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linux/sound/soc/fsl/fsl-asoc-card.c
Kuninori Morimoto 9998d3e2ad
ASoC: fsl: fsl-asoc-card: consider CPU-Platform possibility 
commit e57a4c2f15 ("ASoC: fsl: fsl-asoc-card: don't select unnecessary
Platform")

Current ALSA SoC avoid to add duplicate component to rtd,
and this driver was selecting CPU component as Platform component.
Thus, above patch removed Platform settings from this driver,
because it assumed these are same component.

But, some CPU driver is using generic DMAEngine, in such case, both
CPU component and Platform component will have same of_node/name.
In other words, there are some components which are different but
have same of_node/name.

In such case, Card driver definitely need to select Platform even
though it is same as CPU.
It is depends on CPU driver, but is difficult to know it from Card driver.
This patch reverts above patch.

Fixes: commit e57a4c2f15 ("ASoC: fsl: fsl-asoc-card: don't select unnecessary Platform")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2019-06-28 15:11:17 +01:00

739 lines
21 KiB
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// SPDX-License-Identifier: GPL-2.0
//
// Freescale Generic ASoC Sound Card driver with ASRC
//
// Copyright (C) 2014 Freescale Semiconductor, Inc.
//
// Author: Nicolin Chen <nicoleotsuka@gmail.com>
#include <linux/clk.h>
#include <linux/i2c.h>
#include <linux/module.h>
#include <linux/of_platform.h>
#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
#include <sound/ac97_codec.h>
#endif
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "fsl_esai.h"
#include "fsl_sai.h"
#include "imx-audmux.h"
#include "../codecs/sgtl5000.h"
#include "../codecs/wm8962.h"
#include "../codecs/wm8960.h"
#define CS427x_SYSCLK_MCLK 0
#define RX 0
#define TX 1
/* Default DAI format without Master and Slave flag */
#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
/**
* CODEC private data
*
* @mclk_freq: Clock rate of MCLK
* @mclk_id: MCLK (or main clock) id for set_sysclk()
* @fll_id: FLL (or secordary clock) id for set_sysclk()
* @pll_id: PLL id for set_pll()
*/
struct codec_priv {
unsigned long mclk_freq;
u32 mclk_id;
u32 fll_id;
u32 pll_id;
};
/**
* CPU private data
*
* @sysclk_freq[2]: SYSCLK rates for set_sysclk()
* @sysclk_dir[2]: SYSCLK directions for set_sysclk()
* @sysclk_id[2]: SYSCLK ids for set_sysclk()
* @slot_width: Slot width of each frame
*
* Note: [1] for tx and [0] for rx
*/
struct cpu_priv {
unsigned long sysclk_freq[2];
u32 sysclk_dir[2];
u32 sysclk_id[2];
u32 slot_width;
};
/**
* Freescale Generic ASOC card private data
*
* @dai_link[3]: DAI link structure including normal one and DPCM link
* @pdev: platform device pointer
* @codec_priv: CODEC private data
* @cpu_priv: CPU private data
* @card: ASoC card structure
* @sample_rate: Current sample rate
* @sample_format: Current sample format
* @asrc_rate: ASRC sample rate used by Back-Ends
* @asrc_format: ASRC sample format used by Back-Ends
* @dai_fmt: DAI format between CPU and CODEC
* @name: Card name
*/
struct fsl_asoc_card_priv {
struct snd_soc_dai_link dai_link[3];
struct platform_device *pdev;
struct codec_priv codec_priv;
struct cpu_priv cpu_priv;
struct snd_soc_card card;
u32 sample_rate;
snd_pcm_format_t sample_format;
u32 asrc_rate;
snd_pcm_format_t asrc_format;
u32 dai_fmt;
char name[32];
};
/**
* This dapm route map exsits for DPCM link only.
* The other routes shall go through Device Tree.
*
* Note: keep all ASRC routes in the second half
* to drop them easily for non-ASRC cases.
