max98090 spec states that chip needs to be in turned-on state to supply
mic bias. Enable SHDN dapm widget along with MICBIAS widget to
actually turn on mic bias for proper headset button detection.
This is similar to cht_ti_jack_event in
sound/soc/intel/boards/cht_bsw_max98090_ti.c.
Note that due to ts3a227e reports the jack event right away before the
notifier is registered, if headset is plugged on boot, headset button
will not get detected until headset is unplugged and plugged. This is
still an issue to be fixed.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20191028095229.99438-1-cychiang@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
When non standard names are used it is possible that one of the directions
are not provided, thus the flags needs to be present to tell the core that
we have half duplex setup.
Fixes: 642aafea88 ("ASoC: ti: remove compat dma probing")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20191028115207.5142-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the implementation of sof_set_get_large_ctrl_data() there is a memory
leak in case an error. Release partdata if sof_get_ctrl_copy_params()
fails.
Fixes: 54d198d501 ("ASoC: SOF: Propagate sof_get_ctrl_copy_params() error properly")
Signed-off-by: Navid Emamdoost <navid.emamdoost@gmail.com>
Link: https://lore.kernel.org/r/20191027215330.12729-1-navid.emamdoost@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In the implementation of sof_dfsentry_write() memory allocated for
string is leaked in case of an error. Go to error handling path if the
d_name.name is not valid.
Fixes: 091c12e1f5 ("ASoC: SOF: debug: add new debugfs entries for IPC flood test")
Signed-off-by: Navid Emamdoost <navid.emamdoost@gmail.com>
Link: https://lore.kernel.org/r/20191027194856.4056-1-navid.emamdoost@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We are missing the 'CONFIG_' prefix when using the kernel configure item
SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1, here correct them.
Fixes: 43b2ab9009 ('ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture')
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191025221538.6668-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This is an alternative fix attemp for the issue reported in the commit
caa8422d01 ("ALSA: hda: Flush interrupts on disabling") that was
reverted later due to regressions. Instead of tweaking the hardware
disablement order and the enforced irq flushing, do calling
cancel_work_sync() of the unsol work early enough, and explicitly
ignore the unsol events during the shutdown by checking the
bus->shutdown flag.
Fixes: caa8422d01 ("ALSA: hda: Flush interrupts on disabling")
Cc: Chris Wilson <chris@chris-wilson.co.uk>
Link: https://lore.kernel.org/r/s5h1ruxt9cz.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit caa8422d01.
It turned out that this commit caused a regression at shutdown /
reboot, as the synchronize_irq() calls seems blocking the whole
shutdown. Also another part of the change about shuffling the call
order looks suspicious; the azx_stop_chip() call disables the CORB /
RIRB while the others may still need the CORB/RIRB update.
Since the original commit itself was a cargo-fix, let's revert the
whole patch.
Fixes: caa8422d01 ("ALSA: hda: Flush interrupts on disabling")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205333
BugLinK: https://bugs.freedesktop.org/show_bug.cgi?id=111174
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: Chris Wilson <chris@chris-wilson.co.uk>
Link: https://lore.kernel.org/r/20191028081056.22010-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A helper function of ALSA bebob driver returns negative value in a
function which has a prototype to return unsigned value.
This commit fixes it by changing the prototype.
Fixes: eb7b3a056c ("ALSA: bebob: Add commands and connections/streams management")
Cc: <stable@vger.kernel.org> # v3.16+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191026030620.12077-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
platform_get_irq() will call dev_err() itself on failure,
so there is no need for the driver to also do this.
This is detected by coccinelle.
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20191025093905.14888-1-yuehaibing@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current position update is not precise enough for PulseAudio to
work reliably with the timer-based scheduling on Baytrail,
Cherrytrail, Broadwell.
Disable the NO_PERIOD_WAKEUP capability and use BATCH to signal that
the position is only reliable and updated during period_elapsed
events.
This will be reverted when the firmware provides a more accurate
position for those platforms.
Reviewed-by: Jaska Uimonen <jaska.uimonen@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191024210318.30068-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the INFO_ flags such as PAUSE/NO_PERIOD_WAKEUP are defined
in the SOF PCM core, which doesn't scale. To account for platform
variations, these flags need to be set in DSP ops.
This patch only moves the definitions and does not change any
functionality.
Reviewed-by: Jaska Uimonen <jaska.uimonen@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191024210318.30068-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We see odd FIFO overruns with this, we assume the best thing to do is
to disable the RX I2S frontend first, and then disable the FIFO that
is using it.
This also fixes an issue where using multi-word frames (TDM) have
partial samples stuck in the FIFO which then get read out when the
next capture is started.
Signed-off-by: Ben Dooks <ben.dooks@codethink.co.uk>
Acked-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20191018154833.7560-5-ben.dooks@codethink.co.uk
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a callback to configure TDM settings for the Tegra30 I2S ASoC 'platform'
driver.
Signed-off-by: Edward Cragg <edward.cragg@codethink.co.uk>
[ben.dooks@codethink.co.uk: merge fix for power management]
[ben.dooks@codethink.co.uk: add review change for fsync of 1 clock]
Signed-off-by: Ben Dooks <ben.dooks@codethink.co.uk>
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20191018154833.7560-2-ben.dooks@codethink.co.uk
Signed-off-by: Mark Brown <broonie@kernel.org>
The tegra3 audio can support 24 and 32 bit sample sizes so add the
option to the tegra30_i2s_hw_params to configure the S24_LE or S32_LE
formats when requested.
Signed-off-by: Edward Cragg <edward.cragg@codethink.co.uk>
[ben.dooks@codethink.co.uk: fixup merge of 24 and 32bit]
[ben.dooks@codethink.co.uk: add pm calls around ytdm config]
[ben.dooks@codethink.co.uk: drop debug printing to dev_dbg]
Signed-off-by: Ben Dooks <ben.dooks@codethink.co.uk>
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20191018154833.7560-3-ben.dooks@codethink.co.uk
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix bugs reported by kbuild test robot
Fixes: a0e0d13542 ("ASoC: rt5677: Add a PCM device for streaming hotword via SPI")
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191024184026.183913-1-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The allocation of rt5677_dsp can potentially fail and return null, so add
a null check and return -ENOMEM on a memory allocation failure.
Addresses-Coverity: ("Dereference null return")
Fixes: a0e0d13542 ("ASoC: rt5677: Add a PCM device for streaming hotword via SPI")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20191024124610.18182-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
These 2 ThinkCentres installed a new realtek codec ID 0x623,
it has 2 front mics with the same location on pin 0x18 and 0x19.
Apply fixup ALC283_FIXUP_HEADSET_MIC to change 1 front mic
location to right, then pulseaudio can handle them.
One "Front Mic" and one "Mic" will be shown, and audio output works
fine.
Signed-off-by: Aaron Ma <aaron.ma@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191024114439.31522-1-aaron.ma@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add platform specific data for Kahlee project.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Shirish S <shirish.s@amd.com>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191023212948.92246-1-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The devm conversion of kirkwood was incorrect; on removal, devm takes
effect after the "remove" function has returned. So, the effect of
the conversion was to change the order during remove from:
- snd_soc_unregister_component() (unpublishes interfaces)
- clk_disable_unprepare()
- cleanup resources
After the conversion, this became:
- clk_disable_unprepare() - while the device may still be active
- snd_soc_unregister_component()
- cleanup resources
Hence, it introduces a bug, where the internal clock for the device
may be shut down before the device itself has been shut down. It is
known that Marvell SoCs, including Dove, locks up if registers for a
peripheral that has its clocks disabled are accessed.
Fixes: f98fc0f815 ("ASoC: kirkwood: replace platform to component")
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Link: https://lore.kernel.org/r/E1iNGyP-0004oN-BA@rmk-PC.armlinux.org.uk
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently each SSI unit's busif dma address is calculated by
following calculation formula:
0xec540000 + 0x1000 * id + busif / 4 * 0xA000 + busif % 4 * 0x400
But according to R-Car3 HW manual 41.1.4 Register Configuration,
ssi9 4/5/6/7 busif data register address
(SSI9_4_BUSIF/SSI9_5_BUSIF/SSI9_6_BUSIF/SSI9_7_BUSIF)
are out of this rule.
This patch updates the calculation formula to correct
ssi9 4/5/6/7 busif data register address.
Fixes: 5e45a6fab3 ("ASoc: rsnd: dma: Calculate dma address with consider of BUSIF")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
[erosca: minor improvements in commit description]
Cc: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Cc: stable@vger.kernel.org # v4.20+
Signed-off-by: Eugeniu Rosca <erosca@de.adit-jv.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20191022185429.12769-1-erosca@de.adit-jv.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Kconfig, PCI ID and chip info for JSL platform.
The DSP only has 2 cores for this platform.
Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191022194705.23347-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Such traces should be extremely rare but extremely useful for debug.
Report errors for all calls to sdn_sof_dsp_read_poll_timeout(), but
only on negative values for consistency.
Add traces that enable each timeout to be uniquely identified.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191022192844.21022-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If a ROM timeout is detected, we still stop the DMA but will return
the initial error should the DMA stop also fail.
