + return snd_soc_register_codec(dev, &soc_codec_dev_wm8804, + &wm8804_dai, 1); } +EXPORT_SYMBOL_GPL(wm8804_probe); -static int wm8804_spi_remove(struct spi_device *spi) +void wm8804_remove(struct device *dev) { - snd_soc_unregister_codec(&spi->dev); - return 0; + snd_soc_unregister_codec(dev); } - -static struct spi_driver wm8804_spi_driver = { - .driver = { - .name = "wm8804", - .owner = THIS_MODULE, - .of_match_table = wm8804_of_match, - }, - .probe = wm8804_spi_probe, - .remove = wm8804_spi_remove -}; -#endif - -#if IS_ENABLED(CONFIG_I2C) -static int wm8804_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct wm8804_priv *wm8804; - int ret; - - wm8804 = devm_kzalloc(&i2c->dev, sizeof *wm8804, GFP_KERNEL); - if (!wm8804) - return -ENOMEM; - - wm8804->regmap = devm_regmap_init_i2c(i2c, &wm8804_regmap_config); - if (IS_ERR(wm8804->regmap)) { - ret = PTR_ERR(wm8804->regmap); - return ret; - } - - i2c_set_clientdata(i2c, wm8804); - - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8804, &wm8804_dai, 1); - return ret; -} - -static int wm8804_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_codec(&i2c->dev); - return 0; -} - -static const struct i2c_device_id wm8804_i2c_id[] = { - { "wm8804", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, wm8804_i2c_id); - -static struct i2c_driver wm8804_i2c_driver = { - .driver = { - .name = "wm8804", - .owner = THIS_MODULE, - .of_match_table = wm8804_of_match, - }, - .probe = wm8804_i2c_probe, - .remove = wm8804_i2c_remove, - .id_table = wm8804_i2c_id -}; -#endif - -static int __init wm8804_modinit(void) -{ - int ret = 0; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&wm8804_i2c_driver); - if (ret) { - printk(KERN_ERR "Failed to register wm8804 I2C driver: %d\n", - ret); - } -#endif -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&wm8804_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8804 SPI driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(wm8804_modinit); - -static void __exit wm8804_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&wm8804_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&wm8804_spi_driver); -#endif -} -module_exit(wm8804_exit); +EXPORT_SYMBOL_GPL(wm8804_remove); MODULE_DESCRIPTION("ASoC WM8804 driver"); MODULE_AUTHOR("Dimitris Papastamos "); diff --git a/sound/soc/codecs/wm8804.h b/sound/soc/codecs/wm8804.h index e72d4f4ba6b1..a39a2563dc67 100644 --- a/sound/soc/codecs/wm8804.h +++ b/sound/soc/codecs/wm8804.h @@ -13,6 +13,8 @@ #ifndef _WM8804_H #define _WM8804_H +#include + /* * Register values. */ @@ -62,4 +64,9 @@ #define WM8804_MCLKDIV_256FS 0 #define WM8804_MCLKDIV_128FS 1 +extern const struct regmap_config wm8804_regmap_config; + +int wm8804_probe(struct device *dev, struct regmap *regmap); +void wm8804_remove(struct device *dev); + #endif /* _WM8804_H */ From 6f2c9348095ae1a489abafe2ab3db7deca406e49 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Sun, 22 Feb 2015 16:43:22 +0000 Subject: [PATCH 138/162] ASoC: wm8804: Merge CODEC probe and bus probe All of the things in the CODEC probe, such as getting the regulators and verifying the chip ID, are better done in bus probe. It is better to fail during bus probe if this is the wrong chip and all resource allocation should be done in the bus probe anyway. This patch merges the CODEC probe into bus probe. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 180 +++++++++++++++++--------------------- 1 file changed, 82 insertions(+), 98 deletions(-) diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index b5a04fc5060f..1bd4ace29594 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -182,9 +182,9 @@ static bool wm8804_volatile(struct device *dev, unsigned int reg) } } -static int wm8804_reset(struct snd_soc_codec *codec) +static int wm8804_reset(struct wm8804_priv *wm8804) { - return snd_soc_write(codec, WM8804_RST_DEVID1, 0x0); + return regmap_write(wm8804->regmap, WM8804_RST_DEVID1, 0x0); } static int wm8804_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) @@ -515,100 +515,6 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, return 0; } -static int wm8804_codec_remove(struct snd_soc_codec *codec) -{ - struct wm8804_priv *wm8804; - int i; - - wm8804 = snd_soc_codec_get_drvdata(codec); - - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) - regulator_unregister_notifier(wm8804->supplies[i].consumer, - &wm8804->disable_nb[i]); - return 0; -} - -static int wm8804_codec_probe(struct snd_soc_codec *codec) -{ - struct wm8804_priv *wm8804; - int i, id1, id2, ret; - - wm8804 = snd_soc_codec_get_drvdata(codec); - - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) - wm8804->supplies[i].supply = wm8804_supply_names[i]; - - ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8804->supplies), - wm8804->supplies); - if (ret) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - - wm8804->disable_nb[0].notifier_call = wm8804_regulator_event_0; - wm8804->disable_nb[1].notifier_call = wm8804_regulator_event_1; - - /* This should really be moved into the regulator core */ - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) { - ret = regulator_register_notifier(wm8804->supplies[i].consumer, - &wm8804->disable_nb[i]); - if (ret != 0) { - dev_err(codec->dev, - "Failed to register regulator notifier: %d\n", - ret); - } - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies), - wm8804->supplies); - if (ret) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - return ret; - } - - id1 = snd_soc_read(codec, WM8804_RST_DEVID1); - if (id1 < 0) { - dev_err(codec->dev, "Failed to read device ID: %d\n", id1); - ret = id1; - goto err_reg_enable; - } - - id2 = snd_soc_read(codec, WM8804_DEVID2); - if (id2 < 0) { - dev_err(codec->dev, "Failed to read device ID: %d\n", id2); - ret = id2; - goto err_reg_enable; - } - - id2 = (id2 << 8) | id1; - - if (id2 != 0x8805) { - dev_err(codec->dev, "Invalid device ID: %#x\n", id2); - ret = -EINVAL; - goto err_reg_enable; - } - - ret = snd_soc_read(codec, WM8804_DEVREV); - if (ret < 0) { - dev_err(codec->dev, "Failed to read device revision: %d\n", - ret); - goto err_reg_enable; - } - dev_info(codec->dev, "revision %c\n", ret + 'A'); - - ret = wm8804_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset: %d\n", ret); - goto err_reg_enable; - } - - return 0; - -err_reg_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); - return ret; -} - static const struct snd_soc_dai_ops wm8804_dai_ops = { .hw_params = wm8804_hw_params, .set_fmt = wm8804_set_fmt, @@ -646,8 +552,6 @@ static struct snd_soc_dai_driver wm8804_dai = { }; static const struct snd_soc_codec_driver soc_codec_dev_wm8804 = { - .probe = wm8804_codec_probe, - .remove = wm8804_codec_remove, .set_bias_level = wm8804_set_bias_level, .idle_bias_off = true, @@ -671,6 +575,8 @@ EXPORT_SYMBOL_GPL(wm8804_regmap_config); int wm8804_probe(struct device *dev, struct regmap *regmap) { struct wm8804_priv *wm8804; + unsigned int id1, id2; + int i, ret; wm8804 = devm_kzalloc(dev, sizeof(*wm8804), GFP_KERNEL); if (!wm8804) @@ -680,13 +586,91 @@ int wm8804_probe(struct device *dev, struct regmap *regmap) wm8804->regmap = regmap; + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) + wm8804->supplies[i].supply = wm8804_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(wm8804->supplies), + wm8804->supplies); + if (ret) { + dev_err(dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + wm8804->disable_nb[0].notifier_call = wm8804_regulator_event_0; + wm8804->disable_nb[1].notifier_call = wm8804_regulator_event_1; + + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) { + ret = regulator_register_notifier(wm8804->supplies[i].consumer, + &wm8804->disable_nb[i]); + if (ret != 0) { + dev_err(dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8804->supplies), + wm8804->supplies); + if (ret) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + goto err_reg_enable; + } + + ret = regmap_read(regmap, WM8804_RST_DEVID1, &id1); + if (ret < 0) { + dev_err(dev, "Failed to read device ID: %d\n", ret); + goto err_reg_enable; + } + + ret = regmap_read(regmap, WM8804_DEVID2, &id2); + if (ret < 0) { + dev_err(dev, "Failed to read device ID: %d\n", ret); + goto err_reg_enable; + } + + id2 = (id2 << 8) | id1; + + if (id2 != 0x8805) { + dev_err(dev, "Invalid device ID: %#x\n", id2); + ret = -EINVAL; + goto err_reg_enable; + } + + ret = regmap_read(regmap, WM8804_DEVREV, &id1); + if (ret < 0) { + dev_err(dev, "Failed to read device revision: %d\n", + ret); + goto err_reg_enable; + } + dev_info(dev, "revision %c\n", id1 + 'A'); + + ret = wm8804_reset(wm8804); + if (ret < 0) { + dev_err(dev, "Failed to issue reset: %d\n", ret); + goto err_reg_enable; + } + return snd_soc_register_codec(dev, &soc_codec_dev_wm8804, &wm8804_dai, 1); + +err_reg_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8804->supplies), wm8804->supplies); + return ret; } EXPORT_SYMBOL_GPL(wm8804_probe); void wm8804_remove(struct device *dev) { + struct wm8804_priv *wm8804; + int i; + + wm8804 = dev_get_drvdata(dev); + + for (i = 0; i < ARRAY_SIZE(wm8804->supplies); ++i) + regulator_unregister_notifier(wm8804->supplies[i].consumer, + &wm8804->disable_nb[i]); + snd_soc_unregister_codec(dev); } EXPORT_SYMBOL_GPL(wm8804_remove); From d6482288aadcf19e348cbccff7a605790a3b3875 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" Date: Fri, 6 Mar 2015 16:59:00 +0100 Subject: [PATCH 139/162] ALSA: ac97: Add VT1613 AC97 codec support Patch to add an VT1613 AC97 codec support. This codec has additional DC offset removal control, headphone output and no video input. Signed-off-by: Maciej Szmigiero Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 1 + sound/pci/ac97/ac97_patch.c | 33 +++++++++++++++++++++++++++++++++ 2 files changed, 34 insertions(+) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 5ee2f17c287c..5bca1a33fed6 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -177,6 +177,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] { 0x54584e03, 0xffffffff, "TLV320AIC27", NULL, NULL }, { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, +{ 0x56494120, 0xfffffff0, "VIA1613", patch_vt1613, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF { 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL }, diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index ceaac1c41906..eca22100f70d 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3351,6 +3351,39 @@ static int patch_cm9780(struct snd_ac97 *ac97) return 0; } +/* + * VIA VT1613 codec + */ +static const struct snd_kcontrol_new snd_ac97_controls_vt1613[] = { +AC97_SINGLE("DC Offset removal", 0x5a, 10, 1, 0), +}; + +static int patch_vt1613_specific(struct snd_ac97 *ac97) +{ + int err; + + err = patch_build_controls(ac97, &snd_ac97_controls_vt1613[0], + ARRAY_SIZE(snd_ac97_controls_vt1613)); + if (err) + return err; + + return 0; +}; + +static const struct snd_ac97_build_ops patch_vt1613_ops = { + .build_specific = patch_vt1613_specific +}; + +static int patch_vt1613(struct snd_ac97 *ac97) +{ + ac97->build_ops = &patch_vt1613_ops; + + ac97->flags |= AC97_HAS_NO_VIDEO; + ac97->caps |= AC97_BC_HEADPHONE; + + return 0; +} + /* * VIA VT1616 codec */ From 5371fc0ecdf55b6811ade8a198de8ace2f4e5861 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 6 Mar 2015 13:41:42 -0300 Subject: [PATCH 140/162] ALSA: ac97: ac97_patch: Simplify patch_vt1613_specific() We can simplify the code by returning patch_build_controls() directly. Signed-off-by: Fabio Estevam Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index eca22100f70d..f4234edb878c 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -3360,14 +3360,8 @@ AC97_SINGLE("DC Offset removal", 0x5a, 10, 1, 0), static int patch_vt1613_specific(struct snd_ac97 *ac97) { - int err; - - err = patch_build_controls(ac97, &snd_ac97_controls_vt1613[0], - ARRAY_SIZE(snd_ac97_controls_vt1613)); - if (err) - return err; - - return 0; + return patch_build_controls(ac97, &snd_ac97_controls_vt1613[0], + ARRAY_SIZE(snd_ac97_controls_vt1613)); }; static const struct snd_ac97_build_ops patch_vt1613_ops = { From ce991981311e0ae258982b600564226ad6cb24ea Mon Sep 17 00:00:00 2001 From: Yannick Guerrini Date: Mon, 9 Mar 2015 22:13:03 +0100 Subject: [PATCH 141/162] ALSA: firewire: Fix trivial typos in comments Change 'propper' to 'proper' Change 'paramters' to 'parameters' Change 'SYT_INTEVAL' to 'SYT_INTERVAL' Change 'aligh'/'alighed' to 'align'/'aligned' Signed-off-by: Yannick Guerrini Signed-off-by: Takashi Iwai --- sound/firewire/amdtp.c | 8 ++++---- sound/firewire/fireworks/fireworks_transaction.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 5cc356db5351..e061355f535f 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -166,10 +166,10 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, * One AMDTP packet can include some frames. In blocking mode, the * number equals to SYT_INTERVAL. So the number is 8, 16 or 32, * depending on its sampling rate. For accurate period interrupt, it's - * preferrable to aligh period/buffer sizes to current SYT_INTERVAL. + * preferrable to align period/buffer sizes to current SYT_INTERVAL. * - * TODO: These constraints can be improved with propper rules. - * Currently apply LCM of SYT_INTEVALs. + * TODO: These constraints can be improved with proper rules. + * Currently apply LCM of SYT_INTERVALs. */ err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 32); @@ -270,7 +270,7 @@ static void amdtp_read_s32(struct amdtp_stream *s, * @s: the AMDTP stream to configure * @format: the format of the ALSA PCM device * - * The sample format must be set after the other paramters (rate/PCM channels/ + * The sample format must be set after the other parameters (rate/PCM channels/ * MIDI) and before the stream is started, and must not be changed while the * stream is running. */ diff --git a/sound/firewire/fireworks/fireworks_transaction.c b/sound/firewire/fireworks/fireworks_transaction.c index 2a85e4209f0b..f550808d1784 100644 --- a/sound/firewire/fireworks/fireworks_transaction.c +++ b/sound/firewire/fireworks/fireworks_transaction.c @@ -13,7 +13,7 @@ * * Transaction substance: * At first, 6 data exist. Following to the data, parameters for each command - * exist. All of the parameters are 32 bit alighed to big endian. + * exist. All of the parameters are 32 bit aligned to big endian. * data[0]: Length of transaction substance * data[1]: Transaction version * data[2]: Sequence number. This is incremented by the device From 4ed56666b7fc98c750a23b5263350b75e742b534 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 10 Mar 2015 22:13:30 +0900 Subject: [PATCH 142/162] ALSA: core: use precomputed table to check userspace control params The parameters can be decided in compile time. This commit adds precomputed table to reduce calculating time. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/control.c | 60 +++++++++++++++++++++----------------------- 1 file changed, 28 insertions(+), 32 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 35324a8e83c8..0b85cbc27e4d 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1161,6 +1161,23 @@ static void snd_ctl_elem_user_free(struct snd_kcontrol *kcontrol) static int snd_ctl_elem_add(struct snd_ctl_file *file, struct snd_ctl_elem_info *info, int replace) { + /* The capacity of struct snd_ctl_elem_value.value.