*/
static const struct snd_soc_dapm_route audio_map[] = {
/* 1st half -- Normal DAPM routes */
{"Playback", NULL, "CPU-Playback"},
{"CPU-Capture", NULL, "Capture"},
/* 2nd half -- ASRC DAPM routes */
{"CPU-Playback", NULL, "ASRC-Playback"},
{"ASRC-Capture", NULL, "CPU-Capture"},
};
static const struct snd_soc_dapm_route audio_map_ac97[] = {
/* 1st half -- Normal DAPM routes */
{"Playback", NULL, "AC97 Playback"},
{"AC97 Capture", NULL, "Capture"},
/* 2nd half -- ASRC DAPM routes */
{"AC97 Playback", NULL, "ASRC-Playback"},
{"ASRC-Capture", NULL, "AC97 Capture"},
};
/* Add all possible widgets into here without being redundant */
static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
SND_SOC_DAPM_LINE("Line Out Jack", NULL),
SND_SOC_DAPM_LINE("Line In Jack", NULL),
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_MIC("AMIC", NULL),
SND_SOC_DAPM_MIC("DMIC", NULL),
};
static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv)
{
return priv->dai_fmt == SND_SOC_DAIFMT_AC97;
}
static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
struct cpu_priv *cpu_priv = &priv->cpu_priv;
struct device *dev = rtd->card->dev;
int ret;
priv->sample_rate = params_rate(params);
priv->sample_format = params_format(params);
/*
* If codec-dai is DAI Master and all configurations are already in the
* set_bias_level(), bypass the remaining settings in hw_params().
* Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS.
*/
if ((priv->card.set_bias_level &&
priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) ||
fsl_asoc_card_is_ac97(priv))
return 0;
/* Specific configurations of DAIs starts from here */
ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
cpu_priv->sysclk_freq[tx],
cpu_priv->sysclk_dir[tx]);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set sysclk for cpu dai\n");
return ret;
}
if (cpu_priv->slot_width) {
ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2,
cpu_priv->slot_width);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set TDM slot for cpu dai\n");
return ret;
}
}
return 0;
}
static const struct snd_soc_ops fsl_asoc_card_ops = {
.hw_params = fsl_asoc_card_hw_params,
};
static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
struct snd_interval *rate;
struct snd_mask *mask;
rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
rate->max = rate->min = priv->asrc_rate;
mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
snd_mask_none(mask);
snd_mask_set_format(mask, priv->asrc_format);
return 0;
}
SND_SOC_DAILINK_DEFS(hifi,
DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()));
SND_SOC_DAILINK_DEFS(hifi_fe,
DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_DUMMY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()));
SND_SOC_DAILINK_DEFS(hifi_be,
DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_EMPTY()),
DAILINK_COMP_ARRAY(COMP_DUMMY()));
static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
/* Default ASoC DAI Link*/
{
.name = "HiFi",
.stream_name = "HiFi",
.ops = &fsl_asoc_card_ops,
SND_SOC_DAILINK_REG(hifi),
},
/* DPCM Link between Front-End and Back-End (Optional) */
{
.name = "HiFi-ASRC-FE",
.stream_name = "HiFi-ASRC-FE",
.dpcm_playback = 1,
.dpcm_capture = 1,
.dynamic = 1,
SND_SOC_DAILINK_REG(hifi_fe),
},
{
.name = "HiFi-ASRC-BE",
.stream_name = "HiFi-ASRC-BE",
.be_hw_params_fixup = be_hw_params_fixup,
.ops = &fsl_asoc_card_ops,
.dpcm_playback = 1,
.dpcm_capture = 1,
.no_pcm = 1,
SND_SOC_DAILINK_REG(hifi_be),
},
};
static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_pcm_runtime *rtd;
struct snd_soc_dai *codec_dai;
struct codec_priv *codec_priv = &priv->codec_priv;
struct device *dev = card->dev;
unsigned int pll_out;
int ret;
rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
codec_dai = rtd->codec_dai;
if (dapm->dev != codec_dai->dev)
return 0;
switch (level) {
case SND_SOC_BIAS_PREPARE:
if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
break;
if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
pll_out = priv->sample_rate * 384;
else
pll_out = priv->sample_rate * 256;
ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