Likewise the cleanup is handled regardless of the status, but we
return the initial error.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191022192844.21022-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A=y,
below errors can be seen:
sound/soc/codecs/cros_ec_codec.o: In function `send_ec_host_command':
cros_ec_codec.c:(.text+0x534): undefined reference to `cros_ec_cmd_xfer_status'
cros_ec_codec.c:(.text+0x101c): undefined reference to `cros_ec_get_host_event'
This is because it will select SND_SOC_CROS_EC_CODEC
after commit 2cc3cd5fdc ("ASoC: mediatek: mt8183: support WoV"),
but SND_SOC_CROS_EC_CODEC depends on CROS_EC.
Fixes: 2cc3cd5fdc ("ASoC: mediatek: mt8183: support WoV")
Signed-off-by: Mao Wenan <maowenan@huawei.com>
Link: https://lore.kernel.org/r/20191023063103.44941-1-maowenan@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds missing of_node_put() for
rsnd_parse_tdm_split_mode()
rsnd_parse_connect_graph()
Reported-by: Pavel Machek <pavel@denx.de>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8736fkyzx8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
According to R-Car3 HW manual 40.3.3 (Data Format on Audio Local Bus),
in case of monaural data writing or reading through Audio-DMAC,
it's always in Left Justified format, so both src and dst
DMA Bus width should be equal to physical data width.
Therefore set src and dst's DMA bus width to:
- [monaural case] data width
- [non-monaural case] 32bits (as prior applying the patch)
Cc: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Cc: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Eugeniu Rosca <erosca@de.adit-jv.com>
Link: https://lore.kernel.org/r/20191022185518.12838-1-erosca@de.adit-jv.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC has for_each_rtdcom() which is link list for
rtd-component which is called as rtdcom. The relationship image is like below
rtdcom rtdcom rtdcom
component component component
rtd->component_list -> list -> list -> list ...
Here, the pointer get via normal link list is rtdcom,
Thus, current for_each loop is like below, and need to get
component via rtdcom->component
for_each_rtdcom(rtd, rtdcom) {
component = rtdcom->component;
...
}
but usually, user want to get pointer from for_each_xxx is component
directly, like below.
for_each_rtd_component(rtd, rtdcom, component) {
...
}
This patch expands list_for_each_entry manually, and enable to get
component directly from for_each macro.
Because of it, the macro becoming difficult to read,
but macro itself becoming useful.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/878spm64m4.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When our call to get the external clock fails, we forget to clean up
the enabled internal clock correctly. Enable the clock after we have
obtained all our resources.
Fixes: 84aac6c79b ("ASoC: kirkwood: fix loss of external clock at probe time")
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Link: https://lore.kernel.org/r/E1iNGyK-0004oF-6A@rmk-PC.armlinux.org.uk
Signed-off-by: Mark Brown <broonie@kernel.org>
Reshuffle list of devices by historical order and add correct
information as needed.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20191022174313.29087-2-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For distributions, we need one place where we can decide
which driver will be activated for the auto-configation of the
Intel's HDA hardware with DSP. Actually, we cover three drivers:
* Legacy HDA
* Intel SST
* Intel Sound Open Firmware (SOF)
All those drivers registers similar PCI IDs, so the first
driver probed from the PCI stack can win. But... it is not
guaranteed that the correct driver wins.
This commit changes Intel's NHLT ACPI module to a common
DSP probe module for the Intel's hardware. All above sound
drivers calls this code. The user can force another behaviour
using the module parameter 'dsp_driver' located in
the 'snd-intel-dspcfg' module.
This change allows to add specific dmi checks for the specific
systems. The examples are taken from the pull request:
https://github.com/thesofproject/linux/pull/927
Tested on Lenovo Carbon X1 7th gen.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191022174313.29087-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
soc_remove_link_components() will be called from
soc_cleanup_card_resources(). This patch removes duplicate call.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877e5664lz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
dpcm_prune_paths() is checking widget at 2 parts.
(A) is for CPU, (B) is for Codec.
If we focus to (A) part, continue at (a) is for (1) loop. But,
if we focus to (B) part, continue at (b) is for (2) loop, not for (1).
This is bug.
This patch fixup this issue.
static int dpcm_prune_paths(...)
{
...
(1) for_each_dpcm_be(fe, stream, dpcm) {
...
^ widget = dai_get_widget(...);
|
(A) if (widget && widget_in_list(...))
| (a) continue;
v
^ (2) for_each_rtd_codec_dai(...) {
| widget = dai_get_widget(...);
(B)
| if (widget && widget_in_list(...))
v (b) continue;
}
...
Fixes: 2e5894d737 ("ASoC: pcm: Add support for DAI multicodec")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87blui64mf.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch implements a PCM interface for streaming hotword
phrases over SPI. Userspace can open the PCM device at anytime.
The stream is blocked when no hotword is detected. The mic
audio buffer on the DSP is a ~128KByte ring buffer that holds
~4sec of audio samples recorded from the DMIC (S16_LE, mono,
16KHz). After a hotword is detected, previous 2 seconds of audio
(containing the detected hotword) is streamed first, then live
capture continues until userspace closes the PCM stream.
When transferring, copy one period at a time then call
snd_pcm_period_elapsed(). This reduces the latency of transferring
the initial ~2sec of audio after hotword detect since audio samples
are available for userspace earlier.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191018200449.141123-2-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The recently introduced USB-audio descriptor validator had a stupid
copy&paste error that may lead to an unexpected overlook of too short
descriptors for processing and extension units. It's likely the cause
of the report triggered by syzkaller fuzzer. Let's fix it.
Fixes: 57f8770620 ("ALSA: usb-audio: More validations of descriptor units")
Reported-by: syzbot+0620f79a1978b1133fd7@syzkaller.appspotmail.com
Link: https://lore.kernel.org/r/s5hsgnkdbsl.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nvidia proprietary driver doesn't support runtime power management, so
when a user only wants to use the integrated GPU, it's a common practice
to let dGPU not to bind any driver, and let its upstream port to be
runtime suspended. At the end of runtime suspension the port uses
platform power management to disable power through _OFF method of power
resource, which is listed by _PR3.
After commit b516ea586d ("PCI: Enable NVIDIA HDA controllers"), when
the dGPU comes with an HDA function, the HDA won't be suspended if the
dGPU is unbound, so the power resource can't be turned off by its
upstream port driver.
Commit 37a3a98ef6 ("ALSA: hda - Enable runtime PM only for
discrete GPU") only allows HDA to be runtime suspended once GPU is
bound, to keep APU's HDA working.
However, HDA on dGPU isn't that useful if dGPU is not bound to any
driver. So let's relax the runtime suspend requirement for dGPU's HDA
function, to disable the power source to save lots of power.
BugLink: https://bugs.launchpad.net/bugs/1840835
Fixes: b516ea586d ("PCI: Enable NVIDIA HDA controllers")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20191018073848.14590-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
1. Get EC codec's capabilities.
2. Get and set SHM address if any.
3. Transmit language model to EC codec if needed.
4. Start to read audio data from EC codec if receives host event.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Acked-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Link: https://lore.kernel.org/r/20191019143504.1.I5388b69a7a9c551078fed216a77440cee6dedf49@changeid
Signed-off-by: Mark Brown <broonie@kernel.org>
PM8916 supports an earpiece as another (small) speaker.
The earpiece is routed through RX MIX1 similarly to
the headphones, except that RDAC2 MUX is set to RX1.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20191020153007.206070-2-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Parameter fragments and fragment_size are type of u32. U32_MAX is
the correct check.
Signed-off-by: Xiaojun Sang <xsang@codeaurora.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20191021095432.5639-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the PM8916 Hardware Register Description,
CDC_D_CDC_CONN_HPHR_DAC_CTL has only a single bit (RX_SEL)
to switch between RX1 (0) and RX2 (1). It is not possible to
disable it entirely to achieve the "ZERO" state.
However, at the moment the "RDAC2 MUX" mixer defines three possible
values ("ZERO", "RX2" and "RX1"). Setting the mixer to "ZERO"
actually configures it to RX1. Setting the mixer to "RX1" has
(seemingly) no effect.
Remove "ZERO" and replace it with "RX1" to fix this.
Fixes: 585e881e5b ("ASoC: codecs: Add msm8916-wcd analog codec")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20191020153007.206070-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
A collection of fixes that have arrived since the merge window. There
are a small number of core fixes here but they are smaller ones around
error handling.
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Merge tag 'asoc-fix-v5.4-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.4
A collection of fixes that have arrived since the merge window. There
are a small number of core fixes here but they are smaller ones around
error handling.
Some devices have a quirk to postpone transmission of isoc packet for
several dozen or hundred isoc cycles since configured to transmit.
Furthermore, some devices have a quirk to transmit isoc packet with
discontinued data of its header.
In 1394 OHCI specification, software allows to start isoc context with
certain isoc cycle. Linux firewire subsystem has kernel API to use it
as well.
This commit uses the functionality of 1394 OHCI controller to handle
the quirks. At present, this feature is convenient to ALSA bebob and
fireface driver. As a result, some devices can be safely handled, as
long as I know:
- MAudio FireWire solo
- MAudio ProFire Lightbridge
- MAudio FireWire 410
- Roland FA-66
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191018061911.24909-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit changes AMDTP domain to run on an IT context of 1394 OHCI as
IRQ target. No hardware interrupt is scheduled for the other isoc
contexts. All of the isoc context are processed in a callback for an isoc
context of IRQ target.
The IRQ target is automatically selected from a list of AMDTP streams,
thus users of AMDTP domain should add an AMDTP stream for IT context
at least.