*/ + static const unsigned int value_sizes[] = { + [SNDRV_CTL_ELEM_TYPE_BOOLEAN] = sizeof(long), + [SNDRV_CTL_ELEM_TYPE_INTEGER] = sizeof(long), + [SNDRV_CTL_ELEM_TYPE_ENUMERATED] = sizeof(unsigned int), + [SNDRV_CTL_ELEM_TYPE_BYTES] = sizeof(unsigned char), + [SNDRV_CTL_ELEM_TYPE_IEC958] = sizeof(struct snd_aes_iec958), + [SNDRV_CTL_ELEM_TYPE_INTEGER64] = sizeof(long long), + }; + static const unsigned int max_value_counts[] = { + [SNDRV_CTL_ELEM_TYPE_BOOLEAN] = 128, + [SNDRV_CTL_ELEM_TYPE_INTEGER] = 128, + [SNDRV_CTL_ELEM_TYPE_ENUMERATED] = 128, + [SNDRV_CTL_ELEM_TYPE_BYTES] = 512, + [SNDRV_CTL_ELEM_TYPE_IEC958] = 1, + [SNDRV_CTL_ELEM_TYPE_INTEGER64] = 64, + }; struct snd_card *card = file->card; struct snd_kcontrol kctl, *_kctl; unsigned int access; @@ -1168,8 +1185,6 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, struct user_element *ue; int idx, err; - if (info->count < 1) - return -EINVAL; access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| @@ -1201,37 +1216,18 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, kctl.tlv.c = snd_ctl_elem_user_tlv; access |= SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; } - switch (info->type) { - case SNDRV_CTL_ELEM_TYPE_BOOLEAN: - case SNDRV_CTL_ELEM_TYPE_INTEGER: - private_size = sizeof(long); - if (info->count > 128) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_INTEGER64: - private_size = sizeof(long long); - if (info->count > 64) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_ENUMERATED: - private_size = sizeof(unsigned int); - if (info->count > 128 || info->value.enumerated.items == 0) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_BYTES: - private_size = sizeof(unsigned char); - if (info->count > 512) - return -EINVAL; - break; - case SNDRV_CTL_ELEM_TYPE_IEC958: - private_size = sizeof(struct snd_aes_iec958); - if (info->count != 1) - return -EINVAL; - break; - default: + + if (info->type < SNDRV_CTL_ELEM_TYPE_BOOLEAN || + info->type > SNDRV_CTL_ELEM_TYPE_INTEGER64) return -EINVAL; - } - private_size *= info->count; + if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED && + info->value.enumerated.items == 0) + return -EINVAL; + if (info->count < 1 || + info->count > max_value_counts[info->type]) + return -EINVAL; + + private_size = value_sizes[info->type] * info->count; ue = kzalloc(sizeof(struct user_element) + private_size, GFP_KERNEL); if (ue == NULL) return -ENOMEM; From 2225e79b9b0370bc179f44756bee809b5e7b4d06 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Tue, 10 Mar 2015 22:13:31 +0900 Subject: [PATCH 143/162] ALSA: core: reduce stack usage related to snd_ctl_new() The callers of snd_ctl_new() need to have 'struct snd_kcontrol' data, and pass the data as template. Then, the function allocates the structure data again and copy from the template. This is a waste of resources. Especially, the callers use large stack for the template. This commit removes a need of template for the function, thus, changes the prototype of snd_ctl_new(). Furthermore, this commit changes the code of callers, snd_ctl_new1() and snd_ctl_elem_add() for better shape. Signed-off-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/control.c | 215 ++++++++++++++++++++++++++----------------- 1 file changed, 131 insertions(+), 84 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 0b85cbc27e4d..e1d8e0c816f0 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -192,36 +192,43 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, EXPORT_SYMBOL(snd_ctl_notify); /** - * snd_ctl_new - create a control instance from the template - * @control: the control template - * @access: the default control access + * snd_ctl_new - create a new control instance with some elements + * @kctl: the pointer to store new control instance + * @count: the number of elements in this control + * @access: the default access flags for elements in this control + * @file: given when locking these elements * - * Allocates a new struct snd_kcontrol instance and copies the given template - * to the new instance. It does not copy volatile data (access). + * Allocates a memory object for a new control instance. The instance has + * elements as many as the given number (@count). Each element has given + * access permissions (@access). Each element is locked when @file is given. * - * Return: The pointer of the new instance, or %NULL on failure. + * Return: 0 on success, error code on failure */ -static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, - unsigned int access) +static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, + unsigned int access, struct snd_ctl_file *file) { - struct snd_kcontrol *kctl; + unsigned int size; unsigned int idx; - if (snd_BUG_ON(!control || !control->count)) - return NULL; + if (count == 0 || count > MAX_CONTROL_COUNT) + return -EINVAL; - if (control->count > MAX_CONTROL_COUNT) - return NULL; + size = sizeof(struct snd_kcontrol); + size += sizeof(struct snd_kcontrol_volatile) * count; - kctl = kzalloc(sizeof(*kctl) + sizeof(struct snd_kcontrol_volatile) * control->count, GFP_KERNEL); - if (kctl == NULL) { + *kctl = kzalloc(size, GFP_KERNEL); + if (*kctl == NULL) { pr_err("ALSA: Cannot allocate control instance\n"); - return NULL; + return -ENOMEM; } - *kctl = *control; - for (idx = 0; idx < kctl->count; idx++) - kctl->vd[idx].access = access; - return kctl; + + for (idx = 0; idx < count; idx++) { + (*kctl)->vd[idx].access = access; + (*kctl)->vd[idx].owner = file; + } + (*kctl)->count = count; + + return 0; } /** @@ -238,37 +245,53 @@ static struct snd_kcontrol *snd_ctl_new(struct snd_kcontrol *control, struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, void *private_data) { - struct snd_kcontrol kctl; + struct snd_kcontrol *kctl; + unsigned int count; unsigned int access; + int err; if (snd_BUG_ON(!ncontrol || !ncontrol->info)) return NULL; - memset(&kctl, 0, sizeof(kctl)); - kctl.id.iface = ncontrol->iface; - kctl.id.device = ncontrol->device; - kctl.id.subdevice = ncontrol->subdevice; + + count = ncontrol->count; + if (count == 0) + count = 1; + + access = ncontrol->access; + if (access == 0) + access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + access &= (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE | + SNDRV_CTL_ELEM_ACCESS_INACTIVE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND | + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK); + + err = snd_ctl_new(&kctl, count, access, NULL); + if (err < 0) + return NULL; + + /* The 'numid' member is decided when calling snd_ctl_add(). */ + kctl->id.iface = ncontrol->iface; + kctl->id.device = ncontrol->device; + kctl->id.subdevice = ncontrol->subdevice; if (ncontrol->name) { - strlcpy(kctl.id.name, ncontrol->name, sizeof(kctl.id.name)); - if (strcmp(ncontrol->name, kctl.id.name) != 0) + strlcpy(kctl->id.name, ncontrol->name, sizeof(kctl->id.name)); + if (strcmp(ncontrol->name, kctl->id.name) != 0) pr_warn("ALSA: Control name '%s' truncated to '%s'\n", - ncontrol->name, kctl.id.name); + ncontrol->name, kctl->id.name); } - kctl.id.index = ncontrol->index; - kctl.count = ncontrol->count ? ncontrol->count : 1; - access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : - (ncontrol->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| - SNDRV_CTL_ELEM_ACCESS_VOLATILE| - SNDRV_CTL_ELEM_ACCESS_INACTIVE| - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| - SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND| - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); - kctl.info = ncontrol->info; - kctl.get = ncontrol->get; - kctl.put = ncontrol->put; - kctl.tlv.p = ncontrol->tlv.p; - kctl.private_value = ncontrol->private_value; - kctl.private_data = private_data; - return snd_ctl_new(&kctl, access); + kctl->id.index = ncontrol->index; + + kctl->info = ncontrol->info; + kctl->get = ncontrol->get; + kctl->put = ncontrol->put; + kctl->tlv.p = ncontrol->tlv.p; + + kctl->private_value = ncontrol->private_value; + kctl->private_data = private_data; + + return kctl; } EXPORT_SYMBOL(snd_ctl_new1); @@ -1179,44 +1202,48 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, [SNDRV_CTL_ELEM_TYPE_INTEGER64] = 64, }; struct snd_card *card = file->card; - struct snd_kcontrol kctl, *_kctl; + struct snd_kcontrol *kctl; + unsigned int count; unsigned int access; long private_size; struct user_element *ue; - int idx, err; - - access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : - (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| - SNDRV_CTL_ELEM_ACCESS_INACTIVE| - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE)); - info->id.numid = 0; - memset(&kctl, 0, sizeof(kctl)); + int err; + /* Delete a control to replace them if needed. */ if (replace) { + info->id.numid = 0; err = snd_ctl_remove_user_ctl(file, &info->id); if (err) return err; } - if (card->user_ctl_count >= MAX_USER_CONTROLS) + /* + * The number of userspace controls are counted control by control, + * not element by element. + */ + if (card->user_ctl_count + 1 > MAX_USER_CONTROLS) return -ENOMEM; - memcpy(&kctl.id, &info->id, sizeof(info->id)); - kctl.count = info->owner ? info->owner : 1; - access |= SNDRV_CTL_ELEM_ACCESS_USER; - if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) - kctl.info = snd_ctl_elem_user_enum_info; - else - kctl.info = snd_ctl_elem_user_info; - if (access & SNDRV_CTL_ELEM_ACCESS_READ) - kctl.get = snd_ctl_elem_user_get; - if (access & SNDRV_CTL_ELEM_ACCESS_WRITE) - kctl.put = snd_ctl_elem_user_put; - if (access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE) { - kctl.tlv.c = snd_ctl_elem_user_tlv; - access |= SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; - } + /* Check the number of elements for this userspace control. */ + count = info->owner; + if (count == 0) + count = 1; + /* Arrange access permissions if needed. */ + access = info->access; + if (access == 0) + access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + access &= (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_INACTIVE | + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE); + if (access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE) + access |= SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + access |= SNDRV_CTL_ELEM_ACCESS_USER; + + /* + * Check information and calculate the size of data specific to + * this userspace control. + */ if (info->type < SNDRV_CTL_ELEM_TYPE_BOOLEAN || info->type > SNDRV_CTL_ELEM_TYPE_INTEGER64) return -EINVAL; @@ -1226,11 +1253,27 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1 || info->count > max_value_counts[info->type]) return -EINVAL; - private_size = value_sizes[info->type] * info->count; - ue = kzalloc(sizeof(struct user_element) + private_size, GFP_KERNEL); - if (ue == NULL) + + /* + * Keep memory object for this userspace control. After passing this + * code block, the instance should be freed by snd_ctl_free_one(). + * + * Note that these elements in this control are locked. + */ + err = snd_ctl_new(&kctl, count, access, file); + if (err < 0) + return err; + kctl->private_data = kzalloc(sizeof(struct user_element) + private_size, + GFP_KERNEL); + if (kctl->private_data == NULL) { + kfree(kctl); return -ENOMEM; + } + kctl->private_free = snd_ctl_elem_user_free; + + /* Set private data for this userspace control. */ + ue = (struct user_element *)kctl->private_data; ue->card = card; ue->info = *info; ue->info.access = 0; @@ -1239,21 +1282,25 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (ue->info.type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) { err = snd_ctl_elem_init_enum_names(ue); if (err < 0) { - kfree(ue); + snd_ctl_free_one(kctl); return err; } } - kctl.private_free = snd_ctl_elem_user_free; - _kctl = snd_ctl_new(&kctl, access); - if (_kctl == NULL) { - kfree(ue->priv_data); - kfree(ue); - return -ENOMEM; - } - _kctl->private_data = ue; - for (idx = 0; idx < _kctl->count; idx++) - _kctl->vd[idx].owner = file; - err = snd_ctl_add(card, _kctl); + + /* Set callback functions. */ + if (info->type == SNDRV_CTL_ELEM_TYPE_ENUMERATED) + kctl->info = snd_ctl_elem_user_enum_info; + else + kctl->info = snd_ctl_elem_user_info; + if (access & SNDRV_CTL_ELEM_ACCESS_READ) + kctl->get = snd_ctl_elem_user_get; + if (access & SNDRV_CTL_ELEM_ACCESS_WRITE) + kctl->put = snd_ctl_elem_user_put; + if (access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE) + kctl->tlv.c = snd_ctl_elem_user_tlv; + + /* This function manage to free the instance on failure. */ + err = snd_ctl_add(card, kctl); if (err < 0) return err; From 8d98a0673f761f9b7be51a293ca9142ec0c037ca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 15:39:55 +0100 Subject: [PATCH 144/162] ALSA: seq_oss: Drop superfluous error/debug messages after malloc failures The kernel memory allocators already report the errors when the requested allocation fails, thus we don't need to warn it again in each caller side. Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_init.c | 4 +--- sound/core/seq/oss/seq_oss_midi.c | 5 ++--- sound/core/seq/oss/seq_oss_readq.c | 9 ++++----- sound/core/seq/oss/seq_oss_synth.c | 6 ++---- 4 files changed, 9 insertions(+), 15 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index b0e32e161dd1..2de3feff70d0 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -188,10 +188,8 @@ snd_seq_oss_open(struct file *file, int level) struct seq_oss_devinfo *dp; dp = kzalloc(sizeof(*dp), GFP_KERNEL); - if (!dp) { - pr_err("ALSA: seq_oss: can't malloc device info\n"); + if (!dp) return -ENOMEM; - } dp->cseq = system_client; dp->port = -1; diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index e79cc44b1394..96e8395ae586 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -173,10 +173,9 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) /* * allocate midi info record */ - if ((mdev = kzalloc(sizeof(*mdev), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc midi info\n"); + mdev = kzalloc(sizeof(*mdev), GFP_KERNEL); + if (!mdev) return -ENOMEM; - } /* copy the port information */ mdev->client = pinfo->addr.client; diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c index 654d17a5023c..c080c73cea04 100644 --- a/sound/core/seq/oss/seq_oss_readq.c +++ b/sound/core/seq/oss/seq_oss_readq.c @@ -47,13 +47,12 @@ snd_seq_oss_readq_new(struct seq_oss_devinfo *dp, int maxlen) { struct seq_oss_readq *q; - if ((q = kzalloc(sizeof(*q), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc read queue\n"); + q = kzalloc(sizeof(*q), GFP_KERNEL); + if (!q) return NULL; - } - if ((q->q = kcalloc(maxlen, sizeof(union evrec), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc read queue buffer\n"); + q->q = kcalloc(maxlen, sizeof(union evrec), GFP_KERNEL); + if (!q->q) { kfree(q); return NULL; } diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 835edc80f918..48e4fe1b68ab 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -106,10 +106,9 @@ snd_seq_oss_synth_probe(struct device *_dev) struct snd_seq_oss_reg *reg = SNDRV_SEQ_DEVICE_ARGPTR(dev); unsigned long flags; - if ((rec = kzalloc(sizeof(*rec), GFP_KERNEL)) == NULL) { - pr_err("ALSA: seq_oss: can't malloc synth info\n"); + rec = kzalloc(sizeof(*rec), GFP_KERNEL); + if (!rec) return -ENOMEM; - } rec->seq_device = -1; rec->synth_type = reg->type; rec->synth_subtype = reg->subtype; @@ -249,7 +248,6 @@ snd_seq_oss_synth_setup(struct seq_oss_devinfo *dp) if (info->nr_voices > 0) { info->ch = kcalloc(info->nr_voices, sizeof(struct seq_oss_chinfo), GFP_KERNEL); if (!info->ch) { - pr_err("ALSA: seq_oss: Cannot malloc voices\n"); rec->oper.close(&info->arg); module_put(rec->oper.owner); snd_use_lock_free(&rec->use_lock); From 24db8bbaa3fcfaf0c2faccbff5864b58088ac1f6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 15:41:18 +0100 Subject: [PATCH 145/162] ALSA: seq: Drop superfluous error/debug messages after malloc failures The kernel memory allocators already report the errors when the requested allocation fails, thus we don't need to warn it again in each caller side. Signed-off-by: Takashi Iwai --- sound/core/seq/seq_fifo.c | 4 +--- sound/core/seq/seq_memory.c | 8 ++------ sound/core/seq/seq_ports.c | 4 +--- sound/core/seq/seq_prioq.c | 4 +--- sound/core/seq/seq_queue.c | 4 +--- sound/core/seq/seq_timer.c | 4 +--- 6 files changed, 7 insertions(+), 21 deletions(-) diff --git a/sound/core/seq/seq_fifo.c b/sound/core/seq/seq_fifo.c index 53a403e17c5b..1d5acbe0c08b 100644 --- a/sound/core/seq/seq_fifo.c +++ b/sound/core/seq/seq_fifo.c @@ -33,10 +33,8 @@ struct snd_seq_fifo *snd_seq_fifo_new(int poolsize) struct snd_seq_fifo *f; f = kzalloc(sizeof(*f), GFP_KERNEL); - if (f == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_fifo_new() \n"); + if (!f) return NULL; - } f->pool = snd_seq_pool_new(poolsize); if (f->pool == NULL) { diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index ba8e4a64e13e..801076687bb1 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -387,10 +387,8 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) return 0; pool->ptr = vmalloc(sizeof(struct snd_seq_event_cell) * pool->size); - if (pool->ptr == NULL) { - pr_debug("ALSA: seq: malloc for sequencer events failed\n"); + if (!pool->ptr) return -ENOMEM; - } /* add new cells to the free cell list */ spin_lock_irqsave(&pool->lock, flags); @@ -463,10 +461,8 @@ struct snd_seq_pool *snd_seq_pool_new(int poolsize) /* create pool block */ pool = kzalloc(sizeof(*pool), GFP_KERNEL); - if (pool == NULL) { - pr_debug("ALSA: seq: malloc failed for pool\n"); + if (!pool) return NULL; - } spin_lock_init(&pool->lock); pool->ptr = NULL; pool->free = NULL; diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 46ff593f618d..55170a20ae72 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -141,10 +141,8 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, /* create a new port */ new_port = kzalloc(sizeof(*new_port), GFP_KERNEL); - if (! new_port) { - pr_debug("ALSA: seq: malloc failed for registering client port\n"); + if (!new_port) return NULL; /* failure, out of memory */ - } /* init port data */ new_port->addr.client = client->number; new_port->addr.port = -1; diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c index 021b02bc9330..bc1c8488fc2a 100644 --- a/sound/core/seq/seq_prioq.c +++ b/sound/core/seq/seq_prioq.c @@ -59,10 +59,8 @@ struct snd_seq_prioq *snd_seq_prioq_new(void) struct snd_seq_prioq *f; f = kzalloc(sizeof(*f), GFP_KERNEL); - if (f == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_prioq_new()\n"); + if (!f) return NULL; - } spin_lock_init(&f->lock); f->head = NULL; diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index aad4878cee55..a0cda38205b9 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -111,10 +111,8 @@ static struct snd_seq_queue *queue_new(int owner, int locked) struct snd_seq_queue *q; q = kzalloc(sizeof(*q), GFP_KERNEL); - if (q == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_queue_new()\n"); + if (!q) return NULL; - } spin_lock_init(&q->owner_lock); spin_lock_init(&q->check_lock); diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index e73605393eee..186f1611103c 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -56,10 +56,8 @@ struct snd_seq_timer *snd_seq_timer_new(void) struct snd_seq_timer *tmr; tmr = kzalloc(sizeof(*tmr), GFP_KERNEL); - if (tmr == NULL) { - pr_debug("ALSA: seq: malloc failed for snd_seq_timer_new() \n"); + if (!tmr) return NULL; - } spin_lock_init(&tmr->lock); /* reset setup to defaults */ From ec0e9937aaa8b0a4b0633711c4d70d622acd9a7f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Mar 2015 15:42:14 +0100 Subject: [PATCH 146/162] ALSA: core: Drop superfluous error/debug messages after malloc failures The kernel memory allocators already report the errors when the requested allocation fails, thus we don't need to warn it again in each caller side. Signed-off-by: Takashi Iwai --- sound/core/control.c | 4 +--- sound/core/device.c | 4 +--- sound/core/hwdep.c | 4 +--- sound/core/oss/mixer_oss.c | 4 +--- sound/core/oss/pcm_oss.c | 1 - sound/core/pcm.c | 13 +++---------- sound/core/rawmidi.c | 8 ++------ sound/core/timer.c | 4 +--- 8 files changed, 10 insertions(+), 32 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index e1d8e0c816f0..833b223a363a 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -217,10 +217,8 @@ static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, size += sizeof(struct snd_kcontrol_volatile) * count; *kctl = kzalloc(size, GFP_KERNEL); - if (*kctl == NULL) { - pr_err("ALSA: Cannot allocate control instance\n"); + if (!*kctl) return -ENOMEM; - } for (idx = 0; idx < count; idx++) { (*kctl)->vd[idx].access = access; diff --git a/sound/core/device.c b/sound/core/device.c index 41bec3075ae5..c1a845b42a8b 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -50,10 +50,8 @@ int snd_device_new(struct snd_card *card, enum snd_device_type type, if (snd_BUG_ON(!card || !device_data || !ops)) return -ENXIO; dev = kzalloc(sizeof(*dev), GFP_KERNEL); - if (dev == NULL) { - dev_err(card->dev, "Cannot allocate device, type=%d\n", type); + if (!dev) return -ENOMEM; - } INIT_LIST_HEAD(&dev->list); dev->card = card; dev->type = type; diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 84244a5143cf..51692c8a39ea 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -378,10 +378,8 @@ int snd_hwdep_new(struct snd_card *card, char *id, int device, if (rhwdep) *rhwdep = NULL; hwdep = kzalloc(sizeof(*hwdep), GFP_KERNEL); - if (hwdep == NULL) { - dev_err(card->dev, "hwdep: cannot allocate\n"); + if (!hwdep) return -ENOMEM; - } init_waitqueue_head(&hwdep->open_wait); mutex_init(&hwdep->open_mutex); diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 5e6349f00ecd..056f8e274851 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1212,10 +1212,8 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, /* not changed */ goto __unlock; tbl = kmalloc(sizeof(*tbl), GFP_KERNEL); - if (! tbl) { - pr_err("ALSA: mixer_oss: no memory\n"); + if (!tbl) goto __unlock; - } tbl->oss_id = ch; tbl->name = kstrdup(str, GFP_KERNEL); if (! tbl->name) { diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 80423a4ccab6..58550cc93f28 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -854,7 +854,6 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) params = kmalloc(sizeof(*params), GFP_KERNEL); sparams = kmalloc(sizeof(*sparams), GFP_KERNEL); if (!sw_params || !params || !sparams) { - pcm_dbg(substream->pcm, "No memory\n"); err = -ENOMEM; goto failure; } diff --git a/sound/core/pcm.c b/sound/core/pcm.c index e9b87465c73d..b25bcf5b8644 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -343,11 +343,8 @@ static void snd_pcm_proc_info_read(struct snd_pcm_substream *substream, return; info = kmalloc(sizeof(*info), GFP_KERNEL); - if (! info) { - pcm_dbg(substream->pcm, - "snd_pcm_proc_info_read: cannot malloc\n"); + if (!info) return; - } err = snd_pcm_info(substream, info); if (err < 0) { @@ -717,10 +714,8 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) prev = NULL; for (idx = 0, prev = NULL; idx < substream_count; idx++) { substream = kzalloc(sizeof(*substream), GFP_KERNEL); - if (substream == NULL) { - pcm_err(pcm, "Cannot allocate PCM substream\n"); + if (!substream) return -ENOMEM; - } substream->pcm = pcm; substream->pstr = pstr; substream->number = idx; @@ -774,10 +769,8 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device, if (rpcm) *rpcm = NULL; pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); - if (pcm == NULL) { - dev_err(card->dev, "Cannot allocate PCM\n"); + if (!pcm) return -ENOMEM; - } pcm->card = card; pcm->device = device; pcm->internal = internal; diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index b5a748596fc4..a7759846fbaa 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1429,10 +1429,8 @@ static int snd_rawmidi_alloc_substreams(struct snd_rawmidi *rmidi, for (idx = 0; idx < count; idx++) { substream = kzalloc(sizeof(*substream), GFP_KERNEL); - if (substream == NULL) { - rmidi_err(rmidi, "rawmidi: cannot allocate substream\n"); + if (!substream) return -ENOMEM; - } substream->stream = direction; substream->number = idx; substream->rmidi = rmidi; @@ -1479,10 +1477,8 @@ int snd_rawmidi_new(struct snd_card *card, char *id, int device, if (rrawmidi) *rrawmidi = NULL; rmidi = kzalloc(sizeof(*rmidi), GFP_KERNEL); - if (rmidi == NULL) { - dev_err(card->dev, "rawmidi: cannot allocate\n"); + if (!rmidi) return -ENOMEM; - } rmidi->card = card; rmidi->device = device; mutex_init(&rmidi->open_mutex); diff --git a/sound/core/timer.c b/sound/core/timer.c index 490b489d713d..a9a1a047c521 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -774,10 +774,8 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, if (rtimer) *rtimer = NULL; timer = kzalloc(sizeof(*timer), GFP_KERNEL); - if (timer == NULL) { - pr_err("ALSA: timer: cannot allocate\n"); + if (!timer) return -ENOMEM; - } timer->tmr_class = tid->dev_class; timer->card = card; timer->tmr_device = tid->device; From 4945f1fdc14ef090abe50d1b5682bfc1e4763c06 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Mar 2015 12:50:15 +0100 Subject: [PATCH 147/162] ALSA: seq: Fix init order of snd_seq_device stuff When the sequencer driver is built in kernel, it may panic at boot because of the uninitialized snd_seq_bus_type. Initialize it properly via subsys_initcall() instead of module_init() to assure that the bus is registered beforehand. Reported-by: Fengguang Wu Fixes: 7c37ae5c625a ('ALSA: seq: Rewrite sequencer device binding with standard bus') Signed-off-by: Takashi Iwai --- sound/core/seq/seq_device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 355b34269bd1..d99f99d61983 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -311,5 +311,5 @@ static void __exit alsa_seq_device_exit(void) bus_unregister(&snd_seq_bus_type); } -module_init(alsa_seq_device_init) +subsys_initcall(alsa_seq_device_init) module_exit(alsa_seq_device_exit) From e79d74ab25339437447478e4dfe2b35c5b560512 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Mar 2015 16:57:51 +0100 Subject: [PATCH 148/162] ALSA: control: Fix breakage of user ctl element addition In the commit [2225e79b9b03: 'ALSA: core: reduce stack usage related to snd_ctl_new()'], the id field of the newly added kctl is untouched, thus all attribute like name string remain empty. The fix is just to add the forgotten memcpy of the id field. Fixes: 2225e79b9b03 ('ALSA: core: reduce stack usage related to snd_ctl_new()') Reviewed-by: Takashi Sakamoto Signed-off-by: Takashi Iwai --- sound/core/control.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/control.c b/sound/core/control.c index 54a412af3224..d677c27746e9 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1267,6 +1267,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, err = snd_ctl_new(&kctl, count, access, file); if (err < 0) return err; + memcpy(&kctl->id, &info->id, sizeof(kctl->id)); kctl->private_data = kzalloc(sizeof(struct user_element) + private_size, GFP_KERNEL); if (kctl->private_data == NULL) { From b2a0bafa758256442e04d1f34d6d0746b846d23d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 5 Mar 2015 17:21:32 +0100 Subject: [PATCH 149/162] ALSA: hda - Use shutdown driver ops instead of reboot notifier The driver shutdown ops is simpler than registering reboot notifier manually. There should be no functional change by this -- the codec driver calls its own callback while the bus driver just calls azx_stop() like before. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_bind.c | 10 ++++++++++ sound/pci/hda/hda_codec.c | 18 ------------------ sound/pci/hda/hda_codec.h | 1 - sound/pci/hda/hda_controller.c | 26 -------------------------- sound/pci/hda/hda_controller.h | 6 ------ sound/pci/hda/hda_intel.c | 16 +++++++++++++--- sound/pci/hda/hda_tegra.c | 16 +++++++++++++--- 7 files changed, 36 insertions(+), 57 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index a49bc45c2ea5..1f40ce3c1696 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -9,6 +9,7 @@ #include #include #include +#include #include #include "hda_codec.h" #include "hda_local.h" @@ -142,6 +143,14 @@ static int hda_codec_driver_remove(struct device *dev) return 0; } +static void hda_codec_driver_shutdown(struct device *dev) +{ + struct hda_codec *codec = dev_to_hda_codec(dev); + + if (!pm_runtime_suspended(dev) && codec->patch_ops.reboot_notify) + codec->patch_ops.reboot_notify(codec); +} + int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, struct module *owner) { @@ -150,6 +159,7 @@ int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name, drv->driver.bus = &snd_hda_bus_type; drv->driver.probe = hda_codec_driver_probe; drv->driver.remove = hda_codec_driver_remove; + drv->driver.shutdown = hda_codec_driver_shutdown; drv->driver.pm = &hda_codec_driver_pm; return driver_register(&drv->driver); } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6fecf57c8d7c..3e4fb7a8fdcb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4941,24 +4941,6 @@ static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) } } -/** - * snd_hda_bus_reboot_notify - call the reboot notifier of each codec - * @bus: HD-audio bus - */ -void snd_hda_bus_reboot_notify(struct hda_bus *bus) -{ - struct hda_codec *codec; - - if (!bus) - return; - list_for_each_entry(codec, &bus->codec_list, list) { - if (hda_codec_is_power_on(codec) && - codec->patch_ops.reboot_notify) - codec->patch_ops.reboot_notify(codec); - } -} -EXPORT_SYMBOL_GPL(snd_hda_bus_reboot_notify); - /** * snd_hda_multi_out_dig_open - open the digital out in the exclusive mode * @codec: the HDA codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index bf9efb7e1b9a..70851e6d5f10 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -563,7 +563,6 @@ extern const struct snd_pcm_chmap_elem snd_pcm_2_1_chmaps[]; * Misc */ void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); -void snd_hda_bus_reboot_notify(struct hda_bus *bus); void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index cae50d5ffb81..b1143f22a0c2 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -27,7 +27,6 @@ #include #include #include -#include #include #include #include "hda_controller.h" @@ -1972,30 +1971,5 @@ int azx_init_stream(struct azx *chip) } EXPORT_SYMBOL_GPL(azx_init_stream); -/* - * reboot notifier for hang-up problem at power-down - */ -static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf) -{ - struct azx *chip = container_of(nb, struct azx, reboot_notifier); - snd_hda_bus_reboot_notify(chip->bus); - azx_stop_chip(chip); - return NOTIFY_OK; -} - -void azx_notifier_register(struct azx *chip) -{ - chip->reboot_notifier.notifier_call = azx_halt; - register_reboot_notifier(&chip->reboot_notifier); -} -EXPORT_SYMBOL_GPL(azx_notifier_register); - -void azx_notifier_unregister(struct azx *chip) -{ - if (chip->reboot_notifier.notifier_call) - unregister_reboot_notifier(&chip->reboot_notifier); -} -EXPORT_SYMBOL_GPL(azx_notifier_unregister); - MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Common HDA driver functions"); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 94c1a4719f7f..be1b7ded8d82 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -362,9 +362,6 @@ struct azx { /* for debugging */ unsigned int last_cmd[AZX_MAX_CODECS]; - /* reboot notifier (for mysterious hangup problem at power-down) */ - struct notifier_block reboot_notifier; - #ifdef CONFIG_SND_HDA_DSP_LOADER struct azx_dev saved_azx_dev; #endif @@ -437,7 +434,4 @@ int azx_probe_codecs(struct azx *chip, unsigned int max_slots); int azx_codec_configure(struct azx *chip); int azx_init_stream(struct azx *chip); -void azx_notifier_register(struct azx *chip); -void azx_notifier_unregister(struct azx *chip); - #endif /* __SOUND_HDA_CONTROLLER_H */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index dbc5a593da46..25668fde8480 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1066,8 +1066,6 @@ static int azx_free(struct azx *chip) azx_del_card_list(chip); - azx_notifier_unregister(chip); - hda->init_failed = 1; /* to be sure */ complete_all(&hda->probe_wait); @@ -1900,7 +1898,6 @@ static int azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; - azx_notifier_register(chip); azx_add_card_list(chip); snd_hda_set_power_save(chip->bus, power_save * 1000); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) @@ -1921,6 +1918,18 @@ static void azx_remove(struct pci_dev *pci) snd_card_free(card); } +static void azx_shutdown(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct azx *chip; + + if (!card) + return; + chip = card->private_data; + if (chip && chip->running) + azx_stop_chip(chip); +} + /* PCI IDs */ static const struct pci_device_id azx_ids[] = { /* CPT */ @@ -2143,6 +2152,7 @@ static struct pci_driver azx_driver = { .id_table = azx_ids, .probe = azx_probe, .remove = azx_remove, + .shutdown = azx_shutdown, .driver = { .pm = AZX_PM_OPS, }, diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 7586abe91dfb..2e4fd5c56d3b 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -290,8 +290,6 @@ static int hda_tegra_dev_free(struct snd_device *device) int i; struct azx *chip = device->device_data; - azx_notifier_unregister(chip); - if (chip->initialized) { for (i = 0; i < chip->num_streams; i++) azx_stream_stop(chip, &chip->azx_dev[i]); @@ -502,7 +500,6 @@ static int hda_tegra_probe(struct platform_device *pdev) goto out_free; chip->running = 1; - azx_notifier_register(chip); snd_hda_set_power_save(chip->bus, power_save * 1000); return 0; @@ -517,6 +514,18 @@ static int hda_tegra_remove(struct platform_device *pdev) return snd_card_free(dev_get_drvdata(&pdev->dev)); } +static void hda_tegra_shutdown(struct platform_device *pdev) +{ + struct snd_card *card = dev_get_drvdata(&pdev->dev); + struct azx *chip; + + if (!card) + return; + chip = card->private_data; + if (chip && chip->running) + azx_stop_chip(chip); +} + static struct platform_driver tegra_platform_hda = { .driver = { .name = "tegra-hda", @@ -525,6 +534,7 @@ static struct platform_driver tegra_platform_hda = { }, .probe = hda_tegra_probe, .remove = hda_tegra_remove, + .shutdown = hda_tegra_shutdown, }; module_platform_driver(tegra_platform_hda); From fb83b6351052bf78686df2559f7ea6b10e596850 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Mar 2015 23:34:34 +0100 Subject: [PATCH 150/162] ALSA: hda - Simplify PCM setup overrides This patch does two things: - code refactoring with a local helper function, - allow codec drivers to provide the specific PCM stream info pointers only for overriding the non-NULL entries, instead of copying the whole. This simplifies the codec driver side (currently the only user is alc269's 44kHz fixed rate). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 110 ++++++++++++++++++---------------- sound/pci/hda/patch_realtek.c | 41 ------------- 2 files changed, 60 insertions(+), 91 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ebdbc023583d..27ce54701f0f 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5137,6 +5137,33 @@ static void fill_pcm_stream_name(char *str, size_t len, const char *sfx, strlcat(str, sfx, len); } +/* copy PCM stream info from @default_str, and override non-NULL entries + * from @spec_str and @nid + */ +static void setup_pcm_stream(struct hda_pcm_stream *str, + const struct hda_pcm_stream *default_str, + const struct hda_pcm_stream *spec_str, + hda_nid_t nid) +{ + *str = *default_str; + if (nid) + str->nid = nid; + if (spec_str) { + if (spec_str->substreams) + str->substreams = spec_str->substreams; + if (spec_str->channels_min) + str->channels_min = spec_str->channels_min; + if (spec_str->channels_max) + str->channels_max = spec_str->channels_max; + if (spec_str->rates) + str->rates = spec_str->rates; + if (spec_str->formats) + str->formats = spec_str->formats; + if (spec_str->maxbps) + str->maxbps = spec_str->maxbps; + } +} + /** * snd_hda_gen_build_pcms - build PCM streams based on the parsed results * @codec: the HDA codec @@ -5147,7 +5174,6 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; struct hda_pcm *info; - const struct hda_pcm_stream *p; bool have_multi_adcs; if (spec->no_analog) @@ -5162,11 +5188,10 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) spec->pcm_rec[0] = info; if (spec->multiout.num_dacs > 0) { - p = spec->stream_analog_playback; - if (!p) - p = &pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_analog_playback, + spec->stream_analog_playback, + spec->multiout.dac_nids[0]); info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels; if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT && @@ -5175,15 +5200,11 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) snd_pcm_2_1_chmaps; } if (spec->num_adc_nids) { - p = spec->stream_analog_capture; - if (!p) { - if (spec->dyn_adc_switch) - p = &dyn_adc_pcm_analog_capture; - else - p = &pcm_analog_capture; - } - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + (spec->dyn_adc_switch ? + &dyn_adc_pcm_analog_capture : &pcm_analog_capture), + spec->stream_analog_capture, + spec->adc_nids[0]); } skip_analog: @@ -5202,20 +5223,16 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) info->pcm_type = spec->dig_out_type; else info->pcm_type = HDA_PCM_TYPE_SPDIF; - if (spec->multiout.dig_out_nid) { - p = spec->stream_digital_playback; - if (!p) - p = &pcm_digital_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; - } - if (spec->dig_in_nid) { - p = spec->stream_digital_capture; - if (!p) - p = &pcm_digital_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; - } + if (spec->multiout.dig_out_nid) + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_digital_playback, + spec->stream_digital_playback, + spec->multiout.dig_out_nid); + if (spec->dig_in_nid) + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + &pcm_digital_capture, + spec->stream_digital_capture, + spec->dig_in_nid); } if (spec->no_analog) @@ -5236,31 +5253,24 @@ int snd_hda_gen_build_pcms(struct hda_codec *codec) if (!info) return -ENOMEM; spec->pcm_rec[2] = info; - if (spec->alt_dac_nid) { - p = spec->stream_analog_alt_playback; - if (!p) - p = &pcm_analog_alt_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *p; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = - spec->alt_dac_nid; - } else { - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = - pcm_null_stream; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; - } + if (spec->alt_dac_nid) + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_analog_alt_playback, + spec->stream_analog_alt_playback, + spec->alt_dac_nid); + else + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_PLAYBACK], + &pcm_null_stream, NULL, 0); if (have_multi_adcs) { - p = spec->stream_analog_alt_capture; - if (!p) - p = &pcm_analog_alt_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = *p; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = - spec->adc_nids[1]; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + &pcm_analog_alt_capture, + spec->stream_analog_alt_capture, + spec->adc_nids[1]); info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids - 1; } else { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = - pcm_null_stream; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = 0; + setup_pcm_stream(&info->stream[SNDRV_PCM_STREAM_CAPTURE], + &pcm_null_stream, NULL, 0); } } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2a61bda8115d..124eacf67fc4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2602,53 +2602,12 @@ static int patch_alc268(struct hda_codec *codec) * ALC269 */ -static int playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct hda_gen_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, - hinfo); -} - -static int playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct hda_gen_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, format, substream); -} - -static int playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct hda_gen_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - static const struct hda_pcm_stream alc269_44k_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 8, .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ - /* NID is set in alc_build_pcms */ - .ops = { - .open = playback_pcm_open, - .prepare = playback_pcm_prepare, - .cleanup = playback_pcm_cleanup - }, }; static const struct hda_pcm_stream alc269_44k_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ - /* NID is set in alc_build_pcms */ }; /* different alc269-variants */ From b56df151d3b97a95ba3d3a76c2867d00f2db7782 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:26 +0100 Subject: [PATCH 151/162] ALSA: ice1724: ESI W192M: Correct copy/paste from prodigy driver MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Correct copy/paste name from prodigy driver, no behaviour change, only name. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index bcf30a387b87..f65ac193cf1a 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -463,7 +463,7 @@ static int wtm_add_controls(struct snd_ice1712 *ice) static int wtm_init(struct snd_ice1712 *ice) { - static unsigned short stac_inits_prodigy[] = { + static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, (unsigned short)-1 }; @@ -475,7 +475,7 @@ static int wtm_init(struct snd_ice1712 *ice) ice->force_rdma1 = 1; /*initialize codec*/ - p = stac_inits_prodigy; + p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); From 7127744a5e14455a4b69429757fd811aa50ec4b8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:27 +0100 Subject: [PATCH 152/162] ALSA: ice1724: ESI W192M: Update eeprom structure to C99 standard MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Update eeprom structure to C99 standard to be compliant with change in alsa. It's just a notation change, no configuration change. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index f65ac193cf1a..6e1026e0e56b 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -485,19 +485,19 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ - 0x80, /* ACLINK : I2S */ - 0xf8, /* I2S: vol; 96k, 24bit, 192k */ - 0xc1 /*SPDIF: out-en, spidf ext out*/, - 0x9f, /* GPIO_DIR */ - 0xff, /* GPIO_DIR1 */ - 0x7f, /* GPIO_DIR2 */ - 0x9f, /* GPIO_MASK */ - 0xff, /* GPIO_MASK1 */ - 0x7f, /* GPIO_MASK2 */ - 0x16, /* GPIO_STATE */ - 0x80, /* GPIO_STATE1 */ - 0x00, /* GPIO_STATE2 */ + [ICE_EEP2_SYSCONF] = 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ + [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ + [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ + [ICE_EEP2_GPIO_DIR] = 0x9f, + [ICE_EEP2_GPIO_DIR1] = 0xff, + [ICE_EEP2_GPIO_DIR2] = 0x7f, + [ICE_EEP2_GPIO_MASK] = 0x9f, + [ICE_EEP2_GPIO_MASK1] = 0xff, + [ICE_EEP2_GPIO_MASK2] = 0x7f, + [ICE_EEP2_GPIO_STATE] = 0x16, + [ICE_EEP2_GPIO_STATE1] = 0x80, + [ICE_EEP2_GPIO_STATE2] = 0x00, }; From f8a8b3a835ef9e5f94163c9dc62fc2e2e375b10c Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:28 +0100 Subject: [PATCH 153/162] ALSA: ice1724: ESI W192M: Enable midi i/o of port envy24 chip as available MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Enable midi i/o port of envy24 chip as their are available on ESI W192M soundcard. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 6e1026e0e56b..59483b4d6dd0 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -485,7 +485,8 @@ static int wtm_init(struct snd_ice1712 *ice) static unsigned char wtm_eeprom[] = { - [ICE_EEP2_SYSCONF] = 0x47, /*SYSCONF: clock 192KHz, 4ADC, 8DAC */ + [ICE_EEP2_SYSCONF] = 0x67, /*SYSCONF: clock 192KHz, mpu401, + 4ADC, 8DAC */ [ICE_EEP2_ACLINK] = 0x80, /* ACLINK : I2S */ [ICE_EEP2_I2S] = 0xf8, /* I2S: vol; 96k, 24bit, 192k */ [ICE_EEP2_SPDIF] = 0xc1, /*SPDIF: out-en, spidf ext out*/ From 16ddbe738a5bd2afe80aa10492f762f34b09cbf0 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:29 +0100 Subject: [PATCH 154/162] ALSA: ice1724: ESI W192M: Add TLV support for control value in dB scale MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add TLV support to control volume using dB scale for input and ouput on ESI W192M. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 59483b4d6dd0..c7ffafaa1c5c 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,6 +29,7 @@ #include #include #include +#include #include "ice1712.h" #include "envy24ht.h" @@ -380,17 +381,25 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, return change; } + +/*Limits value in dB for fader*/ +static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); + /* * Control tabs */ static struct snd_kcontrol_new stac9640_controls[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), .name = "Master Playback Switch", .info = stac9460_dac_mute_info, .get = stac9460_dac_mute_get, .put = stac9460_dac_mute_put, - .private_value = 1 + .private_value = 1, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -419,11 +428,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "DAC Volume", .count = 8, .info = stac9460_dac_vol_info, .get = stac9460_dac_vol_get, .put = stac9460_dac_vol_put, + .tlv = { .p = db_scale_dac } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -435,12 +448,15 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "ADC Volume", .count = 2, .info = stac9460_adc_vol_info, .get = stac9460_adc_vol_get, .put = stac9460_adc_vol_put, - + .tlv = { .p = db_scale_adc } } }; From ae8a9a11256a0906831a7db39b8cbcdec51ae28a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:30 +0100 Subject: [PATCH 155/162] ALSA: ice1724: ESI W192M: Add text Line in/Mic for selecting input gain state MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add text Line in/Mic for selecting input gain state in mixer for ESI W192M. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index c7ffafaa1c5c..6d3412f72855 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -339,8 +339,14 @@ static int stac9460_adc_vol_put(struct snd_kcontrol *kcontrol, /* * MIC / LINE switch fonction */ +static int stac9460_mic_sw_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { "Line In", "Mic" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} -#define stac9460_mic_sw_info snd_ctl_boolean_mono_info static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -354,7 +360,7 @@ static int stac9460_mic_sw_get(struct snd_kcontrol *kcontrol, val = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else val = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - ucontrol->value.integer.value[0] = ~val>>7 & 0x1; + ucontrol->value.enumerated.item[0] = (val >> 7) & 0x1; return 0; } @@ -370,7 +376,7 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, old = stac9460_get(ice, STAC946X_GENERAL_PURPOSE); else old = stac9460_2_get(ice, STAC946X_GENERAL_PURPOSE); - new = (~ucontrol->value.integer.value[0] << 7 & 0x80) | (old & ~0x80); + new = (ucontrol->value.enumerated.item[0] << 7 & 0x80) | (old & ~0x80); change = (new != old); if (change) { if (id == 0) @@ -411,7 +417,7 @@ static struct snd_kcontrol_new stac9640_controls[] = { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "MIC/Line switch", + .name = "MIC/Line Input Enum", .count = 2, .info = stac9460_mic_sw_info, .get = stac9460_mic_sw_get, From 1aa9a4ea4fea5e4afe8be0229774b8f98db2e6c3 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Cl=C3=A9ment=20Guedez?