codec_priv->mclk_id,
codec_priv->mclk_freq, pll_out);
if (ret) {
dev_err(dev, "failed to start FLL: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
pll_out, SND_SOC_CLOCK_IN);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set SYSCLK: %d\n", ret);
return ret;
}
break;
case SND_SOC_BIAS_STANDBY:
if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
break;
ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
codec_priv->mclk_freq,
SND_SOC_CLOCK_IN);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to switch away from FLL: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
if (ret) {
dev_err(dev, "failed to stop FLL: %d\n", ret);
return ret;
}
break;
default:
break;
}
return 0;
}
static int fsl_asoc_card_audmux_init(struct device_node *np,
struct fsl_asoc_card_priv *priv)
{
struct device *dev = &priv->pdev->dev;
u32 int_ptcr = 0, ext_ptcr = 0;
int int_port, ext_port;
int ret;
ret = of_property_read_u32(np, "mux-int-port", &int_port);
if (ret) {
dev_err(dev, "mux-int-port missing or invalid\n");
return ret;
}
ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
if (ret) {
dev_err(dev, "mux-ext-port missing or invalid\n");
return ret;
}
/*
* The port numbering in the hardware manual starts at 1, while
* the AUDMUX API expects it starts at 0.
*/
int_port--;
ext_port--;
/*
* Use asynchronous mode (6 wires) for all cases except AC97.
* If only 4 wires are needed, just set SSI into
* synchronous mode and enable 4 PADs in IOMUX.
*/
switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_RFSDIR |
IMX_AUDMUX_V2_PTCR_RCLKDIR |
IMX_AUDMUX_V2_PTCR_TFSDIR |
IMX_AUDMUX_V2_PTCR_TCLKDIR;
break;
case SND_SOC_DAIFMT_CBM_CFS:
int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_RCLKDIR |
IMX_AUDMUX_V2_PTCR_TCLKDIR;
ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
IMX_AUDMUX_V2_PTCR_RFSDIR |
IMX_AUDMUX_V2_PTCR_TFSDIR;
break;
case SND_SOC_DAIFMT_CBS_CFM:
int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_RFSDIR |
IMX_AUDMUX_V2_PTCR_TFSDIR;
ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
IMX_AUDMUX_V2_PTCR_RCLKDIR |
IMX_AUDMUX_V2_PTCR_TCLKDIR;
break;
case SND_SOC_DAIFMT_CBS_CFS:
ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
IMX_AUDMUX_V2_PTCR_RFSDIR |
IMX_AUDMUX_V2_PTCR_RCLKDIR |
IMX_AUDMUX_V2_PTCR_TFSDIR |
IMX_AUDMUX_V2_PTCR_TCLKDIR;
break;
default:
if (!fsl_asoc_card_is_ac97(priv))
return -EINVAL;
}
if (fsl_asoc_card_is_ac97(priv)) {
int_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
IMX_AUDMUX_V2_PTCR_TCLKDIR;
ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN |
IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
IMX_AUDMUX_V2_PTCR_TFSDIR;
}
/* Asynchronous mode can not be set along with RCLKDIR */
if (!fsl_asoc_card_is_ac97(priv)) {
unsigned int pdcr =
IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port);
ret = imx_audmux_v2_configure_port(int_port, 0,
pdcr);
if (ret) {
dev_err(dev, "audmux internal port setup failed\n");
return ret;
}
}
ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
if (ret) {
dev_err(dev, "audmux internal port setup failed\n");
return ret;
}
if (!fsl_asoc_card_is_ac97(priv)) {
unsigned int pdcr =
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port);
ret = imx_audmux_v2_configure_port(ext_port, 0,
pdcr);
if (ret) {
dev_err(dev, "audmux external port setup failed\n");
return ret;
}
}
ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
if (ret) {
dev_err(dev, "audmux external port setup failed\n");
return ret;
}
return 0;
}
static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
{
struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
struct snd_soc_pcm_runtime *rtd = list_first_entry(
&card->rtd_list, struct snd_soc_pcm_runtime, list);
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct codec_priv *codec_priv = &priv->codec_priv;
struct device *dev = card->dev;
int ret;
if (fsl_asoc_card_is_ac97(priv)) {
#if IS_ENABLED(CONFIG_SND_AC97_CODEC)
struct snd_soc_component *component = rtd->codec_dai->component;
struct snd_ac97 *ac97 = snd_soc_component_get_drvdata(component);
/*
* Use slots 3/4 for S/PDIF so SSI won't try to enable