The reason to select IT context as IRQ target is that the IT context
runs on local 1394 OHCI controller and it can be used as reliable,
constant IRQ generator. On the other hand, IR context can include skip
cycle according to isoc packet transferred by device.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191018061911.24909-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The aim of AMDTP domain is to process several isoc context in the same
time. However, current implementation is against this idea because it
flushes each isoc context in the end of processing context callback.
This commit cancels it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191018061911.24909-5-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An isoc context for AMDTP stream is flushed to queue packet
by a call of pcm.ack. This commit extends this for AMDTP
domain.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191018061911.24909-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An isoc context for AMDTP stream is flushed to queue packet
by a call of pcm.pointer. This commit extends this for AMDTP
domain.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191018061911.24909-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to handle several IR/IT contexts in the same
domain by tasklet context for one of the IT context. Such IT context is
stored to AMDTP domain structure as 'irq_target'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191018061911.24909-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Read max DMIC gain from EC codec instead of DTS. Also removes the
dt-binding of max-dmic-gain.
Acked-by: Rob Herring <robh@kernel.org>
Acked-by: Benson Leung <bleung@chromium.org>
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20191017213539.05.Id4657c864d544634f2b5c1c9b34fa8232ecba44d@changeid
Signed-off-by: Mark Brown <broonie@kernel.org>
aic32x4_set_dai_sysclk misses a check for devm_clk_get and may miss the
failure.
Add a check to fix it.
Signed-off-by: Chuhong Yuan <hslester96@gmail.com>
Link: https://lore.kernel.org/r/20191018081448.8486-1-hslester96@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Refactor by the following items:
- reformat copyright declaration
- use more specific name "i2s rx"
- use verbose symbol names to separate namespaces
- make some short functions inline
- remove unused TDM-related code
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Acked-By: Benson Leung <bleung@chromium.org>
Link: https://lore.kernel.org/r/20191014180059.02.I43373b9a66dbb70196b3f216b3aa86111c410836@changeid
Signed-off-by: Mark Brown <broonie@kernel.org>
All the kcontrol put() functions are currently returning 0 when
successful. This does not go well with alsamixer as it does
not seem to get notified on SND_CTL_EVENT_MASK_VALUE callbacks
when values change for (some of) the sof kcontrols.
This patch fixes that by returning true for volume, switch
and enum type kcontrols when values do change in put().
Signed-off-by: Dragos Tarcatu <dragos_tarcatu@mentor.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191018123806.18063-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit below, adds a call to sysclk callback on shutdown.
This introduces a regression in stm32 SAI driver, as some clock
services are called twice, leading to unbalanced calls.
Move processing related to mclk from shutdown to sysclk callback.
When requested frequency is 0, assume shutdown and release mclk.
Fixes: 2458adb8f9 ("SoC: simple-card-utils: set 0Hz to sysclk when shutdown")
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/20191018082040.31022-1-olivier.moysan@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
sof_audio_probe misses a check for devm_clk_get and may cause problems.
Add a check for it to fix the bug.
Signed-off-by: Chuhong Yuan <hslester96@gmail.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191017025044.31474-1-hslester96@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Unlike other format-related DAI parameters, rdai->bit_clk_inv flag
is not properly re-initialized when setting format for new stream
processing. The inversion, if requested, is then applied not to default,
but to a previous value, which leads to SCKP bit in SSICR register being
set incorrectly.
Fix this by re-setting the flag to its initial value, determined by format.
Fixes: 1a7889ca8a ("ASoC: rsnd: fixup SND_SOC_DAIFMT_xB_xF behavior")
Cc: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Cc: Jiada Wang <jiada_wang@mentor.com>
Cc: Timo Wischer <twischer@de.adit-jv.com>
Cc: stable@vger.kernel.org # v3.17+
Signed-off-by: Junya Monden <jmonden@jp.adit-jv.com>
Signed-off-by: Eugeniu Rosca <erosca@de.adit-jv.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20191016124255.7442-1-erosca@de.adit-jv.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In IEC 61883-1/6, one isoc packet can transfer events up to the value of
syt interval. This comes from the interval of isoc cycle. As 1394 OHCI
controller can generate hardware IRQ per isoc packet, the interval is
calculated as 125 usec.
In IEC 61883-1/6, two ways of transmission is described; blocking and
non-blocking methods. In blocking method, the sequence of packet includes
'empty' or 'NODATA' packets which include no events. In non-blocking
method, the number of events per packet is variable up to the syt
interval.
This commit uses double of the value of syt interval as minimum available
size of PCM period due to the above protocol design.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-13-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Linux driver for 1394 OHCI controller voluntarily flushes isoc context
when total size of accumulated context header reached PAGE_SIZE. This
kicks tasklet for the isoc context. This is inconvenient to process
runtime of PCM substream.
This commit adds a restriction of the maximum size of PCM period to
avoid this behaviour.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-12-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA IEC 61883-1/6 packet streaming engine controls 1394 OHCI controller
to generate hardware IRQ for fixed number of isochronous packets (=16)
since its first commit.
This commit allow the engine to generate it for variable period according
to the number of event to handle. For outgoing stream, internal
calculator is used to check the accumulated events. For incoming stream,
the number of data block in the packet of stream is used to check the
accumulated events. When it's unavailable, fixed number of packet
roughly calculated in advance is used instead of event counting.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-11-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows ALSA fireface driver to share PCM buffer size for
both capture and playback PCM substream. When AMDTP domain starts for
one of the PCM substream, buffer size of the PCM substream is stores to
AMDTP domain structure. Some AMDTP streams have already run with the
buffer size when another PCM substream starts, therefore the PCM
substream has a constraint to its buffer size.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-10-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows ALSA firewire-motu driver to share PCM buffer
size for both capture and playback PCM substream. When AMDTP domain
starts for one of the PCM substream, buffer size of the PCM substream
is stores to AMDTP domain structure. Some AMDTP streams have already
run with the buffer size when another PCM substream starts, therefore
the PCM substream has a constraint to its buffer size.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-9-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows ALSA firewire-tascam driver to share PCM buffer
size for both capture and playback PCM substream. When AMDTP domain
starts for one of the PCM substream, buffer size of the PCM substream
is stores to AMDTP domain structure. Some AMDTP streams have already
run with the buffer size when another PCM substream starts, therefore
the PCM substream has a constraint to its buffer size.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-8-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows ALSA firewire-digi00x driver to share PCM buffer
size for both capture and playback PCM substream. When AMDTP domain
starts for one of the PCM substream, buffer size of the PCM substream
is stores to AMDTP domain structure. Some AMDTP streams have already
run with the buffer size when another PCM substream starts, therefore
the PCM substream has a constraint to its buffer size.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows ALSA dice driver to share PCM buffer size for both
capture and playback PCM substream. When AMDTP domain starts for one
of the PCM substream, buffer size of the PCM substream is stores to
AMDTP domain structure. Some AMDTP streams have already run with the
buffer size when another PCM substream starts, therefore the PCM
substream has a constraint to its buffer size.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows ALSA oxfw driver to share PCM buffer size for both
capture and playback PCM substream. When AMDTP domain starts for one
of the PCM substream, buffer size of the PCM substream is stores to
AMDTP domain structure. Some AMDTP streams have already run with the
buffer size when another PCM substream starts, therefore the PCM
substream has a constraint to its buffer size.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-5-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows ALSA fireworks driver to share PCM buffer size for
both capture and playback PCM substream. When AMDTP domain starts for
one of the PCM substream, buffer size of the PCM substream is stores
to AMDTP domain structure. Some AMDTP streams have already run with the
buffer size when another PCM substream starts, therefore the PCM
substream has a constraint to its buffer size.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit allows ALSA bebob driver to share PCM buffer size for both
capture and playback PCM substream. When AMDTP domain starts for one
of the PCM substream, buffer size of the PCM substream is stores to
AMDTP domain structure. Some AMDTP streams have already run with the
buffer size when another PCM substream starts, therefore the PCM
substream has a constraint to its buffer size.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The number of packets in packet buffer has been fixed number (=48) since
first commit of ALSA IEC 61883-1/6 packet streaming engine.
This commit allows the engine to use variable number of packets in the
buffer. The size is calculated by a parameter in AMDTP domain structure
surely to store the number of events in the packets of buffer. Although
the value of parameter is expected to come from 'period size' parameter
of PCM substream, at present 48 is still used.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191017155424.885-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Przemysław Kopa reports that since commit b516ea586d ("PCI: Enable
NVIDIA HDA controllers"), the discrete GPU Nvidia GeForce GT 540M on his
2011 Samsung laptop refuses to runtime suspend, resulting in a power
regression and excessive heat.
Rivera Valdez witnesses the same issue with a GeForce GT 525M (GF108M)
of the same era, as does another Arch Linux user named "R0AR" with a
more recent GeForce GTX 1050 Ti (GP107M).
The commit exposes the discrete GPU's HDA controller and all four codecs
on the controller do not set the CLKSTOP and EPSS bits in the Supported
Power States Response. They also do not set the PS-ClkStopOk bit in the
Get Power State Response. hda_codec_runtime_suspend() therefore does
not call snd_hdac_codec_link_down(), which prevents each codec and the
PCI device from runtime suspending.
The same issue is present on some AMD discrete GPUs and we addressed it
by forcing runtime PM despite the bits not being set, see commit
57cb54e53b ("ALSA: hda - Force to link down at runtime suspend on
ATI/AMD HDMI").