= Date: Wed, 18 Mar 2015 02:26:31 +0100 Subject: [PATCH 156/162] ALSA: ice1724: ESI W192M: Add sampling rate control of the ADC/DAC MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add sampling rate control for ADC/DAC for ESI W192M. Allow to switch between 48K/96K/192K sampling rate. All DAC need to be mute when changing samplerate. Signed-off-by: Clément Guedez Signed-off-by: Takashi Iwai --- sound/pci/ice1712/wtm.c | 107 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 107 insertions(+) diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 6d3412f72855..9906119e0954 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -30,12 +30,18 @@ #include #include #include +#include #include "ice1712.h" #include "envy24ht.h" #include "wtm.h" #include "stac946x.h" +struct wtm_spec { + /* rate change needs atomic mute/unmute of all dacs*/ + struct mutex mute_mutex; +}; + /* * 2*ADC 6*DAC no1 ringbuffer r/w on i2c bus @@ -69,15 +75,65 @@ static inline unsigned char stac9460_2_get(struct snd_ice1712 *ice, int reg) /* * DAC mute control */ +static void stac9460_dac_mute_all(struct snd_ice1712 *ice, unsigned char mute, + unsigned short int *change_mask) +{ + unsigned char new, old; + int id, idx, change; + + /*stac9460 1*/ + for (id = 0; id < 7; id++) { + if (*change_mask & (0x01 << id)) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } + + /*stac9460 2*/ + for (id = 0; id < 3; id++) { + if (*change_mask & (0x01 << (id + 7))) { + if (id == 0) + idx = STAC946X_MASTER_VOLUME; + else + idx = STAC946X_LF_VOLUME - 1 + id; + old = stac9460_2_get(ice, idx); + new = (~mute << 7 & 0x80) | (old & ~0x80); + change = (new != old); + if (change) { + stac9460_2_put(ice, idx, new); + *change_mask = *change_mask | (0x01 << id); + } else { + *change_mask = *change_mask & ~(0x01 << id); + } + } + } +} + + + #define stac9460_dac_mute_info snd_ctl_boolean_mono_info static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + struct wtm_spec *spec = ice->spec; unsigned char val; int idx, id; + mutex_lock(&spec->mute_mutex); + if (kcontrol->private_value) { idx = STAC946X_MASTER_VOLUME; id = 0; @@ -90,6 +146,8 @@ static int stac9460_dac_mute_get(struct snd_kcontrol *kcontrol, else val = stac9460_2_get(ice, idx - 6); ucontrol->value.integer.value[0] = (~val >> 7) & 0x1; + + mutex_unlock(&spec->mute_mutex); return 0; } @@ -388,6 +446,44 @@ static int stac9460_mic_sw_put(struct snd_kcontrol *kcontrol, } +/* + * Handler for setting correct codec rate - called when rate change is detected + */ +static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned char old, new; + unsigned short int changed; + struct wtm_spec *spec = ice->spec; + + if (rate == 0) /* no hint - S/PDIF input is master, simply return */ + return; + else if (rate <= 48000) + new = 0x08; /* 256x, base rate mode */ + else if (rate <= 96000) + new = 0x11; /* 256x, mid rate mode */ + else + new = 0x12; /* 128x, high rate mode */ + + old = stac9460_get(ice, STAC946X_MASTER_CLOCKING); + if (old == new) + return; + /* change detected, setting master clock, muting first */ + /* due to possible conflicts with mute controls - mutexing */ + mutex_lock(&spec->mute_mutex); + /* we have to remember current mute status for each DAC */ + changed = 0xFFFF; + stac9460_dac_mute_all(ice, 0, &changed); + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); + stac9460_2_put(ice, STAC946X_MASTER_CLOCKING, new); + udelay(10); + /* unmuting - only originally unmuted dacs - + * i.e. those changed when muting */ + stac9460_dac_mute_all(ice, 1, &changed); + mutex_unlock(&spec->mute_mutex); +} + + /*Limits value in dB for fader*/ static const DECLARE_TLV_DB_SCALE(db_scale_dac, -19125, 75, 0); static const DECLARE_TLV_DB_SCALE(db_scale_adc, 0, 150, 0); @@ -487,21 +583,32 @@ static int wtm_init(struct snd_ice1712 *ice) { static unsigned short stac_inits_wtm[] = { STAC946X_RESET, 0, + STAC946X_MASTER_CLOCKING, 0x11, (unsigned short)-1 }; unsigned short *p; + struct wtm_spec *spec; /*WTM 192M*/ ice->num_total_dacs = 8; ice->num_total_adcs = 4; ice->force_rdma1 = 1; + /*init mutex for dac mute conflict*/ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + ice->spec = spec; + mutex_init(&spec->mute_mutex); + + /*initialize codec*/ p = stac_inits_wtm; for (; *p != (unsigned short)-1; p += 2) { stac9460_put(ice, p[0], p[1]); stac9460_2_put(ice, p[0], p[1]); } + ice->gpio.set_pro_rate = stac9460_set_rate_val; return 0; } From e6feb5d08509be1af2ebc894dae35f32f7b92ab6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Mar 2015 21:32:11 +0100 Subject: [PATCH 157/162] ALSA: hda - Support advanced power state controls This patch enables the finer power state control of each widget depending on the jack plug state and streaming state in addition to the existing power_down_unused power optimization. The new feature is enabled only when codec->power_mgmt flag is set. Two new flags, pin_enabled and stream_enabled, are introduced in nid_path struct for marking the two individual power states: the pin plug/unplug and DAC/ADC stream, respectively. They can be set statically in case they are static routes (e.g. some mixer paths), too. The power up and down events for each pin are triggered via the standard hda_jack table. The call order is hard-coded, relying on the current implementation of jack event chain (a la FILO/stack order). One point to be dealt carefully is that DAC/ADC cannot be powered on/off while streaming. They are pinned as long as the stream is running. For controlling the power of DAC/ADC, a new patch_ops is added. The generic parser provides the default callback for that. As of this patch, only IDT/Sigmatel codec driver enables the flag. The support on other codecs will follow. An assumption we made in this code is that the widget state (e.g. amp, pinctl, connections) remains after the widget power transition (not about FG power transition). This is true for IDT codecs, at least. But if the widget state is lost at widget power transition, we'd need to implement additional code to sync the cached amp/verbs for the specific NID. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 + sound/pci/hda/hda_codec.h | 2 + sound/pci/hda/hda_generic.c | 301 ++++++++++++++++++++++++++++----- sound/pci/hda/hda_generic.h | 5 +- sound/pci/hda/patch_sigmatel.c | 5 + 5 files changed, 269 insertions(+), 48 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3e4fb7a8fdcb..7e38d6f7314b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1502,6 +1502,8 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, if (!p) return; + if (codec->patch_ops.stream_pm) + codec->patch_ops.stream_pm(codec, nid, true); if (codec->pcm_format_first) update_pcm_format(codec, p, nid, format); update_pcm_stream_id(codec, p, nid, stream_tag, channel_id); @@ -1570,6 +1572,8 @@ static void really_cleanup_stream(struct hda_codec *codec, ); memset(q, 0, sizeof(*q)); q->nid = nid; + if (codec->patch_ops.stream_pm) + codec->patch_ops.stream_pm(codec, nid, false); } /* clean up the all conflicting obsolete streams */ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 70851e6d5f10..148e84ce61cf 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -200,6 +200,7 @@ struct hda_codec_ops { int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid); #endif void (*reboot_notify)(struct hda_codec *codec); + void (*stream_pm)(struct hda_codec *codec, hda_nid_t nid, bool on); }; /* record for amp information cache */ @@ -370,6 +371,7 @@ struct hda_codec { unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ + unsigned int power_mgmt:1; /* advanced PM for each widget */ #ifdef CONFIG_PM unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ atomic_t in_pm; /* suspend/resume being performed */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 27ce54701f0f..8a5055d296f5 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -140,6 +140,9 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "single_adc_amp"); if (val >= 0) codec->single_adc_amp = !!val; + val = snd_hda_get_bool_hint(codec, "power_mgmt"); + if (val >= 0) + codec->power_mgmt = !!val; val = snd_hda_get_bool_hint(codec, "auto_mute"); if (val >= 0) @@ -648,12 +651,21 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, unsigned int dir, unsigned int idx) { struct hda_gen_spec *spec = codec->spec; + int type = get_wcaps_type(get_wcaps(codec, nid)); int i, n; for (n = 0; n < spec->paths.used; n++) { struct nid_path *path = snd_array_elem(&spec->paths, n); if (!path->active) continue; + if (codec->power_mgmt) { + if (!path->stream_enabled) + continue; + /* ignore unplugged paths except for DAC/ADC */ + if (!path->pin_enabled && + type != AC_WID_AUD_OUT && type != AC_WID_AUD_IN) + continue; + } for (i = 0; i < path->depth; i++) { if (path->path[i] == nid) { if (dir == HDA_OUTPUT || path->idx[i] == idx) @@ -807,6 +819,42 @@ static void activate_amp_in(struct hda_codec *codec, struct nid_path *path, } } +/* sync power of each widget in the the given path */ +static hda_nid_t path_power_update(struct hda_codec *codec, + struct nid_path *path, + bool allow_powerdown) +{ + hda_nid_t nid, changed = 0; + int i, state; + + for (i = 0; i < path->depth; i++) { + nid = path->path[i]; + if (!allow_powerdown || is_active_nid_for_any(codec, nid)) + state = AC_PWRST_D0; + else + state = AC_PWRST_D3; + if (!snd_hda_check_power_state(codec, nid, state)) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_POWER_STATE, state); + changed = nid; + /* here we assume that widget attributes (e.g. amp, + * pinctl connection) don't change with local power + * state change. If not, need to sync the cache. + */ + } + } + return changed; +} + +/* do sync with the last power state change */ +static void sync_power_state_change(struct hda_codec *codec, hda_nid_t nid) +{ + if (nid) { + msleep(10); + snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); + } +} + /** * snd_hda_activate_path - activate or deactivate the given path * @codec: the HDA codec @@ -825,15 +873,13 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, if (!enable) path->active = false; + /* make sure the widget is powered up */ + if (enable && (spec->power_down_unused || codec->power_mgmt)) + path_power_update(codec, path, codec->power_mgmt); + for (i = path->depth - 1; i >= 0; i--) { hda_nid_t nid = path->path[i]; - if (enable && spec->power_down_unused) { - /* make sure the widget is powered up */ - if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D0)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D0); - } + if (enable && path->multi[i]) snd_hda_codec_update_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, @@ -853,28 +899,10 @@ EXPORT_SYMBOL_GPL(snd_hda_activate_path); static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) { struct hda_gen_spec *spec = codec->spec; - bool changed = false; - int i; - if (!spec->power_down_unused || path->active) + if (!(spec->power_down_unused || codec->power_mgmt) || path->active) return; - - for (i = 0; i < path->depth; i++) { - hda_nid_t nid = path->path[i]; - if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3) && - !is_active_nid_for_any(codec, nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_POWER_STATE, - AC_PWRST_D3); - changed = true; - } - } - - if (changed) { - msleep(10); - snd_hda_codec_read(codec, path->path[0], 0, - AC_VERB_GET_POWER_STATE, 0); - } + sync_power_state_change(codec, path_power_update(codec, path, true)); } /* turn on/off EAPD on the given pin */ @@ -1574,6 +1602,7 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) return 0; /* print_nid_path(codec, "output-aamix", path); */ path->active = false; /* unused as default */ + path->pin_enabled = true; /* static route */ return snd_hda_get_path_idx(codec, path); } @@ -2998,6 +3027,7 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, } path->active = true; + path->stream_enabled = true; /* no DAC/ADC involved */ err = add_loopback_list(spec, mix_nid, idx); if (err < 0) return err; @@ -3009,6 +3039,8 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, if (path) { print_nid_path(codec, "loopback-merge", path); path->active = true; + path->pin_enabled = true; /* static route */ + path->stream_enabled = true; /* no DAC/ADC involved */ spec->loopback_merge_path = snd_hda_get_path_idx(codec, path); } @@ -3810,6 +3842,7 @@ static void parse_digital(struct hda_codec *codec) continue; print_nid_path(codec, "digout", path); path->active = true; + path->pin_enabled = true; /* no jack detection */ spec->digout_paths[i] = snd_hda_get_path_idx(codec, path); set_pin_target(codec, pin, PIN_OUT, false); if (!nums) { @@ -3837,6 +3870,7 @@ static void parse_digital(struct hda_codec *codec) if (path) { print_nid_path(codec, "digin", path); path->active = true; + path->pin_enabled = true; /* no jack */ spec->dig_in_nid = dig_nid; spec->digin_path = snd_hda_get_path_idx(codec, path); set_pin_target(codec, pin, PIN_IN, false); @@ -3896,6 +3930,148 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, return 1; } +/* power up/down widgets in the all paths that match with the given NID + * as terminals (either start- or endpoint) + * + * returns the last changed NID, or zero if unchanged. + */ +static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, + int pin_state, int stream_state) +{ + struct hda_gen_spec *spec = codec->spec; + hda_nid_t last, changed = 0; + struct nid_path *path; + int n; + + for (n = 0; n < spec->paths.used; n++) { + path = snd_array_elem(&spec->paths, n); + if (path->path[0] == nid || + path->path[path->depth - 1] == nid) { + bool pin_old = path->pin_enabled; + bool stream_old = path->stream_enabled; + + if (pin_state >= 0) + path->pin_enabled = pin_state; + if (stream_state >= 0) + path->stream_enabled = stream_state; + if (path->pin_enabled != pin_old || + path->stream_enabled != stream_old) { + last = path_power_update(codec, path, true); + if (last) + changed = last; + } + } + } + return changed; +} + +/* power up/down the paths of the given pin according to the jack state; + * power = 0/1 : only power up/down if it matches with the jack state, + * < 0 : force power up/down to follow the jack sate + * + * returns the last changed NID, or zero if unchanged. + */ +static hda_nid_t set_pin_power_jack(struct hda_codec *codec, hda_nid_t pin, + int power) +{ + bool on; + + if (!codec->power_mgmt) + return 0; + + on = snd_hda_jack_detect_state(codec, pin) != HDA_JACK_NOT_PRESENT; + if (power >= 0 && on != power) + return 0; + return set_path_power(codec, pin, on, -1); +} + +static void pin_power_callback(struct hda_codec *codec, + struct hda_jack_callback *jack, + bool on) +{ + if (jack && jack->tbl->nid) + sync_power_state_change(codec, + set_pin_power_jack(codec, jack->tbl->nid, on)); +} + +/* callback only doing power up -- called at first */ +static void pin_power_up_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + pin_power_callback(codec, jack, true); +} + +/* callback only doing power down -- called at last */ +static void pin_power_down_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + pin_power_callback(codec, jack, false); +} + +/* set up the power up/down callbacks */ +static void add_pin_power_ctls(struct hda_codec *codec, int num_pins, + const hda_nid_t *pins, bool on) +{ + int i; + hda_jack_callback_fn cb = + on ? pin_power_up_callback : pin_power_down_callback; + + for (i = 0; i < num_pins && pins[i]; i++) { + if (is_jack_detectable(codec, pins[i])) + snd_hda_jack_detect_enable_callback(codec, pins[i], cb); + else + set_path_power(codec, pins[i], true, -1); + } +} + +/* enabled power callback to each available I/O pin with jack detections; + * the digital I/O pins are excluded because of the unreliable detectsion + */ +static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + if (!codec->power_mgmt) + return; + add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + add_pin_power_ctls(codec, cfg->hp_outs, cfg->hp_pins, on); + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) + add_pin_power_ctls(codec, cfg->speaker_outs, cfg->speaker_pins, on); + for (i = 0; i < cfg->num_inputs; i++) + add_pin_power_ctls(codec, 1, &cfg->inputs[i].pin, on); +} + +/* sync path power up/down with the jack states of given pins */ +static void sync_pin_power_ctls(struct hda_codec *codec, int num_pins, + const hda_nid_t *pins) +{ + int i; + + for (i = 0; i < num_pins && pins[i]; i++) + if (is_jack_detectable(codec, pins[i])) + set_pin_power_jack(codec, pins[i], -1); +} + +/* sync path power up/down with pins; called at init and resume */ +static void sync_all_pin_power_ctls(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + if (!codec->power_mgmt) + return; + sync_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins); + if (cfg->line_out_type != AUTO_PIN_HP_OUT) + sync_pin_power_ctls(codec, cfg->hp_outs, cfg->hp_pins); + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) + sync_pin_power_ctls(codec, cfg->speaker_outs, cfg->speaker_pins); + for (i = 0; i < cfg->num_inputs; i++) + sync_pin_power_ctls(codec, 1, &cfg->inputs[i].pin); +} /* * Jack detections for HP auto-mute and mic-switch @@ -3933,6 +4109,10 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, if (!nid) break; + oldval = snd_hda_codec_get_pin_target(codec, nid); + if (oldval & PIN_IN) + continue; /* no mute for inputs */ + if (spec->auto_mute_via_amp) { struct nid_path *path; hda_nid_t mute_nid; @@ -3947,29 +4127,33 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, spec->mute_bits |= (1ULL << mute_nid); else spec->mute_bits &= ~(1ULL << mute_nid); - set_pin_eapd(codec, nid, !mute); continue; + } else { + /* don't reset VREF value in case it's controlling + * the amp (see alc861_fixup_asus_amp_vref_0f()) + */ + if (spec->keep_vref_in_automute) + val = oldval & ~PIN_HP; + else + val = 0; + if (!mute) + val |= oldval; + /* here we call update_pin_ctl() so that the pinctl is + * changed without changing the pinctl target value; + * the original target value will be still referred at + * the init / resume again + */ + update_pin_ctl(codec, nid, val); } - oldval = snd_hda_codec_get_pin_target(codec, nid); - if (oldval & PIN_IN) - continue; /* no mute for inputs */ - /* don't reset VREF value in case it's controlling - * the amp (see alc861_fixup_asus_amp_vref_0f()) - */ - if (spec->keep_vref_in_automute) - val = oldval & ~PIN_HP; - else - val = 0; - if (!mute) - val |= oldval; - /* here we call update_pin_ctl() so that the pinctl is changed - * without changing the pinctl target value; - * the original target value will be still referred at the - * init / resume again - */ - update_pin_ctl(codec, nid, val); set_pin_eapd(codec, nid, !mute); + if (codec->power_mgmt) { + bool on = !mute; + if (on) + on = snd_hda_jack_detect_state(codec, nid) + != HDA_JACK_NOT_PRESENT; + set_path_power(codec, nid, on, -1); + } } } @@ -4465,6 +4649,21 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) } } +/** + * snd_hda_gen_stream_pm - Stream power management callback + * @codec: the HDA codec + * @nid: audio widget + * @on: power on/off flag + * + * Set this in patch_ops.stream_pm. Only valid with power_mgmt flag. + */ +void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on) +{ + if (codec->power_mgmt) + set_path_power(codec, nid, -1, on); +} +EXPORT_SYMBOL_GPL(snd_hda_gen_stream_pm); + /** * snd_hda_gen_parse_auto_config - Parse the given BIOS configuration and * set up the hda_gen_spec @@ -4549,6 +4748,9 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, if (err < 0) return err; + /* add power-down pin callbacks at first */ + add_all_pin_power_ctls(codec, false); + spec->const_channel_count = spec->ext_channel_count; /* check the multiple speaker and headphone pins */ if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) @@ -4618,6 +4820,9 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, } } + /* add power-up pin callbacks at last */ + add_all_pin_power_ctls(codec, true); + /* mute all aamix input initially */ if (spec->mixer_nid) mute_all_mixer_nid(codec, spec->mixer_nid); @@ -4625,7 +4830,7 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, dig_only: parse_digital(codec); - if (spec->power_down_unused) + if (spec->power_down_unused || codec->power_mgmt) codec->power_filter = snd_hda_gen_path_power_filter; if (!spec->no_analog && spec->beep_nid) { @@ -5478,6 +5683,8 @@ int snd_hda_gen_init(struct hda_codec *codec) clear_unsol_on_unused_pins(codec); + sync_all_pin_power_ctls(codec); + /* call init functions of standard auto-mute helpers */ update_automute_all(codec); diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index b211f889b335..54659b51fe16 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -46,7 +46,9 @@ struct nid_path { unsigned char idx[MAX_NID_PATH_DEPTH]; unsigned char multi[MAX_NID_PATH_DEPTH]; unsigned int ctls[NID_PATH_NUM_CTLS]; /* NID_PATH_XXX_CTL */ - bool active; + bool active:1; /* activated by driver */ + bool pin_enabled:1; /* pins are enabled */ + bool stream_enabled:1; /* stream is active */ }; /* mic/line-in auto switching entry */ @@ -340,5 +342,6 @@ int snd_hda_gen_check_power_status(struct hda_codec *codec, hda_nid_t nid); unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state); +void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on); #endif /* __SOUND_HDA_GENERIC_H */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 2956a6ba6bf0..86b944a6b0ed 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4394,6 +4394,7 @@ static const struct hda_codec_ops stac_patch_ops = { #ifdef CONFIG_PM .suspend = stac_suspend, #endif + .stream_pm = snd_hda_gen_stream_pm, .reboot_notify = stac_shutup, }; @@ -4487,6 +4488,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; spec = codec->spec; + codec->power_mgmt = 1; spec->linear_tone_beep = 0; spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; @@ -4592,6 +4594,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; + codec->power_mgmt = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4641,6 +4644,7 @@ static int patch_stac92hd95(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; + codec->power_mgmt = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4682,6 +4686,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return err; spec = codec->spec; + codec->power_mgmt = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; From 688b12cc3ca8a5155b95ce8d01e0e43006813b27 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 17 Mar 2015 15:56:05 +0100 Subject: [PATCH 158/162] ALSA: hda - Use the new power control for VIA codecs VIA codecs used to have the own power controls but they were disabled at transition to the generic parser due to the coding assuming the fixed routes. Now we get the proper support of equivalently fine power management in the generic parser, and the old kludges can be replaced with it. This results in the reduction of lots of dead codes. The advanced PM feature is disabled as default like before for keeping the compatible behavior. It's enabled via "Dynamic Power-Control" mixer element. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 662 +------------------------------------- 1 file changed, 12 insertions(+), 650 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 2112fbe9e577..d5d1dca4f11b 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -99,7 +99,6 @@ struct via_spec { /* HP mode source */ unsigned int dmic_enabled; - unsigned int no_pin_power_ctl; enum VIA_HDA_CODEC codec_type; /* analog low-power control */ @@ -108,9 +107,6 @@ struct via_spec { /* work to check hp jack state */ int hp_work_active; int vt1708_jack_detect; - - void (*set_widgets_power_state)(struct hda_codec *codec); - unsigned int dac_stream_tag[4]; }; static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec); @@ -133,11 +129,12 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) /* VT1708BCE & VT1708S are almost same */ if (spec->codec_type == VT1708BCE) spec->codec_type = VT1708S; - spec->no_pin_power_ctl = 1; spec->gen.indep_hp = 1; spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; + codec->power_mgmt = 1; + spec->gen.power_down_unused = 1; return spec; } @@ -229,90 +226,6 @@ static void vt1708_update_hp_work(struct hda_codec *codec) vt1708_stop_hp_work(codec); } -static void set_widgets_power_state(struct hda_codec *codec) -{ -#if 0 /* FIXME: the assumed connections don't match always with the - * actual routes by the generic parser, so better to disable - * the control for safety. - */ - struct via_spec *spec = codec->spec; - if (spec->set_widgets_power_state) - spec->set_widgets_power_state(codec); -#endif -} - -static void update_power_state(struct hda_codec *codec, hda_nid_t nid, - unsigned int parm) -{ - if (snd_hda_check_power_state(codec, nid, parm)) - return; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); -} - -static void update_conv_power_state(struct hda_codec *codec, hda_nid_t nid, - unsigned int parm, unsigned int index) -{ - struct via_spec *spec = codec->spec; - unsigned int format; - - if (snd_hda_check_power_state(codec, nid, parm)) - return; - format = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - if (format && (spec->dac_stream_tag[index] != format)) - spec->dac_stream_tag[index] = format; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); - if (parm == AC_PWRST_D0) { - format = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - if (!format && (spec->dac_stream_tag[index] != format)) - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - spec->dac_stream_tag[index]); - } -} - -static bool smart51_enabled(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - return spec->gen.ext_channel_count > 2; -} - -static bool is_smart51_pins(struct hda_codec *codec, hda_nid_t pin) -{ - struct via_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->gen.multi_ios; i++) - if (spec->gen.multi_io[i].pin == pin) - return true; - return false; -} - -static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, - unsigned int *affected_parm) -{ - unsigned parm; - unsigned def_conf = snd_hda_codec_get_pincfg(codec, nid); - unsigned no_presence = (def_conf & AC_DEFCFG_MISC) - >> AC_DEFCFG_MISC_SHIFT - & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ - struct via_spec *spec = codec->spec; - unsigned present = 0; - - no_presence |= spec->no_pin_power_ctl; - if (!no_presence) - present = snd_hda_jack_detect(codec, nid); - if ((smart51_enabled(codec) && is_smart51_pins(codec, nid)) - || ((no_presence || present) - && get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)) { - *affected_parm = AC_PWRST_D0; /* if it's connected */ - parm = AC_PWRST_D0; - } else - parm = AC_PWRST_D3; - - update_power_state(codec, nid, parm); -} - static int via_pin_power_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -323,8 +236,7 @@ static int via_pin_power_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct via_spec *spec = codec->spec; - ucontrol->value.enumerated.item[0] = !spec->no_pin_power_ctl; + ucontrol->value.enumerated.item[0] = codec->power_mgmt; return 0; } @@ -333,12 +245,12 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - unsigned int val = !ucontrol->value.enumerated.item[0]; + bool val = !!ucontrol->value.enumerated.item[0]; - if (val == spec->no_pin_power_ctl) + if (val == codec->power_mgmt) return 0; - spec->no_pin_power_ctl = val; - set_widgets_power_state(codec); + codec->power_mgmt = val; + spec->gen.power_down_unused = val; analog_low_current_mode(codec); return 1; } @@ -383,7 +295,7 @@ static void __analog_low_current_mode(struct hda_codec *codec, bool force) bool enable; unsigned int verb, parm; - if (spec->no_pin_power_ctl) + if (!codec->power_mgmt) enable = false; else enable = is_aa_path_mute(codec) && !spec->gen.active_streams; @@ -440,8 +352,7 @@ static int via_build_controls(struct hda_codec *codec) if (err < 0) return err; - if (spec->set_widgets_power_state) - spec->mixers[spec->num_mixers++] = via_pin_power_ctl_enum; + spec->mixers[spec->num_mixers++] = via_pin_power_ctl_enum; for (i = 0; i < spec->num_mixers; i++) { err = snd_hda_add_new_ctls(codec, spec->mixers[i]); @@ -485,7 +396,6 @@ static int via_suspend(struct hda_codec *codec) static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) { struct via_spec *spec = codec->spec; - set_widgets_power_state(codec); analog_low_current_mode(codec); vt1708_update_hp_work(codec); return snd_hda_check_amp_list_power(codec, &spec->gen.loopback, nid); @@ -573,34 +483,6 @@ static const struct snd_kcontrol_new vt1708_jack_detect_ctl[] = { {} /* terminator */ }; -static void via_jack_powerstate_event(struct hda_codec *codec, - struct hda_jack_callback *tbl) -{ - set_widgets_power_state(codec); -} - -static void via_set_jack_unsol_events(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->gen.autocfg; - hda_nid_t pin; - int i; - - for (i = 0; i < cfg->line_outs; i++) { - pin = cfg->line_out_pins[i]; - if (pin && is_jack_detectable(codec, pin)) - snd_hda_jack_detect_enable_callback(codec, pin, - via_jack_powerstate_event); - } - - for (i = 0; i < cfg->num_inputs; i++) { - pin = cfg->line_out_pins[i]; - if (pin && is_jack_detectable(codec, pin)) - snd_hda_jack_detect_enable_callback(codec, pin, - via_jack_powerstate_event); - } -} - static const struct badness_table via_main_out_badness = { .no_primary_dac = 0x10000, .no_dac = 0x4000, @@ -634,7 +516,9 @@ static int via_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - via_set_jack_unsol_events(codec); + /* disable widget PM at start for compatibility */ + codec->power_mgmt = 0; + spec->gen.power_down_unused = 0; return 0; } @@ -647,7 +531,6 @@ static int via_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->init_verbs[i]); /* init power states */ - set_widgets_power_state(codec); __analog_low_current_mode(codec, true); snd_hda_gen_init(codec); @@ -767,78 +650,6 @@ static int patch_vt1709(struct hda_codec *codec) return 0; } -static void set_widgets_power_state_vt1708B(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - int is_8ch = 0; - if ((spec->codec_type != VT1708B_4CH) && - (codec->vendor_id != 0x11064397)) - is_8ch = 1; - - /* SW0 (17h) = stereo mixer */ - imux_is_smixer = - (snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) - == ((spec->codec_type == VT1708S) ? 5 : 0)); - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW 0/1 (13h/14h) */ - update_power_state(codec, 0x17, parm); - update_power_state(codec, 0x13, parm); - update_power_state(codec, 0x14, parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1b, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x11, parm); - - /* PW6 (22h), SW2 (26h), AOW2 (24h) */ - if (is_8ch) { - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x22, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1a, &parm); - update_power_state(codec, 0x26, parm); - update_power_state(codec, 0x24, parm); - } else if (codec->vendor_id == 0x11064397) { - /* PW7(23h), SW2(27h), AOW2(25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x23, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1a, &parm); - update_power_state(codec, 0x27, parm); - update_power_state(codec, 0x25, parm); - } - - /* PW 3/4/7 (1ch/1dh/23h) */ - parm = AC_PWRST_D3; - /* force to D0 for internal Speaker */ - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - if (is_8ch) - set_pin_power_state(codec, 0x23, &parm); - - /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ - update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); - update_power_state(codec, 0x10, parm); - if (is_8ch) { - update_power_state(codec, 0x25, parm); - update_power_state(codec, 0x27, parm); - } else if (codec->vendor_id == 0x11064397 && spec->gen.indep_hp_enabled) - update_power_state(codec, 0x25, parm); -} - static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B(struct hda_codec *codec) { @@ -863,9 +674,6 @@ static int patch_vt1708B(struct hda_codec *codec) } codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1708B; - return 0; } @@ -931,8 +739,6 @@ static int patch_vt1708S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708S_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1708B; return 0; } @@ -946,36 +752,6 @@ static const struct hda_verb vt1702_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1702(struct hda_codec *codec) -{ - int imux_is_smixer = - snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - unsigned int parm; - /* inputs */ - /* PW 1/2/5 (14h/15h/18h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x14, &parm); - set_pin_power_state(codec, 0x15, &parm); - set_pin_power_state(codec, 0x18, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */ - /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ - update_power_state(codec, 0x13, parm); - update_power_state(codec, 0x12, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x20, parm); - - /* outputs */ - /* PW 3/4 (16h/17h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x17, &parm); - set_pin_power_state(codec, 0x16, &parm); - /* MW0 (1ah), AOW 0/1 (10h/1dh) */ - update_power_state(codec, 0x1a, imux_is_smixer ? AC_PWRST_D0 : parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x1d, parm); -} - static int patch_vt1702(struct hda_codec *codec) { struct via_spec *spec; @@ -1005,8 +781,6 @@ static int patch_vt1702(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1702_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1702; return 0; } @@ -1021,71 +795,6 @@ static const struct hda_verb vt1718S_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1718S(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm, parm2; - /* MUX6 (1eh) = stereo mixer */ - imux_is_smixer = - snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x27, &parm); - update_power_state(codec, 0x1a, parm); - parm2 = parm; /* for pin 0x0b */ - - /* PW2 (26h), AOW2 (ah) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x2b, &parm); - update_power_state(codec, 0xa, parm); - - /* PW0 (24h), AOW0 (8h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (!spec->gen.indep_hp_enabled) /* check for redirected HP */ - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x8, parm); - if (!spec->gen.indep_hp_enabled && parm2 != AC_PWRST_D3) - parm = parm2; - update_power_state(codec, 0xb, parm); - /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm); - - /* PW1 (25h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x2a, &parm); - update_power_state(codec, 0x9, parm); - - if (spec->gen.indep_hp_enabled) { - /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x1b, parm); - update_power_state(codec, 0x34, parm); - update_power_state(codec, 0xc, parm); - } -} - /* Add a connection to the primary DAC from AA-mixer for some codecs * This isn't listed from the raw info, but the chip has a secret connection. */ @@ -1146,9 +855,6 @@ static int patch_vt1718S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1718S_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1718S; - return 0; } @@ -1188,7 +894,6 @@ static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_CONNECT_SEL, index); spec->dmic_enabled = index; - set_widgets_power_state(codec); return 1; } @@ -1223,95 +928,6 @@ static const struct hda_verb vt1716S_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1716S(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - unsigned int mono_out, present; - /* SW0 (17h) = stereo mixer */ - imux_is_smixer = - (snd_hda_codec_read(codec, 0x17, 0, - AC_VERB_GET_CONNECT_SEL, 0x00) == 5); - /* inputs */ - /* PW 1/2/5 (1ah/1bh/1eh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1a, &parm); - set_pin_power_state(codec, 0x1b, &parm); - set_pin_power_state(codec, 0x1e, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* SW0 (17h), AIW0(13h) */ - update_power_state(codec, 0x17, parm); - update_power_state(codec, 0x13, parm); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1e, &parm); - /* PW11 (22h) */ - if (spec->dmic_enabled) - set_pin_power_state(codec, 0x22, &parm); - else - update_power_state(codec, 0x22, AC_PWRST_D3); - - /* SW2(26h), AIW1(14h) */ - update_power_state(codec, 0x26, parm); - update_power_state(codec, 0x14, parm); - - /* outputs */ - /* PW0 (19h), SW1 (18h), AOW1 (11h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x19, &parm); - /* Smart 5.1 PW2(1bh) */ - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1b, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x11, parm); - - /* PW7 (23h), SW3 (27h), AOW3 (25h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x23, &parm); - /* Smart 5.1 PW1(1ah) */ - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1a, &parm); - update_power_state(codec, 0x27, parm); - - /* Smart 5.1 PW5(1eh) */ - if (smart51_enabled(codec)) - set_pin_power_state(codec, 0x1e, &parm); - update_power_state(codec, 0x25, parm); - - /* Mono out */ - /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ - present = snd_hda_jack_detect(codec, 0x1c); - - if (present) - mono_out = 0; - else { - present = snd_hda_jack_detect(codec, 0x1d); - if (!spec->gen.indep_hp_enabled && present) - mono_out = 0; - else - mono_out = 1; - } - parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; - update_power_state(codec, 0x28, parm); - update_power_state(codec, 0x29, parm); - update_power_state(codec, 0x2a, parm); - - /* PW 3/4 (1ch/1dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x1c, &parm); - set_pin_power_state(codec, 0x1d, &parm); - /* HP Independent Mode, power on AOW3 */ - if (spec->gen.indep_hp_enabled) - update_power_state(codec, 0x25, parm); - - /* force to D0 for internal Speaker */ - /* MW0 (16h), AOW0 (10h) */ - update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm); - update_power_state(codec, 0x10, mono_out ? AC_PWRST_D0 : parm); -} - static int patch_vt1716S(struct hda_codec *codec) { struct via_spec *spec; @@ -1339,8 +955,6 @@ static int patch_vt1716S(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1716S; return 0; } @@ -1366,98 +980,6 @@ static const struct hda_verb vt1802_init_verbs[] = { { } }; -static void set_widgets_power_state_vt2002P(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm; - unsigned int present; - /* MUX9 (1eh) = stereo mixer */ - imux_is_smixer = - snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - parm = AC_PWRST_D0; - /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* AOW0 (8h)*/ - update_power_state(codec, 0x8, parm); - - if (spec->codec_type == VT1802) { - /* PW4 (28h), MW4 (18h), MUX4(38h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x38, parm); - } else { - /* PW4 (26h), MW4 (1ch), MUX4(37h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - update_power_state(codec, 0x1c, parm); - update_power_state(codec, 0x37, parm); - } - - if (spec->codec_type == VT1802) { - /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x15, parm); - update_power_state(codec, 0x35, parm); - } else { - /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x19, parm); - update_power_state(codec, 0x35, parm); - } - - if (spec->gen.indep_hp_enabled) - update_power_state(codec, 0x9, AC_PWRST_D0); - - /* Class-D */ - /* PW0 (24h), MW0(18h/14h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - if (spec->codec_type == VT1802) - update_power_state(codec, 0x14, parm); - else - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x34, parm); - - /* Mono Out */ - present = snd_hda_jack_detect(codec, 0x26); - - parm = present ? AC_PWRST_D3 : AC_PWRST_D0; - if (spec->codec_type == VT1802) { - /* PW15 (33h), MW8(1ch), MUX8(3ch) */ - update_power_state(codec, 0x33, parm); - update_power_state(codec, 0x1c, parm); - update_power_state(codec, 0x3c, parm); - } else { - /* PW15 (31h), MW8(17h), MUX8(3bh) */ - update_power_state(codec, 0x31, parm); - update_power_state(codec, 0x17, parm); - update_power_state(codec, 0x3b, parm); - } - /* MW9 (21h) */ - if (imux_is_smixer || !is_aa_path_mute(codec)) - update_power_state(codec, 0x21, AC_PWRST_D0); - else - update_power_state(codec, 0x21, AC_PWRST_D3); -} - /* * pin fix-up */ @@ -1541,8 +1063,6 @@ static int patch_vt2002P(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt2002P_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt2002P; return 0; } @@ -1556,81 +1076,6 @@ static const struct hda_verb vt1812_init_verbs[] = { { } }; -static void set_widgets_power_state_vt1812(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - unsigned int parm; - unsigned int present; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - parm = AC_PWRST_D0; - /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* AOW0 (8h)*/ - update_power_state(codec, 0x8, AC_PWRST_D0); - - /* PW4 (28h), MW4 (18h), MUX4(38h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x18, parm); - update_power_state(codec, 0x38, parm); - - /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x15, parm); - update_power_state(codec, 0x35, parm); - if (spec->gen.indep_hp_enabled) - update_power_state(codec, 0x9, AC_PWRST_D0); - - /* Internal Speaker */ - /* PW0 (24h), MW0(14h), MUX0(34h) */ - present = snd_hda_jack_detect(codec, 0x25); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - if (present) { - update_power_state(codec, 0x14, AC_PWRST_D3); - update_power_state(codec, 0x34, AC_PWRST_D3); - } else { - update_power_state(codec, 0x14, AC_PWRST_D0); - update_power_state(codec, 0x34, AC_PWRST_D0); - } - - - /* Mono Out */ - /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ - present = snd_hda_jack_detect(codec, 0x28); - - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x31, &parm); - if (present) { - update_power_state(codec, 0x1c, AC_PWRST_D3); - update_power_state(codec, 0x3c, AC_PWRST_D3); - update_power_state(codec, 0x3e, AC_PWRST_D3); - } else { - update_power_state(codec, 0x1c, AC_PWRST_D0); - update_power_state(codec, 0x3c, AC_PWRST_D0); - update_power_state(codec, 0x3e, AC_PWRST_D0); - } - - /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x33, &parm); - update_power_state(codec, 0x1d, parm); - update_power_state(codec, 0x3d, parm); - -} - /* patch for vt1812 */ static int patch_vt1812(struct hda_codec *codec) { @@ -1657,8 +1102,6 @@ static int patch_vt1812(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1812_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt1812; return 0; } @@ -1674,84 +1117,6 @@ static const struct hda_verb vt3476_init_verbs[] = { { } }; -static void set_widgets_power_state_vt3476(struct hda_codec *codec) -{ - struct via_spec *spec = codec->spec; - int imux_is_smixer; - unsigned int parm, parm2; - /* MUX10 (1eh) = stereo mixer */ - imux_is_smixer = - snd_hda_codec_read(codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 4; - /* inputs */ - /* PW 5/6/7 (29h/2ah/2bh) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x29, &parm); - set_pin_power_state(codec, 0x2a, &parm); - set_pin_power_state(codec, 0x2b, &parm); - if (imux_is_smixer) - parm = AC_PWRST_D0; - /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ - update_power_state(codec, 0x1e, parm); - update_power_state(codec, 0x1f, parm); - update_power_state(codec, 0x10, parm); - update_power_state(codec, 0x11, parm); - - /* outputs */ - /* PW3 (27h), MW3(37h), AOW3 (bh) */ - if (spec->codec_type == VT1705CF) { - parm = AC_PWRST_D3; - update_power_state(codec, 0x27, parm); - update_power_state(codec, 0x37, parm); - } else { - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x27, &parm); - update_power_state(codec, 0x37, parm); - } - - /* PW2 (26h), MW2(36h), AOW2 (ah) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x26, &parm); - update_power_state(codec, 0x36, parm); - if (smart51_enabled(codec)) { - /* PW7(2bh), MW7(3bh), MUX7(1Bh) */ - set_pin_power_state(codec, 0x2b, &parm); - update_power_state(codec, 0x3b, parm); - update_power_state(codec, 0x1b, parm); - } - update_conv_power_state(codec, 0xa, parm, 2); - - /* PW1 (25h), MW1(35h), AOW1 (9h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x25, &parm); - update_power_state(codec, 0x35, parm); - if (smart51_enabled(codec)) { - /* PW6(2ah), MW6(3ah), MUX6(1ah) */ - set_pin_power_state(codec, 0x2a, &parm); - update_power_state(codec, 0x3a, parm); - update_power_state(codec, 0x1a, parm); - } - update_conv_power_state(codec, 0x9, parm, 1); - - /* PW4 (28h), MW4 (38h), MUX4(18h), AOW3(bh)/AOW0(8h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x28, &parm); - update_power_state(codec, 0x38, parm); - update_power_state(codec, 0x18, parm); - if (spec->gen.indep_hp_enabled) - update_conv_power_state(codec, 0xb, parm, 3); - parm2 = parm; /* for pin 0x0b */ - - /* PW0 (24h), MW0(34h), MW9(3fh), AOW0 (8h) */ - parm = AC_PWRST_D3; - set_pin_power_state(codec, 0x24, &parm); - update_power_state(codec, 0x34, parm); - if (!spec->gen.indep_hp_enabled && parm2 != AC_PWRST_D3) - parm = parm2; - update_conv_power_state(codec, 0x8, parm, 0); - /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ - update_power_state(codec, 0x3f, imux_is_smixer ? AC_PWRST_D0 : parm); -} - static int patch_vt3476(struct hda_codec *codec) { struct via_spec *spec; @@ -1775,9 +1140,6 @@ static int patch_vt3476(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt3476_init_verbs; codec->patch_ops = via_patch_ops; - - spec->set_widgets_power_state = set_widgets_power_state_vt3476; - return 0; } From 5ccf835cc76d89bc0d426659c63d81f609050842 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 18 Mar 2015 09:23:10 +0100 Subject: [PATCH 159/162] ALSA: hda - Adjust power of beep widget and outputs As the widget PM may turn off the pins, this might lead to the silent output for beep when no explicit paths are given. This patch adds fake output paths for the beep widget so that the output pins are dynamically powered upon beep on/off. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 29 +++++++++------- sound/pci/hda/hda_beep.h | 1 + sound/pci/hda/hda_generic.c | 69 +++++++++++++++++++++++++++++++++++++ 3 files changed, 86 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 581b7fdef0e3..4cdac3a71cae 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -33,30 +33,36 @@ enum { DIGBEEP_HZ_MAX = 12000000, /* 12 KHz */ }; -static void snd_hda_generate_beep(struct work_struct *work) +/* generate or stop tone */ +static void generate_tone(struct hda_beep *beep, int tone) { - struct hda_beep *beep = - container_of(work, struct hda_beep, beep_work); struct hda_codec *codec = beep->codec; - int tone; - if (!beep->enabled) - return; - - tone = beep->tone; if (tone && !beep->playing) { snd_hda_power_up(codec); + if (beep->power_hook) + beep->power_hook(beep, true); beep->playing = 1; } - /* generate tone */ snd_hda_codec_write(codec, beep->nid, 0, AC_VERB_SET_BEEP_CONTROL, tone); if (!tone && beep->playing) { beep->playing = 0; + if (beep->power_hook) + beep->power_hook(beep, false); snd_hda_power_down(codec); } } +static void snd_hda_generate_beep(struct work_struct *work) +{ + struct hda_beep *beep = + container_of(work, struct hda_beep, beep_work); + + if (beep->enabled) + generate_tone(beep, beep->tone); +} + /* (non-standard) Linear beep tone calculation for IDT/STAC codecs * * The tone frequency of beep generator on IDT/STAC codecs is @@ -130,10 +136,7 @@ static void turn_off_beep(struct hda_beep *beep) cancel_work_sync(&beep->beep_work); if (beep->playing) { /* turn off beep */ - snd_hda_codec_write(beep->codec, beep->nid, 0, - AC_VERB_SET_BEEP_CONTROL, 0); - beep->playing = 0; - snd_hda_power_down(beep->codec); + generate_tone(beep, 0); } } diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index a63b5e077332..46524ff7e79e 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -40,6 +40,7 @@ struct hda_beep { unsigned int playing:1; struct work_struct beep_work; /* scheduled task for beep event */ struct mutex mutex; + void (*power_hook)(struct hda_beep *beep, bool on); }; #ifdef CONFIG_SND_HDA_INPUT_BEEP diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8a5055d296f5..d7ca388651da 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -654,6 +654,9 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, int type = get_wcaps_type(get_wcaps(codec, nid)); int i, n; + if (nid == codec->afg) + return true; + for (n = 0; n < spec->paths.used; n++) { struct nid_path *path = snd_array_elem(&spec->paths, n); if (!path->active) @@ -829,6 +832,8 @@ static hda_nid_t path_power_update(struct hda_codec *codec, for (i = 0; i < path->depth; i++) { nid = path->path[i]; + if (nid == codec->afg) + continue; if (!allow_powerdown || is_active_nid_for_any(codec, nid)) state = AC_PWRST_D0; else @@ -4073,6 +4078,64 @@ static void sync_all_pin_power_ctls(struct hda_codec *codec) sync_pin_power_ctls(codec, 1, &cfg->inputs[i].pin); } +/* add fake paths if not present yet */ +static int add_fake_paths(struct hda_codec *codec, hda_nid_t nid, + int num_pins, const hda_nid_t *pins) +{ + struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; + int i; + + for (i = 0; i < num_pins; i++) { + if (!pins[i]) + break; + if (get_nid_path(codec, nid, pins[i], 0)) + continue; + path = snd_array_new(&spec->paths); + if (!path) + return -ENOMEM; + memset(path, 0, sizeof(*path)); + path->depth = 2; + path->path[0] = nid; + path->path[1] = pins[i]; + path->active = true; + } + return 0; +} + +/* create fake paths to all outputs from beep */ +static int add_fake_beep_paths(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid = spec->beep_nid; + int err; + + if (!codec->power_mgmt || !nid) + return 0; + err = add_fake_paths(codec, nid, cfg->line_outs, cfg->line_out_pins); + if (err < 0) + return err; + if (cfg->line_out_type != AUTO_PIN_HP_OUT) { + err = add_fake_paths(codec, nid, cfg->hp_outs, cfg->hp_pins); + if (err < 0) + return err; + } + if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + err = add_fake_paths(codec, nid, cfg->speaker_outs, + cfg->speaker_pins); + if (err < 0) + return err; + } + return 0; +} + +/* power up/down beep widget and its output paths */ +static void beep_power_hook(struct hda_beep *beep, bool on) +{ + set_path_power(beep->codec, beep->nid, -1, on); +} + /* * Jack detections for HP auto-mute and mic-switch */ @@ -4837,6 +4900,12 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, err = snd_hda_attach_beep_device(codec, spec->beep_nid); if (err < 0) return err; + if (codec->beep && codec->power_mgmt) { + err = add_fake_beep_paths(codec); + if (err < 0) + return err; + codec->beep->power_hook = beep_power_hook; + } } return 1; From b24062bda7baba62781c2a67d126126ce0bc8899 Mon Sep 17 00:00:00 2001 From: Fabian Frederick Date: Wed, 18 Mar 2015 17:48:56 +0100 Subject: [PATCH 160/162] ALSA: aoa: constify of_device_id array of_device_id is always used as const. (See driver.of_match_table and open firmware functions) Signed-off-by: Fabian Frederick Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/i2sbus/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index b9737fae656a..1cbf210080a1 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -31,7 +31,7 @@ module_param(force, int, 0444); MODULE_PARM_DESC(force, "Force loading i2sbus even when" " no layout-id property is present"); -static struct of_device_id i2sbus_match[] = { +static const struct of_device_id i2sbus_match[] = { { .name = "i2s" }, { } }; From 6b275b140094b701f7ad15272f0597e9d954e5e4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Mar 2015 18:11:05 +0100 Subject: [PATCH 161/162] ALSA: hda - Fix power of pins used for mute LED with vrefs Some pins are used for controlling the LED with the VREF value. This patch changes the power behavior of such pins to be constantly up. A new state, pin_fixed, is introduced to nid_path to indicate that the path contains the fixed pin. This improves also the readability a bit for other static routes, too. Then a helper function snd_hda_gen_fix_pin_power() is called from the codec driver for such fixed pins, and it will create fake paths containing only these pins with pin_fixed=1 flag. Reported-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 37 +++++++++++++++++++++++++++------- sound/pci/hda/hda_generic.h | 2 ++ sound/pci/hda/patch_sigmatel.c | 6 ++++++ 3 files changed, 38 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index d7ca388651da..1cafcbb9d391 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -665,7 +665,7 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, if (!path->stream_enabled) continue; /* ignore unplugged paths except for DAC/ADC */ - if (!path->pin_enabled && + if (!(path->pin_enabled || path->pin_fixed) && type != AC_WID_AUD_OUT && type != AC_WID_AUD_IN) continue; } @@ -1607,7 +1607,7 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) return 0; /* print_nid_path(codec, "output-aamix", path); */ path->active = false; /* unused as default */ - path->pin_enabled = true; /* static route */ + path->pin_fixed = true; /* static route */ return snd_hda_get_path_idx(codec, path); } @@ -3044,7 +3044,7 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, if (path) { print_nid_path(codec, "loopback-merge", path); path->active = true; - path->pin_enabled = true; /* static route */ + path->pin_fixed = true; /* static route */ path->stream_enabled = true; /* no DAC/ADC involved */ spec->loopback_merge_path = snd_hda_get_path_idx(codec, path); @@ -3847,7 +3847,7 @@ static void parse_digital(struct hda_codec *codec) continue; print_nid_path(codec, "digout", path); path->active = true; - path->pin_enabled = true; /* no jack detection */ + path->pin_fixed = true; /* no jack detection */ spec->digout_paths[i] = snd_hda_get_path_idx(codec, path); set_pin_target(codec, pin, PIN_OUT, false); if (!nums) { @@ -3875,7 +3875,7 @@ static void parse_digital(struct hda_codec *codec) if (path) { print_nid_path(codec, "digin", path); path->active = true; - path->pin_enabled = true; /* no jack */ + path->pin_fixed = true; /* no jack */ spec->dig_in_nid = dig_nid; spec->digin_path = snd_hda_get_path_idx(codec, path); set_pin_target(codec, pin, PIN_IN, false); @@ -3959,8 +3959,8 @@ static hda_nid_t set_path_power(struct hda_codec *codec, hda_nid_t nid, path->pin_enabled = pin_state; if (stream_state >= 0) path->stream_enabled = stream_state; - if (path->pin_enabled != pin_old || - path->stream_enabled != stream_old) { + if ((!path->pin_fixed && path->pin_enabled != pin_old) + || path->stream_enabled != stream_old) { last = path_power_update(codec, path, true); if (last) changed = last; @@ -4136,6 +4136,29 @@ static void beep_power_hook(struct hda_beep *beep, bool on) set_path_power(beep->codec, beep->nid, -1, on); } +/** + * snd_hda_gen_fix_pin_power - Fix the power of the given pin widget to D0 + * @codec: the HDA codec + * @pin: NID of pin to fix + */ +int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin) +{ + struct hda_gen_spec *spec = codec->spec; + struct nid_path *path; + + path = snd_array_new(&spec->paths); + if (!path) + return -ENOMEM; + memset(path, 0, sizeof(*path)); + path->depth = 1; + path->path[0] = pin; + path->active = true; + path->pin_fixed = true; + path->stream_enabled = true; + return 0; +} +EXPORT_SYMBOL_GPL(snd_hda_gen_fix_pin_power); + /* * Jack detections for HP auto-mute and mic-switch */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 54659b51fe16..56e4139b9032 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -48,6 +48,7 @@ struct nid_path { unsigned int ctls[NID_PATH_NUM_CTLS]; /* NID_PATH_XXX_CTL */ bool active:1; /* activated by driver */ bool pin_enabled:1; /* pins are enabled */ + bool pin_fixed:1; /* path with fixed pin */ bool stream_enabled:1; /* stream is active */ }; @@ -343,5 +344,6 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, hda_nid_t nid, unsigned int power_state); void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on); +int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin); #endif /* __SOUND_HDA_GENERIC_H */ diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 86b944a6b0ed..7e531d5cde51 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4225,6 +4225,12 @@ static int stac_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + if (spec->vref_mute_led_nid) { + err = snd_hda_gen_fix_pin_power(codec, spec->vref_mute_led_nid); + if (err < 0) + return err; + } + /* setup analog beep controls */ if (spec->anabeep_nid > 0) { err = stac_auto_create_beep_ctls(codec, From 967b1307b69b8ada8b331e01046ad1ef83742e99 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Mar 2015 18:21:03 +0100 Subject: [PATCH 162/162] ALSA: hda - Rename power_mgmt flag with power_save_node David suggested that the name "power_mgmt" is too ambiguous. Rename the flag with a bit clearer one "power_save_node". Also, add the corresponding description to HD-Audio.txt, too. Reported-by: David Henningsson Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 6 +++++- sound/pci/hda/hda_codec.h | 2 +- sound/pci/hda/hda_generic.c | 30 +++++++++++++-------------- sound/pci/hda/patch_sigmatel.c | 8 +++---- sound/pci/hda/patch_via.c | 12 +++++------ 5 files changed, 31 insertions(+), 27 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index 42a0a39b77e6..e7193aac669c 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -466,7 +466,11 @@ The generic parser supports the following hints: - add_jack_modes (bool): add "xxx Jack Mode" enum controls to each I/O jack for allowing to change the headphone amp and mic bias VREF capabilities -- power_down_unused (bool): power down the unused widgets +- power_save_node (bool): advanced power management for each widget, + controlling the power sate (D0/D3) of each widget node depending on + the actual pin and stream states +- power_down_unused (bool): power down the unused widgets, a subset of + power_save_node, and will be dropped in future - add_hp_mic (bool): add the headphone to capture source if possible - hp_mic_detect (bool): enable/disable the hp/mic shared input for a single built-in mic case; default true diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 148e84ce61cf..ccf355d4a8fa 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -371,7 +371,7 @@ struct hda_codec { unsigned int cached_write:1; /* write only to caches */ unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ unsigned int dump_coef:1; /* dump processing coefs in codec proc file */ - unsigned int power_mgmt:1; /* advanced PM for each widget */ + unsigned int power_save_node:1; /* advanced PM for each widget */ #ifdef CONFIG_PM unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ atomic_t in_pm; /* suspend/resume being performed */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 1cafcbb9d391..0ef2459cd05f 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -140,9 +140,9 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "single_adc_amp"); if (val >= 0) codec->single_adc_amp = !!val; - val = snd_hda_get_bool_hint(codec, "power_mgmt"); + val = snd_hda_get_bool_hint(codec, "power_save_node"); if (val >= 0) - codec->power_mgmt = !!val; + codec->power_save_node = !!val; val = snd_hda_get_bool_hint(codec, "auto_mute"); if (val >= 0) @@ -661,7 +661,7 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, struct nid_path *path = snd_array_elem(&spec->paths, n); if (!path->active) continue; - if (codec->power_mgmt) { + if (codec->power_save_node) { if (!path->stream_enabled) continue; /* ignore unplugged paths except for DAC/ADC */ @@ -879,8 +879,8 @@ void snd_hda_activate_path(struct hda_codec *codec, struct nid_path *path, path->active = false; /* make sure the widget is powered up */ - if (enable && (spec->power_down_unused || codec->power_mgmt)) - path_power_update(codec, path, codec->power_mgmt); + if (enable && (spec->power_down_unused || codec->power_save_node)) + path_power_update(codec, path, codec->power_save_node); for (i = path->depth - 1; i >= 0; i--) { hda_nid_t nid = path->path[i]; @@ -905,7 +905,7 @@ static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) { struct hda_gen_spec *spec = codec->spec; - if (!(spec->power_down_unused || codec->power_mgmt) || path->active) + if (!(spec->power_down_unused || codec->power_save_node) || path->active) return; sync_power_state_change(codec, path_power_update(codec, path, true)); } @@ -3981,7 +3981,7 @@ static hda_nid_t set_pin_power_jack(struct hda_codec *codec, hda_nid_t pin, { bool on; - if (!codec->power_mgmt) + if (!codec->power_save_node) return 0; on = snd_hda_jack_detect_state(codec, pin) != HDA_JACK_NOT_PRESENT; @@ -4038,7 +4038,7 @@ static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - if (!codec->power_mgmt) + if (!codec->power_save_node) return; add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); if (cfg->line_out_type != AUTO_PIN_HP_OUT) @@ -4067,7 +4067,7 @@ static void sync_all_pin_power_ctls(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; int i; - if (!codec->power_mgmt) + if (!codec->power_save_node) return; sync_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins); if (cfg->line_out_type != AUTO_PIN_HP_OUT) @@ -4111,7 +4111,7 @@ static int add_fake_beep_paths(struct hda_codec *codec) hda_nid_t nid = spec->beep_nid; int err; - if (!codec->power_mgmt || !nid) + if (!codec->power_save_node || !nid) return 0; err = add_fake_paths(codec, nid, cfg->line_outs, cfg->line_out_pins); if (err < 0) @@ -4233,7 +4233,7 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, } set_pin_eapd(codec, nid, !mute); - if (codec->power_mgmt) { + if (codec->power_save_node) { bool on = !mute; if (on) on = snd_hda_jack_detect_state(codec, nid) @@ -4741,11 +4741,11 @@ static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) * @nid: audio widget * @on: power on/off flag * - * Set this in patch_ops.stream_pm. Only valid with power_mgmt flag. + * Set this in patch_ops.stream_pm. Only valid with power_save_node flag. */ void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on) { - if (codec->power_mgmt) + if (codec->power_save_node) set_path_power(codec, nid, -1, on); } EXPORT_SYMBOL_GPL(snd_hda_gen_stream_pm); @@ -4916,14 +4916,14 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, dig_only: parse_digital(codec); - if (spec->power_down_unused || codec->power_mgmt) + if (spec->power_down_unused || codec->power_save_node) codec->power_filter = snd_hda_gen_path_power_filter; if (!spec->no_analog && spec->beep_nid) { err = snd_hda_attach_beep_device(codec, spec->beep_nid); if (err < 0) return err; - if (codec->beep && codec->power_mgmt) { + if (codec->beep && codec->power_save_node) { err = add_fake_beep_paths(codec); if (err < 0) return err; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7e531d5cde51..5b7c173adcb8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4494,7 +4494,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.mixer_nid = 0x1d; spec->have_spdif_mux = 1; @@ -4600,7 +4600,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4650,7 +4650,7 @@ static int patch_stac92hd95(struct hda_codec *codec) codec->epss = 0; /* longer delay needed for D3 */ spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; @@ -4692,7 +4692,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return err; spec = codec->spec; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->linear_tone_beep = 0; spec->gen.own_eapd_ctl = 1; spec->gen.power_down_unused = 1; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index d5d1dca4f11b..485663bb9101 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -133,7 +133,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; spec->gen.add_stereo_mix_input = HDA_HINT_STEREO_MIX_AUTO; - codec->power_mgmt = 1; + codec->power_save_node = 1; spec->gen.power_down_unused = 1; return spec; } @@ -236,7 +236,7 @@ static int via_pin_power_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = codec->power_mgmt; + ucontrol->value.enumerated.item[0] = codec->power_save_node; return 0; } @@ -247,9 +247,9 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol, struct via_spec *spec = codec->spec; bool val = !!ucontrol->value.enumerated.item[0]; - if (val == codec->power_mgmt) + if (val == codec->power_save_node) return 0; - codec->power_mgmt = val; + codec->power_save_node = val; spec->gen.power_down_unused = val; analog_low_current_mode(codec); return 1; @@ -295,7 +295,7 @@ static void __analog_low_current_mode(struct hda_codec *codec, bool force) bool enable; unsigned int verb, parm; - if (!codec->power_mgmt) + if (!codec->power_save_node) enable = false; else enable = is_aa_path_mute(codec) && !spec->gen.active_streams; @@ -517,7 +517,7 @@ static int via_parse_auto_config(struct hda_codec *codec) return err; /* disable widget PM at start for compatibility */ - codec->power_mgmt = 0; + codec->power_save_node = 0; spec->gen.power_down_unused = 0; return 0; }