* other slots and send some samples there
* due to SLOTREQ bits for S/PDIF received from codec
*/
snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS,
AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4);
#endif
return 0;
}
ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
if (ret && ret != -ENOTSUPP) {
dev_err(dev, "failed to set sysclk in %s\n", __func__);
return ret;
}
return 0;
}
static int fsl_asoc_card_probe(struct platform_device *pdev)
{
struct device_node *cpu_np, *codec_np, *asrc_np;
struct device_node *np = pdev->dev.of_node;
struct platform_device *asrc_pdev = NULL;
struct platform_device *cpu_pdev;
struct fsl_asoc_card_priv *priv;
struct i2c_client *codec_dev;
const char *codec_dai_name;
u32 width;
int ret;
priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
cpu_np = of_parse_phandle(np, "audio-cpu", 0);
/* Give a chance to old DT binding */
if (!cpu_np)
cpu_np = of_parse_phandle(np, "ssi-controller", 0);
if (!cpu_np) {
dev_err(&pdev->dev, "CPU phandle missing or invalid\n");
ret = -EINVAL;
goto fail;
}
cpu_pdev = of_find_device_by_node(cpu_np);
if (!cpu_pdev) {
dev_err(&pdev->dev, "failed to find CPU DAI device\n");
ret = -EINVAL;
goto fail;
}
codec_np = of_parse_phandle(np, "audio-codec", 0);
if (codec_np)
codec_dev = of_find_i2c_device_by_node(codec_np);
else
codec_dev = NULL;
asrc_np = of_parse_phandle(np, "audio-asrc", 0);
if (asrc_np)
asrc_pdev = of_find_device_by_node(asrc_np);
/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
if (codec_dev) {
struct clk *codec_clk = clk_get(&codec_dev->dev, NULL);
if (!IS_ERR(codec_clk)) {
priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
clk_put(codec_clk);
}
}
/* Default sample rate and format, will be updated in hw_params() */
priv->sample_rate = 44100;
priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
/* Assign a default DAI format, and allow each card to overwrite it */
priv->dai_fmt = DAI_FMT_BASE;
/* Diversify the card configurations */
if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
codec_dai_name = "cs42888";
priv->card.set_bias_level = NULL;
priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
priv->cpu_priv.slot_width = 32;
priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
} else if (of_device_is_compatible(np, "fsl,imx-audio-cs427x")) {
codec_dai_name = "cs4271-hifi";
priv->codec_priv.mclk_id = CS427x_SYSCLK_MCLK;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
codec_dai_name = "sgtl5000";
priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
codec_dai_name = "wm8962";
priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
priv->codec_priv.pll_id = WM8962_FLL;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) {
codec_dai_name = "wm8960-hifi";
priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO;
priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO;
priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
} else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) {
codec_dai_name = "ac97-hifi";
priv->card.set_bias_level = NULL;
priv->dai_fmt = SND_SOC_DAIFMT_AC97;
} else {
dev_err(&pdev->dev, "unknown Device Tree compatible\n");
ret = -EINVAL;
goto asrc_fail;
}
if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) {
dev_err(&pdev->dev, "failed to find codec device\n");
ret = -EINVAL;
goto asrc_fail;
}
/* Common settings for corresponding Freescale CPU DAI driver */
if (of_node_name_eq(cpu_np, "ssi")) {
/* Only SSI needs to configure AUDMUX */
ret = fsl_asoc_card_audmux_init(np, priv);
if (ret) {
dev_err(&pdev->dev, "failed to init audmux\n");
goto asrc_fail;
}
} else if (of_node_name_eq(cpu_np, "esai")) {
priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
} else if (of_node_name_eq(cpu_np, "sai")) {
priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
}
snprintf(priv->name, sizeof(priv->name), "%s-audio",
fsl_asoc_card_is_ac97(priv) ? "ac97" :
codec_dev->name);
/* Initialize sound card */
priv->pdev = pdev;
priv->card.dev = &pdev->dev;
priv->card.name = priv->name;
priv->card.dai_link = priv->dai_link;
priv->card.dapm_routes = fsl_asoc_card_is_ac97(priv) ?