Do the same for Nvidia HDMI codecs.
Fixes: b516ea586d ("PCI: Enable NVIDIA HDA controllers")
Link: https://bbs.archlinux.org/viewtopic.php?pid=1865512
Link: https://bugs.freedesktop.org/show_bug.cgi?id=75985#c81
Reported-by: Przemysław Kopa <prymoo@gmail.com>
Reported-by: Rivera Valdez <riveravaldez@ysinembargo.com>
Signed-off-by: Lukas Wunner <lukas@wunner.de>
Cc: Daniel Drake <dan@reactivated.net>
Cc: stable@vger.kernel.org # v5.3+
Link: https://lore.kernel.org/r/3086bc75135c1e3567c5bc4f3cc4ff5cbf7a56c2.1571324194.git.lukas@wunner.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation, when opening a PCM substream, it's needed to
check whether the opposite PCM substream runs. This is to assign
effectual constraints (e.g. sampling rate) to opened PCM substream.
The number of PCM substreams on AMDTP streams in domain is recorded in
own structure. Usage of this count is an alternative of the above check.
This is better because the count is incremented in pcm.hw_params earlier
than pcm.trigger.
This commit replaces the check with the substream count and the value for
the size of PCM period. Unlike the other drivers in ALSA firewire stack,
no MIDI substream is multiplexed into AMDTP stream.
I note that Fireface AMDTP protocol has a quirk that tx stream includes
blank isochronous cycle. The packet for blank cycle is equivalent to
empty or NODATA packet in IEC 61883-6, thus the protocol is similar to
blocking transmission method of IEC 61883-6. On the other hand, rx
stream adopts non-blocking transmission method. Although the difference
of transmission method between tx/rx streams precisely brings different
timing for a certain amount of events due to their different calculation
for data blocks per packet, it's possible to approximate enough amount
of events mostly has the same timing. Actually current ALSA IEC 61883-1/6
engine uses large amount of data blocks for each hardware IRQ
(=16 packets).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-18-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation, when opening a PCM substream, it's needed to
check whether the opposite PCM substream runs. This is to assign
effectual constraints (e.g. sampling rate) to opened PCM substream.
The number of PCM substreams and MIDI substreams on AMDTP streams in
domain is recorded in own structure. Usage of this count is an
alternative of the above check. This is better because the count is
incremented in pcm.hw_params earlier than pcm.trigger.
This idea has one issue because it's incremented for MIDI substreams as
well. In current implementation, for a case that any MIDI substream run
and a PCM substream is going to start, PCM application to start the PCM
substream can decide hardware parameters by restart packet streaming.
Just checking the substream count can brings regression.
Now AMDTP domain structure has a member for the size of PCM period in
PCM substream which starts AMDTP streams in domain. When the value has
zero and the substream count is greater than 1, it means that any MIDI
substream starts AMDTP streams in domain. Usage of the value can resolve
the above issue.
This commit replaces the check with the substream count and the value for
the size of PCM period.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-17-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation, when opening a PCM substream, it's needed to
check whether the opposite PCM substream runs. This is to assign
effectual constraints (e.g. sampling rate) to opened PCM substream.
The number of PCM substreams on AMDTP streams in domain is recorded in
own structure. Usage of this count is an alternative of the above check.
This is better because the count is incremented in pcm.hw_params earlier
than pcm.trigger.
This commit replaces the check with the substream count and the value for
the size of PCM period. Unlike the other drivers in ALSA firewire stack,
no MIDI substream is multiplexed into AMDTP stream.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-16-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation, when opening a PCM substream, it's needed to
check whether the opposite PCM substream runs. This is to assign
effectual constraints (e.g. sampling rate) to opened PCM substream.
The number of PCM substreams and MIDI substreams on AMDTP streams in
domain is recorded in own structure. Usage of this count is an
alternative of the above check. This is better because the count is
incremented in pcm.hw_params earlier than pcm.trigger.
This idea has one issue because it's incremented for MIDI substreams as
well. In current implementation, for a case that any MIDI substream run
and a PCM substream is going to start, PCM application to start the PCM
substream can decide hardware parameters by restart packet streaming.
Just checking the substream count can brings regression.
Now AMDTP domain structure has a member for the size of PCM period in
PCM substream which starts AMDTP streams in domain. When the value has
zero and the substream count is greater than 1, it means that any MIDI
substream starts AMDTP streams in domain. Usage of the value can resolve
the above issue.
This commit replaces the check with the substream count and the value for
the size of PCM period.
I note that DOT AMDTP protocol has a quirk to use different transmission
method of IEC 61883-6 for tx/rx streams; non-blocking in tx stream and
blocking in rx stream. Although the difference of transmission method
between tx/rx streams precisely brings different timing for a certain
amount of events due to their different calculation for data blocks per
packet, it's possible to approximate enough amount of events mostly has
the same timing. Actually current ALSA IEC 61883-1/6 engine uses large
amount of data blocks for each hardware IRQ (=16 packets).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-15-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation, when opening a PCM substream, it's needed to
check whether the opposite PCM substream runs. This is to assign
effectual constraints (e.g. sampling rate) to opened PCM substream.
The number of PCM substreams and MIDI substreams on AMDTP streams in
domain is recorded in own structure. Usage of this count is an
alternative of the above check. This is better because the count is
incremented in pcm.hw_params earlier than pcm.trigger.
This idea has one issue because it's incremented for MIDI substreams as
well. In current implementation, for a case that any MIDI substream run
and a PCM substream is going to start, PCM application to start the PCM
substream can decide hardware parameters by restart packet streaming.
Just checking the substream count can brings regression.
Now AMDTP domain structure has a member for the size of PCM period in
PCM substream which starts AMDTP streams in domain. When the value has
zero and the substream count is greater than 1, it means that any MIDI
substream starts AMDTP streams in domain. Usage of the value can resolve
the above issue.
This commit replaces the check with the substream count and the value for
the size of PCM period.
Dice hardware has a quirk called as 'Dual Wire'. For a case of higher
sampling transmission frequency, this commit performs calculations between
the number of PCM frames and the number of events in AMDTP stream.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-14-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation, when opening a PCM substream, it's needed to
check whether the opposite PCM substream runs. This is to assign
effectual constraints (e.g. sampling rate) to opened PCM substream.
The number of PCM substreams and MIDI substreams on AMDTP streams in
domain is recorded in own structure. Usage of this count is an
alternative of the above check. This is better because the count is
incremented in pcm.hw_params earlier than pcm.trigger.
This idea has one issue because it's incremented for MIDI substreams as
well. In current implementation, for a case that any MIDI substream run
and a PCM substream is going to start, PCM application to start the PCM
substream can decide hardware parameters by restart packet streaming.
Just checking the substream count can brings regression.
Now AMDTP domain structure has a member for the size of PCM period in
PCM substream which starts AMDTP streams in domain. When the value has
zero and the substream count is greater than 1, it means that any MIDI
substream starts AMDTP streams in domain. Usage of the value can resolve
the above issue.
This commit replaces the check with the substream count and the value for
the size of PCM period.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-13-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation, when opening a PCM substream, it's needed to
check whether the opposite PCM substream runs. This is to assign
effectual constraints (e.g. sampling rate) to opened PCM substream.
The number of PCM substreams and MIDI substreams on AMDTP streams in
domain is recorded in own structure. Usage of this count is an
alternative of the above check. This is better because the count is
incremented in pcm.hw_params earlier than pcm.trigger.
This idea has one issue because it's incremented for MIDI substreams as
well. In current implementation, for a case that any MIDI substream run
and a PCM substream is going to start, PCM application to start the PCM
substream can decide hardware parameters by restart packet streaming.
Just checking the substream count can brings regression.
Now AMDTP domain structure has a member for the size of PCM period in
PCM substream which starts AMDTP streams in domain. When the value has
zero and the substream count is greater than 1, it means that any MIDI
substream starts AMDTP streams in domain. Usage of the value can resolve
the above issue.
This commit replaces the check with the substream count and the value for
the size of PCM period.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-12-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In current implementation, when opening a PCM substream, it's needed to
check whether the opposite PCM substream runs. This is to assign
effectual constraints (e.g. sampling rate) to opened PCM substream.
The number of PCM substreams and MIDI substreams on AMDTP streams in
domain is recorded in own structure. Usage of this count is an
alternative of the above check. This is better because the count is
incremented in pcm.hw_params earlier than pcm.trigger.
This idea has one issue because it's incremented for MIDI substreams as
well. In current implementation, for a case that any MIDI substream run
and a PCM substream is going to start, PCM application to start the PCM
substream can decide hardware parameters by restart packet streaming.
Just checking the substream count can brings regression.
Now AMDTP domain structure has a member for the size of PCM period in
PCM substream which starts AMDTP streams in domain. When the value has
zero and the substream count is greater than 1, it means that any MIDI
substream starts AMDTP streams in domain. Usage of the value can resolve
the above issue.