audio_map_ac97 : audio_map;
priv->card.late_probe = fsl_asoc_card_late_probe;
priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
/* Drop the second half of DAPM routes -- ASRC */
if (!asrc_pdev)
priv->card.num_dapm_routes /= 2;
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].cpus->of_node = cpu_np;
priv->dai_link[0].codecs->dai_name = codec_dai_name;
if (!fsl_asoc_card_is_ac97(priv))
priv->dai_link[0].codecs->of_node = codec_np;
else {
u32 idx;
ret = of_property_read_u32(cpu_np, "cell-index", &idx);
if (ret) {
dev_err(&pdev->dev,
"cannot get CPU index property\n");
goto asrc_fail;
}
priv->dai_link[0].codecs->name =
devm_kasprintf(&pdev->dev, GFP_KERNEL,
"ac97-codec.%u",
(unsigned int)idx);
if (!priv->dai_link[0].codecs->name) {
ret = -ENOMEM;
goto asrc_fail;
}
}
priv->dai_link[0].platforms->of_node = cpu_np;
priv->dai_link[0].dai_fmt = priv->dai_fmt;
priv->card.num_links = 1;
if (asrc_pdev) {
/* DPCM DAI Links only if ASRC exsits */
priv->dai_link[1].cpus->of_node = asrc_np;
priv->dai_link[1].platforms->of_node = asrc_np;
priv->dai_link[2].codecs->dai_name = codec_dai_name;
priv->dai_link[2].codecs->of_node = codec_np;
priv->dai_link[2].codecs->name =
priv->dai_link[0].codecs->name;
priv->dai_link[2].cpus->of_node = cpu_np;
priv->dai_link[2].dai_fmt = priv->dai_fmt;
priv->card.num_links = 3;
ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
&priv->asrc_rate);
if (ret) {
dev_err(&pdev->dev, "failed to get output rate\n");
ret = -EINVAL;
goto asrc_fail;
}
ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
if (ret) {
dev_err(&pdev->dev, "failed to get output rate\n");
ret = -EINVAL;
goto asrc_fail;
}
if (width == 24)
priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
else
priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
}
/* Finish card registering */
platform_set_drvdata(pdev, priv);
snd_soc_card_set_drvdata(&priv->card, priv);
ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
if (ret && ret != -EPROBE_DEFER)
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
asrc_fail:
of_node_put(asrc_np);
of_node_put(codec_np);
put_device(&cpu_pdev->dev);
fail:
of_node_put(cpu_np);
return ret;
}
static const struct of_device_id fsl_asoc_card_dt_ids[] = {
{ .compatible = "fsl,imx-audio-ac97", },
{ .compatible = "fsl,imx-audio-cs42888", },
{ .compatible = "fsl,imx-audio-cs427x", },
{ .compatible = "fsl,imx-audio-sgtl5000", },
{ .compatible = "fsl,imx-audio-wm8962", },
{ .compatible = "fsl,imx-audio-wm8960", },
{}
};
MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids);
static struct platform_driver fsl_asoc_card_driver = {
.probe = fsl_asoc_card_probe,
.driver = {
.name = "fsl-asoc-card",
.pm = &snd_soc_pm_ops,
.of_match_table = fsl_asoc_card_dt_ids,
},
};
module_platform_driver(fsl_asoc_card_driver);
MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
MODULE_ALIAS("platform:fsl-asoc-card");
MODULE_LICENSE("GPL");
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