This commit replaces the check with the substream count and the value for
the size of PCM period.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-11-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to share the size of PCM period between
PCM substreams on AMDTP streams in the same domain. At this time,
the size of PCM period in PCM substream which starts AMDTP streams in the
same domain is recorded.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-10-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to share the size of PCM period between
PCM substreams on AMDTP streams in the same domain. At this time,
the size of PCM period in PCM substream which starts AMDTP streams in the
same domain is recorded.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-9-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to share the size of PCM period between
PCM substreams on AMDTP streams in the same domain. At this time,
the size of PCM period in PCM substream which starts AMDTP streams in the
same domain is recorded.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-8-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to share the size of PCM period between
PCM substreams on AMDTP streams in the same domain. At this time,
the size of PCM period in PCM substream which starts AMDTP streams in the
same domain is recorded.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-7-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to share the size of PCM period between
PCM substreams on AMDTP streams in the same domain. At this time,
the size of PCM period in PCM substream which starts AMDTP streams in the
same domain is recorded.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-6-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to share the size of PCM period between
PCM substreams on AMDTP streams in the same domain. At this time,
the size of PCM period in PCM substream which starts AMDTP streams in the
same domain is recorded.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-5-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to share the size of PCM period between
PCM substreams on AMDTP streams in the same domain. At this time,
the size of PCM period in PCM substream which starts AMDTP streams in the
same domain is recorded.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-4-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a preparation to share the size of PCM period between
PCM substreams on AMDTP streams in the same domain. At this time,
the size of PCM period in PCM substream which starts AMDTP streams in the
same domain is recorded.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-3-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, it's called as 'event' what has presentation time
represented by timestamp in CIP header. Although the ratio of the number
of event against the number of data block is different depending on
event data type represented by the specific field in CIP header, it's
just one in the most cases supported by ALSA IEC 61883-1/6 engine.
In 1394 OHCI specification, applications can schedule hardware IRQ
by configuring descriptor with IRQ flag for packet against each
isochronous cycle. For future commit, I use the hardware IRQ for
isoc IT context to acknowledge the elapse of PCM period for both
playback/capture directions on AMDTP streams in the same domain.
This commit is a preparation for the above idea. This commit adds
a member into AMDTP domain structure to record the number of PCM frames.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20191007110532.30270-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Asus MJ401TA (with Realtek ALC256), the headset mic is connected to
pin 0x19, with default configuration value 0x411111f0 (indicating no
physical connection).
Enable this by quirking the pin. Mic jack detection was also tested and
found to be working.
This enables use of the headset mic on this product.
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20191017081501.17135-1-drake@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Typically, the r0 (calibration data) and temperature were measured in the factory.
This information is written into the non-volatile area
where keeps data whether factory reset or OS update.
In Chromium OS case, the coreboot will read the info from VPD and create
the device property for each rt1011.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Tested-By: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20191016085845.11672-1-shumingf@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Simplify the memcpy/be32_to_cpu() code by simply using
get_unaligned_be32() throughout and makes the code nicer
to look at.
This fixes the following warnings from sparse:
sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64
sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64
sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64
sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64
sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64
sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64
sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64
sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64
sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64
sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64
sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32
sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32
Signed-off-by: Ben Dooks <ben.dooks@codethink.co.uk>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20191016120149.5860-1-ben.dooks@codethink.co.uk
Signed-off-by: Mark Brown <broonie@kernel.org>
fmt 0 is perfectly valid (PCM3168A_FMT_I2S). Remove the return in case
fmt == 0.
Fixes: ("ASoC: pcm3168a: Use fixup instead of constraint for channels and formats")
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20191015090037.23271-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
rt5663 and rt5514 needs mclk/sclk early to synchronize its internal
clocks.
Signed-off-by: Naveen M <naveen.m@intel.com>
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Brent Lu <brent.lu@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/1571070480-25666-1-git-send-email-brent.lu@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When LEDS_TRIGGER_AUDIO is m and SND_SOC_SOF is y,
sound/soc/sof/control.o: In function `snd_sof_switch_put':
control.c:(.text+0x587): undefined reference to `ledtrig_audio_set'
control.c:(.text+0x593): undefined reference to `ledtrig_audio_set'
Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: 5d43001ae4 ("ASoC: SOF: acpi led support for switch controls")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191014091308.23688-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
sof_parse_tokens() returns a value that is checked on every call
except for LED tokens, fix with explicit test.
Detected with cppcheck warning:
sound/soc/sof/topology.c:973:6: style: Variable 'ret' is assigned a
value that is never used. [unreadVariable]
ret = sof_parse_tokens(scomp, &scontrol->led_ctl, led_tokens,
^
Fixes: 5d43001ae4 ("ASoC: SOF: acpi led support for switch controls")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191011164312.7988-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Address cppcheck warning:
sound/soc/sof/topology.c:2322:6: style: Condition 'pcm' is always true
[knownConditionTrueFalse]
if (pcm) {
^
sound/soc/sof/topology.c:2311:6: note: Assuming that condition '!pcm'
is not redundant
if (!pcm)
^
sound/soc/sof/topology.c:2322:6: note: Condition 'pcm' is always true
if (pcm) {
^
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191011164312.7988-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
gcc warn about this:
sound/soc/fsl/fsl_mqs.c:146:1: warning:
static is not at beginning of declaration [-Wold-style-declaration]
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20191011143538.15300-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When STM32 SPDIFRX is in sync state, allow multiple
synchro attempts, instead of exiting on first unsuccessful
trial. This is useful when spdif signal is not immediately
available on input. This also allows Pulseaudio to check
iec capture device availability when no signal is present.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/20191011084816.14279-1-olivier.moysan@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ESAI parameters are read for topology file, packed into
sof_ipc_dai_esai_parms struct and then sent to DSP.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191008164443.1358-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently sof doesn't support acpi leds with mute switches. So implement
acpi leds following quite shamelessly existing HDA implementation by
Takashi Iwai.
Mute leds can be enabled in topology by adding led and direction token
in switch control private data.
Signed-off-by: Jaska Uimonen <jaska.uimonen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191008164443.1358-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add debug parameter for snd-sof-pci.
One of the usages for this debug parameter to disable pm_runtime,
which can be useful for platform bringup, or keep the parent device
active while enabling pm_runtime for child devices (e.g. with
SoundWire or MFD). This can also be useful to measure suspend-resume
latencies or child devices.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191008164443.1358-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add debug parameter for snd-sof-acpi.
One of the usages for this debug parameter to disable pm_runtime,
which can be useful for platform bringup, or keep the parent device
active while enabling pm_runtime for child devices (e.g. with
SoundWire or MFD). This can also be useful to measure suspend-resume
latencies or child devices.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191008164443.1358-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 64ca9d9fcb.
The workaround is no longer needed after configuring HDAC
bus with sync_write=1.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191008164443.1358-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently sof pga element supports only 1 kcontrol and you can't create
for example a mixer element with combined volume slider and mute switch.
So enable sof pga to have more than 1 kcontrol associated with it. Also
check for possible NULL tlv pointer as switch element might not have it.
Signed-off-by: Jaska Uimonen <jaska.uimonen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191008164443.1358-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Align SOF HDA implementation with snd-hda-intel driver and enable
sync_write flag for all supported Intel platforms in SOF. When set,
a sync is issued after each verb write.
Sync after write has helped to overcome intermittent delays in
system resume flow on Intel Coffee Lake systems, and most recently
probe errors related to the HDMI codec on Ice Lake systems.
Matches the snd-hda-intel driver change done in commit 2756d9143a
("ALSA: hda - Fix intermittent CORB/RIRB stall on Intel chips").
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191008164443.1358-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for the 8 channel PDM-to-I2S/TDM converter. The
ADAU7118 converts four stereo pulse density modulation (PDM) bitstreams
into one pulse code modulation (PCM) output stream. The source for the PDM
data can be eight microphones or other PDM sources. The PCM audio data is
output on a serial audio interface port in either inter-IC serial (I2S) or
time domain multiplexed (TDM) format.
Signed-off-by: Nuno Sá <nuno.sa@analog.com>
Link: https://lore.kernel.org/r/20191010074234.7344-1-nuno.sa@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds missing MIX2 path on RX1/2 which take IIR1 and
IIR2 as inputs.
Without this patch sound card fails to intialize with below warning:
ASoC: no sink widget found for RX1 MIX2 INP1
ASoC: Failed to add route IIR1 -> IIR1 -> RX1 MIX2 INP1
ASoC: no sink widget found for RX2 MIX2 INP1
ASoC: Failed to add route IIR1 -> IIR1 -> RX2 MIX2 INP1
ASoC: no sink widget found for RX1 MIX2 INP1
ASoC: Failed to add route IIR2 -> IIR2 -> RX1 MIX2 INP1
ASoC: no sink widget found for RX2 MIX2 INP1
ASoC: Failed to add route IIR2 -> IIR2 -> RX2 MIX2 INP1
Reported-by: Stephan Gerhold <stephan@gerhold.net>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20191009111944.28069-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
After commit 67ad656bdd ("ASoC: jz4740: Use
of_device_get_match_data()"), the match local variable is unused and the
compiler rightly warns.
sound/soc/jz4740/jz4740-i2s.c: In function 'jz4740_i2s_dev_probe':
sound/soc/jz4740/jz4740-i2s.c:500:29: warning: unused variable 'match' [-Wunused-variable]
500 | const struct of_device_id *match;
Drop it.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Cc: Arnd Bergmann <arnd@arndb.de>
Cc: Geert Uytterhoeven <geert@linux-m68k.org>
Cc: Paul Cercueil <paul@crapouillou.net>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Rob Herring <robh+dt@kernel.org>
Cc: Frank Rowand <frowand.list@gmail.com>
Cc: <alsa-devel@alsa-project.org>
Fixes: 67ad656bdd ("ASoC: jz4740: Use of_device_get_match_data()")
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Link: https://lore.kernel.org/r/20191008235212.228856-1-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_rtdcom_lookup() will be removed.
It is used at fsl_dma_isr(), but it is just for dev_err(dev..).
rtd->dev is very enough for it.
This patch replace component->dev to rtd->dev
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877e5n90cn.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_pcm_ops is no longer needed.
Let's use component driver callback.
*Note*
Only Intel skl-pcm has .get_time_info implementation, but ALSA SoC
framework doesn't call it so far.
To keep its implementation, this patch keeps .get_time_info,
but it is still not called.
Intel guy need to support it in the future.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o8yzaf2f.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current snd_soc_component_driver has pcm_new/pcm_free, but,
it doesn't have "component" at parameter.
Thus, each callback can't know it is called for which component.
Each callback currently is getting "component" by using
snd_soc_rtdcom_lookup() with driver name.
It works today, but, will not work in the future if we support multi
CPU/Codec/Platform, because 1 rtd might have multiple same driver
name component.
To solve this issue, each callback need to be called with component.
This patch adds new pcm_construct/pcm_destruct with "component"
parameter.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sgobaf3g.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current snd_soc_component_driver has snd_pcm_ops, and each driver can
have callback via it (1).
But, it is mainly created for ALSA, thus, it doesn't have "component"
as parameter for ALSA SoC (1)(2).
Thus, each callback can't know it is called for which component.
Thus, each callback currently is getting "component" by using
snd_soc_rtdcom_lookup() with driver name (3).
--- ALSA SoC ---
...
if (component->driver->ops &&
component->driver->ops->open)
(1) return component->driver->ops->open(substream);
...
--- driver ---
(2) static int xxx_open(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
(3) struct snd_soc_component *component = snd_soc_rtdcom_lookup(..);
...
}
It works today, but, will not work in the future if we support multi
CPU/Codec/Platform, because 1 rtd might have multiple components which
have same driver name.
To solve this issue, each callback needs to be called with component.
We already have many component driver callback.
This patch copies each snd_pcm_ops member under component driver,
and having "component" as parameter.
--- ALSA SoC ---
...
if (component->driver->open)
=> return component->driver->open(component, substream);
...
--- driver ---
=> static int xxx_open(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
...
}
*Note*
Only Intel skl-pcm has .get_time_info implementation, but ALSA SoC
framework doesn't call it so far.
To keep its implementation, this patch keeps .get_time_info,
but it is still not called.
Intel guy need to support it in the future.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tv8raf3r.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Before regmap_reinit_cache we must reset audio regs as default values.
So we use reset controller unit(toprgu) to reset audio hw.
Signed-off-by: Jiaxin Yu <jiaxin.yu@mediatek.com>
Reviewed-by: Yingjoe Chen <yingjoe.chen@mediatek.com>
Link: https://lore.kernel.org/r/1569580317-21181-5-git-send-email-jiaxin.yu@mediatek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_pcm_hw_constraint_minmax() works fine when a single codec is
connected to a single CPU DAI, but in multicodec or DPCM setup the
constraints placed by the driver will apply to the whole PCM stream (FE
included) and thus prevents more than 8 playback channels for example.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20191008115720.7135-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
According the documentation for snd_soc_update_bits the API
will return a 1 if the update was successful with a value change,
a 0 if the update was successful with no value change or a negative
if the command just failed.
So the value of return in the driver needs to be checked for being less
then 0 or the caller may indicate failure when the value actually
changed.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20191007171157.17813-1-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove unused defines and structure variables that are not
referenced by the code. If these are needed for future
enhancements then they should be added at that time.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20191007171157.17813-3-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove the unneeded and incorrect read of the TDM_CFG3 register.
The read is done but the value is never used.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20191007171157.17813-2-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There are several problems in the error handling in fsl_mqs_probe().
1) "ret" isn't initialized on some paths. GCC has a feature which
warns about uninitialized variables but the code initializes "ret"
to zero at the start of the function so the checking is turned off.
2) "gpr_np" is a pointer so initializing it to zero is confusing and
generates a Sparse warning.
3) of_parse_phandle() doesn't return error pointers on error, it returns
NULL.
4) If devm_snd_soc_register_component() fails then the function should
free the "gpr_np".
Fixes: 9e28f6532c ("ASoC: fsl_mqs: Add MQS component driver")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20191004102208.GB823@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/soc/fsl/fsl_mqs.c: In function fsl_mqs_hw_params:
sound/soc/fsl/fsl_mqs.c:54:6: warning: variable bclk set but not used [-Wunused-but-set-variable]
It is never used, so can be removed.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20191006105522.58560-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If GPIOLIB is not set, building fails:
sound/soc/codecs/tas2770.c: In function tas2770_reset:
sound/soc/codecs/tas2770.c:38:3: error: implicit declaration of function gpiod_set_value_cansleep; did you mean gpio_set_value_cansleep? [-Werror=implicit-function-declaration]
gpiod_set_value_cansleep(tas2770->reset_gpio, 0);
^~~~~~~~~~~~~~~~~~~~~~~~
gpio_set_value_cansleep
sound/soc/codecs/tas2770.c: In function tas2770_i2c_probe:
sound/soc/codecs/tas2770.c:749:24: error: implicit declaration of function devm_gpiod_get_optional; did you mean devm_regulator_get_optional? [-Werror=implicit-function-declaration]
tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev,
^~~~~~~~~~~~~~~~~~~~~~~
devm_regulator_get_optional
sound/soc/codecs/tas2770.c:751:13: error: GPIOD_OUT_HIGH undeclared (first use in this function); did you mean GPIOF_INIT_HIGH?
GPIOD_OUT_HIGH);
^~~~~~~~~~~~~~
GPIOF_INIT_HIGH
Reported-by: Hulk Robot <hulkci@huawei.com>
Fixes: 1a476abc72 ("tas2770: add tas2770 smart PA kernel driver")
Suggested-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20191006104631.60608-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This probe function is only called if the device is backed by a DT node,
so switch this call to of_device_get_match_data() to reduce code size
and simplify a bit. This also avoids needing to reference a potentially
undefined variable because of_device_get_match_data() doesn't need to
know anything beyond the struct device to find the match table.
Cc: Arnd Bergmann <arnd@arndb.de>
Cc: Geert Uytterhoeven <geert@linux-m68k.org>
Cc: Paul Cercueil <paul@crapouillou.net>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Rob Herring <robh+dt@kernel.org>
Cc: Frank Rowand <frowand.list@gmail.com>
Cc: <alsa-devel@alsa-project.org>
Signed-off-by: Stephen Boyd <swboyd@chromium.org>
Link: https://lore.kernel.org/r/20191004214334.149976-8-swboyd@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
We can have 2 dcpm-s with the same backend and frontend name
(capture + playback pair), this causes the following debugfs error
on Intel Bay Trail systems:
[ 298.969049] debugfs: Directory 'SSP2-Codec' with parent 'Baytrail Audio Port' already present!
This commit adds a ":playback" or ":capture" postfix to the debugfs dir
name fixing this.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20191005212202.5206-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix potential DMA hang upon starting playback on devices in HDA mode
on Intel platforms (Gemini Lake/Whiskey Lake/Comet Lake/Ice Lake). It
doesn't affect platforms before Gemini Lake or any Intel device in
non-HDA mode.
The reset value for the LOSDIV register is all output streams valid.
Clear this register to invalidate non-existent streams when the bus
is powered up.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190930142945.7805-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Headphone on XPS 9350/9360 produces a background white noise. The The
noise level somehow correlates with "Headphone Mic Boost", when it sets
to 1 the noise disappears. However, doing this has a side effect, which
also decreases the overall headphone volume so I didn't send the patch
upstream.
The noise was bearable back then, but after commit 717f43d81a ("ALSA:
hda/realtek - Update headset mode for ALC256") the noise exacerbates to
a point it starts hurting ears.
So let's use the workaround to set "Headphone Mic Boost" to 1 and lock
it so it's not touchable by userspace.
Fixes: 717f43d81a ("ALSA: hda/realtek - Update headset mode for ALC256")
BugLink: https://bugs.launchpad.net/bugs/1654448
BugLink: https://bugs.launchpad.net/bugs/1845810
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20191003043919.10960-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adjust indentation from spaces to tab (+optional two spaces) as in
coding style with command like:
$ sed -e 's/^ /\t/' -i */Kconfig
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20191004144931.3851-1-krzk@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_limit_volume() is finding snd_kcontrol by using original coding,
but we already have snd_soc_card_get_kcontrol().
Let's use existing function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y2y3afgd.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
card->rtd_num is used to count rtd. Initialize it at
snd_soc_instantiate_card() is very natural and less confusion.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zhijafgk.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
dai / component related resources are created when component is
registered, and it will be freed when component was unregistered.
These resources are not re-used after that.
This means, we can use devm_xxx for dai / component, without
thinking about kfree().
This patch uses devm_xxx for these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8736gbbu1a.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA needs to setup shortname, longname, and driver.
These methods are very similar.
This patch adds new soc_setup_card_name() and setup these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874l0rbu1i.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
It is easy to read code if it is cleanly using paired function/naming,
like start <-> stop, register <-> unregister, etc, etc.
But, current ALSA SoC code is very random, unbalance, not paired, etc.
It is easy to create bug at the such code, and it will be difficult to
debug.
soc_cleanup_card_resources() (a) which is paired function of
snd_soc_instantiate_card() (A) is calling soc_remove_dai_links() (*)
to remove card related resources, but it is breaking
add/remove balance (B)(b)(C)(c)(D)(d), in other words
these should be called from soc_cleanup_card_resources() (a)
from balance point of view.
More headacke is that it is using original removing method for
dai_link even though we already have snd_soc_remove_dai_link()
which is the function for it (d).
This patch removes snd_soc_remove_dai_links() and balance up code.
static void soc_remove_dai_links(...)
{
...
(b) soc_remove_link_dais(card);
(c) soc_remove_link_components(card);
for_each_card_links_safe(card, link, _link) {
...
/* it should use snd_soc_remove_dai_link() here */
(d) list_del(&link->list);
}
}
(a) static int soc_cleanup_card_resources(...)
{
...
/* remove and free each DAI */
(*) soc_remove_dai_links(card);
...
}
(A) static int snd_soc_instantiate_card(struct snd_soc_card *card)
{
...
/* add predefined DAI links to the list */
for_each_card_prelinks(card, i, dai_link)
(B) snd_soc_add_dai_link(card, dai_link);
...
/* probe all components used by DAI links on this card */
(C) ret = soc_probe_link_components(card);
...
/* probe all DAI links on this card */
(D) ret = soc_probe_link_dais(card);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875zl7bu1r.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rtd, rtd->dev, rtd->codec_dais are created by normal kzalloc(),
but we want to use devm_kzalloc() as much as possible.
Created rtd->dev is registered by device_register() at
soc_new_pcm_runtime(), and it will be freed at
soc_free_pcm_runtime() by device_unregister().
This means, if we can use devm_kzalloc(rtd->dev, xxx) for
rtd / rtd->codec_dais, all these are automatically freed
via soc_free_pcm_runtime().
This patch uses devm_kzalloc(rtd->dev, xxx) for rtd / rtd->codec_dais.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877e5nbu1z.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/tas2770.c:796:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Fixes: 1a476abc72 ("tas2770: add tas2770 smart PA kernel driver")
CC: Frank Shi <shifu0704@thundersoft.com>
Signed-off-by: kbuild test robot <lkp@intel.com>
Link: https://lore.kernel.org/r/20191001175504.ufhyxh66psrht42k@332d0cec05f4
Signed-off-by: Mark Brown <broonie@kernel.org>
If rockchip_pcm_platform_register() fails, e.g. upon deferring to wait
for an absent DMA channel, we return without disabling RPM, which makes
subsequent re-probe attempts scream with errors about the unbalanced
enable. Don't do that.
Fixes: ebb75c0bdb ("ASoC: rockchip: i2s: Adjust devm usage")
Signed-off-by: Robin Murphy <robin.murphy@arm.com>
Link: https://lore.kernel.org/r/bcb12a849a05437fb18372bc7536c649b94bdf07.1570029862.git.robin.murphy@arm.com
Signed-off-by: Mark Brown <broonie@kernel.org>
User space always expects to be able to read ALSA controls, so ensure
no kcontrols are generated without an appropriate READ flag. In the case
of a read of such a control zeros will be returned.
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20191002084240.21589-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Rename arndale_rt5631.c to just arnddale.c as we support other CODECs
than RT5631. While at it replace spaces in Kconfig with tabs.
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Link: https://lore.kernel.org/r/20191002105652.24821-3-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Arndale boards come with different types of the audio daughter
board. In order to support the WM1811 one we add new definition of
an ASoC card which will be registered when the driver matches on
"samsung,arndale-wm1811" compatible. There is no runtime detection of
the audio daughter board type at the moment, compatible string of the
audio card needs to be adjusted in DT, e.g. by the bootloader,
depending on actual audio board (CODEC) used.
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Link: https://lore.kernel.org/r/20191002105652.24821-2-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is only one DAI link so we can drop an unnecessary loop statement.
Use card->dai_link in place of direct static arndale_rt5631_dai[] array
dereference as a prerequisite for adding support for other CODECs.
Unnecessary assignment of dai_link->codecs->name to NULL is removed.
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Link: https://lore.kernel.org/r/20191002105652.24821-1-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Helps with FW debug as it provides DSP IPC processing context.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-13-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch prevents the reset of host period bytes
and uses no_stream_position to record requests
for stream position.
Signed-off-by: Marcin Rajwa <marcin.rajwa@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
fw_ready should be a mandatory op. Make sure fw_ready ops is set
during probe.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add config option to prevent DSP entering D3 after any FW exception.
This can then be used to dump FW context for debug.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
In a number of debug cases, the DMA-based trace can add problems
(e.g. with HDaudio channel allocation). It also generates additional
traffic on the bus and if the DMA handling is unreliable will prevent
audio use-cases from working normally. Using the trace also requires
tools to be installed on the target.
The trace can be instead handled as dynamic debug. We can use a
Kconfig to force the trace to be enabled in all cases, or use a module
parameter to enable it on a need-basis, e.g. by setting "options
snd_sof sof_debug=0x1" in a /etc/modprobe.d file.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-4-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When removing sof module the support_button_press function will oops
because hp_jack pointer is not checked for NULL. So add a check to fix
this.
Signed-off-by: Jaska Uimonen <jaska.uimonen@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927201408.925-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When removing sof module the rt5682 jack handler will oops
if jack detection is not disabled. So add remove function,
which disables the jack detection.
Signed-off-by: Jaska Uimonen <jaska.uimonen@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927201408.925-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a known issue on some Intel platforms which causes
pause/release to run into xrun's during capture usecases.
The suggested workaround to address the issue is to
disable the entry of lower power L1 state in the physical
DMI link when there is a capture stream open.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-14-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
FW mailbox offset was not set before use and HDR size was not validated.
Fix this.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The "snd_pcm_substream" handle was not initialized properly
in hda-loader.c for firmware load.
When the HDA DMAs were used to load the firmware,
the interrupts related to firmware load also triggered
calls to snd_sof_pcm_period_elapsed() on a non-existent ALSA
PCM stream.
This caused runtime kernel warnings from
pcm_lib.c:snd_pcm_period_elapsed().
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The old STOP sequence is: 1. stop DMA 2. send STOP ipc
If delay happen before the steps 1 and 2, the DMA buffer will be empty in
short time and cause pipeline xrun then stop the pipeline.
Then the step 2 ipc stop will return error as pipeline is already stopped.
Suggested change to avoid the issue is to switch the order of steps 1 and 2
for the stop sequence.
Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix a bug in sof_pcm_hw_free() where some cleanup actions were
skipped if STREAM_PCM_FREE IPC was already successfully sent to
DSP when the stream was stopped or suspended. This is incorrect
as hw_free should clean up also other resources, including pcm
lib page allocations, period elapsed work queue and call to
platform hw_free.
Fixes: c29d96c3b9 ("ASoC: SOF: reset DMA state in prepare")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We are using sof_parse_word_tokens() to parse tokens with
bool/byte/short/word tuple types, here add the missing check, to fix the
parsing failure at byte/bool tuple types.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When we fail to boot the firmware, we encounter a kernel oops in
hda_dsp_get_registers(), which is called conditionally in
hda_dsp_dump() when the sdev_>boot_complete flag is set.
Setting this flag _after_ dumping the data fixes the issue and does
not change the programming flow.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927200538.660-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix potential NULL pointer dereference for alg_region in
wm_adsp_buffer_parse_legacy. In practice this can never happen as
loading the firmware should have failed at the wm_adsp2_setup_algs
stage, however probably better for the code to be robust against
future changes and this is more helpful for static analysis.
Signed-off-by: Li Xu <li.xu@cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20191001130911.19238-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Below Oops is caused in a system which uses ACPI instead of device node:
of_get_named_gpiod_flags: can't parse 'maxim,reset-gpio' property of node '(null)[0]'
BUG: kernel NULL pointer dereference, address: 0000000000000010
This patch avoids NULL pointer deferencing by adding a check before parsing
and initializes to make reset-gpio pin as invalid.
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Jairaj Arava <jairaj.arava@intel.com>
Link: https://lore.kernel.org/r/1569702150-11976-1-git-send-email-sathyanarayana.nujella@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
On stream stop currently we stop the DMA first followed by the CPU DAI.
This can cause underflow (playback) or overflow (capture) on the DAI side
as the DMA is no longer feeding data while the DAI is still active.
It can be observed easily if the DAI side does not have FIFO (or it is
disabled) to survive the time while the DMA is stopped, but still can
happen on relatively slow CPUs when relatively high sampling rate is used:
the FIFO is drained between the time the DMA is stopped and the DAI is
stopped.
It can only fixed by using different sequence within trigger for 'stop' and
'start':
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
Trigger order: dai_link, DMA, CPU DAI then the codec
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
Trigger order: codec, CPU DAI, DMA then dai_link
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190927071646.22319-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is error "aplay: pcm_write:2023: write error: Input/output error"
on i.MX8QM/i.MX8QXP platform for S24_3LE format.
In i.MX8QM/i.MX8QXP, the DMA is EDMA, which don't support 24bit
sample, but we didn't add any constraint, that cause issues.
So we need to query the caps of dma, then update the hw parameters
according to the caps.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/b6a4de2bbf960ef291ee902afe4388bd0fc1d347.1569493933.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When set the runtime hardware parameters, we may need to query
the capability of DMA to complete the parameters.
This patch is to Extract this operation from
dmaengine_pcm_set_runtime_hwparams function to a separate function
snd_dmaengine_pcm_refine_runtime_hwparams, that other components
which need this feature can call this function.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/d728f65194e9978cbec4132b522d4fed420d704a.1569493933.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The ASRC support 24bit/16bit/8bit input width, which is
data width, not slot width.
For the S20_3LE format, the data with is 20bit, slot width
is 24bit, if we set ASRMCR1n.IWD to be 24bits, the result
is the volume is lower than expected, it likes 24bit data
right shift 4 bits
So replace S20_3LE with S24_3LE in supported list and add S8
format in TX supported list
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/45a7c383f43cc1dd9d0934846447aee653278c03.1569493933.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_pcm_format_t is more formal than enum asrc_word_width, which has
two property, width and physical width, which is more accurate than
enum asrc_word_width. So it is better to use in(out)put_format
instead of in(out)put_word_width.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/7937c1404ee327ce141cb03b3575b02ea01a740c.1569493933.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The adau1761 has a automatic level control block that can adjust the gain
for the differential input PGA. This patch adds ALSA controls for enabling
and changing the parameter settings for the ALC.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Alexandru Ardelean <alexandru.ardelean@analog.com>
Link: https://lore.kernel.org/r/20190926115012.24049-2-alexandru.ardelean@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The PGA Slew Time control allows to configure the rate with which the PGA
gain control ramps up/down to the target setting.
The PGA slew control is done via the ALC Control 0 register. There are 2
bits on that reg, that control PGA slew time, while the other bits control
parts of the ALC (automatic level control) block.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Alexandru Ardelean <alexandru.ardelean@analog.com>
Link: https://lore.kernel.org/r/20190926115012.24049-1-alexandru.ardelean@analog.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For a sound card ignore_machine means that existing FEs links should be
ignored and existing BEs links should be overridden with some information
from the matching component driver.
Current code make some confusions about this so fix it!
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20190925183358.11955-1-daniel.baluta@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If we can use devm_kzalloc(rtd->dev, xxx) for rtdcom,
we don't need to call snd_soc_rtdcom_del_all() for kfree().
This patch uses devm_kzalloc(rtd->dev, xxx) for rtdcom,
and remove snd_soc_rtdcom_del_all().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875zlyf7ln.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch allocs dev first at soc_new_pcm_runtime().
This is prepare for rtd->dev, rtd->codec_dais alloc cleanup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877e6ef7m1.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
"rtd" is handled by soc_xxx_pcm_runtime(), and
"rtd->dev" is handled by soc_rtd_xxx().
There is no reason to separate these, and it makes code complex.
We can free these in the same time.
Here soc_rtd_free() (A) which frees rtd->dev is called from
soc_remove_link_dais() many times (1).
Then, it is using dev_registered flags to avoid multi kfree() (2).
This is no longer needed if we can merge these functions.
static void soc_remove_link_dais(...)
{
...
(1) for_each_comp_order(order) {
(1) for_each_card_rtds(card, rtd) {
(A) soc_rtd_free(rtd);
...
}
}
}
(A) static void soc_rtd_free(...)
{
(2) if (rtd->dev_registered) {
/* we don't need to call kfree() for rtd->dev */
device_unregister(rtd->dev);
(2) rtd->dev_registered = 0;
}
}
This patch merges soc_rtd_free() into soc_free_pcm_runtime().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/878squf7oi.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
"rtd" is handled by soc_xxx_pcm_runtime(), and
"rtd->dev" is handled by soc_rtd_xxx().
There is no reason to separate these, and it makes code complex.
We can create these in the same time.
This patch merges soc_rtd_init() into soc_new_pcm_runtime().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a7baf7pz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_new_pcm_runtime() allocs rtd and rtd->codec_dais.
This patch allocs both first, and setup these after that.
This is prepare for soc_new_pcm_runtime() cleanup.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87blvqf7qz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ALSA SoC is calling list_del(&rtd->list) at (1)
static void soc_remove_pcm_runtimes(...)
{
...
for_each_card_rtds_safe(card, rtd, _rtd) {
(1) list_del(&rtd->list);
(2) soc_free_pcm_runtime(rtd);
}
...
}
But, we will call soc_free_pcm_runtime() after that (2).
&rtd->list is connected at soc_new_pcm_runtime(),
Thus, it should be disconnected at soc_free_pcm_runtime().
This patch calls list_del(&rtd->list) at soc_free_pcm_runtime().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87d0g6f7s5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We have soc_new_pcm_runtime() which allocs rtd and its related memory,
and soc_add_pcm_runtime() which connects rtd to card.
But we don't need to separate these, we can alloc and connect rtd
in the same time.
Current implementation is just makes code complex.
This patch merges these into one.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ef0mf7sl.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch moves soc_free_pcm_runtime() next to soc_new_pcm_runtime().
This is prepare for soc_xxx_pcm_runtime() cleanup.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87ftl2f7sx.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As an intermediate step before covering the clocking subsystem
of the CODEC entirely by the clk API add handling of external CODEC's
master clocks in DAPM events when the AIFn clocks are sourced directly
from MCLKn; when FLLn are used we enable/disable respective MCLKn
before/after FLLn is enabled/disabled.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20190920130218.32690-5-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Extend the set_sysclk() handler so we also set frequency of the MCLK1,
MCLK2 clocks through clk API when those clocks are specified in DT.
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20190920130218.32690-4-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Simplify this function implementation by using a known wrapper function.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Link: https://lore.kernel.org/r/8f7cf483-6ab3-d00f-5606-863e9f5b31fc@web.de
Signed-off-by: Mark Brown <broonie@kernel.org>
As of commit 648e921888 ("clk: x86: Stop marking clocks as
CLK_IS_CRITICAL"), the cht_bsw_rt5645 driver needs to enable the clock
it's using for the codec's mclk. It does this from commit 7735bce05a
("ASoC: Intel: boards: use devm_clk_get() unconditionally"), enabling
pmc_plt_clk_3. However, Strago family Chromebooks use pmc_plt_clk_0 for
the codec mclk, resulting in white noise with some digital microphones.
Add a DMI-based quirk for Strago family Chromebooks to use pmc_plt_clk_0
instead - mirroring the changes made to cht_bsw_max98090_ti in
commit a182ecd380 ("ASoC: intel: cht_bsw_max98090_ti: Add quirk for
boards using pmc_plt_clk_0") and making use of the existing
dmi_check_system() call and related infrastructure added in
commit 22af29114e ("ASoC: Intel: cht-bsw-rt5645: add quirks for
SSP0/AIF1/AIF2 routing").
Signed-off-by: Sam McNally <sammc@chromium.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190917054933.209335-1-sammc@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
MQS (medium quality sound), is used to generate medium quality
audio via a standard digital output pin. It can be used to
connect stereo speakers or headphones simply via power amplifier
stages without an additional DAC chip. It only accepts 2-channel,
LSB-valid 16bit, MSB shift-out first, frame sync asserting with
the first bit of the frame, data shifted with the posedge of
bit clock, 44.1 kHz or 48 kHz signals from SAI1 in left justified
format; and it provides the SNR target as no more than 20dB for
the signals below 10 kHz. The signals above 10 kHz will have
worse THD+N values.
MQS provides only simple audio reproduction. No internal pop,
click or distortion artifact reduction methods are provided.
The MQS receives the audio data from the SAI1 Tx section.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/74dfc73a92d2df4213225abe7d2a3db82672fe0f.1568367274.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The "template.id" variable is an enum and in this context GCC will
treat it as an unsigned int so it can never be less than zero.
Fixes: 8a9782346d ("ASoC: topology: Add topology core")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20190925110624.GR3264@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
A few small remaining wrap-up for this merge window.
Most of patches are device-specific (HD-audio and USB-audio quirks,
FireWire, pcm316a, fsl, rsnd, Atmel, and TI fixes), while there is
a simple fix (actually two commits) for ASoC core.
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Merge tag 'sound-fix-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A few small remaining wrap-up for this merge window.
Most of patches are device-specific (HD-audio and USB-audio quirks,
FireWire, pcm316a, fsl, rsnd, Atmel, and TI fixes), while there is a
simple fix (actually two commits) for ASoC core"
* tag 'sound-fix-5.4-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: Add DSD support for EVGA NU Audio
ALSA: hda - Add laptop imic fixup for ASUS M9V laptop
ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies
ASoC: pcm3168a: The codec does not support S32_LE
ASoC: core: use list_del_init and move it back to soc_cleanup_component
ALSA: hda/realtek - PCI quirk for Medion E4254
ALSA: hda - Apply AMD controller workaround for Raven platform
ASoC: rsnd: do error check after rsnd_channel_normalization()
ASoC: atmel_ssc_dai: Remove wrong spinlock usage
ASoC: core: delete component->card_list in soc_remove_component only
ASoC: fsl_sai: Fix noise when using EDMA
ALSA: usb-audio: Add Hiby device family to quirks for native DSD support
ALSA: hda/realtek - Fix alienware headset mic
ALSA: dice: fix wrong packet parameter for Alesis iO26
EVGA NU Audio is actually a USB audio device on a PCIexpress card,
with it's own USB controller. It supports both PCM and DSD.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20190924071143.30911-1-jussi@sonarnerd.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There were a couple places where the return value wasn't assigned so the
error handling wouldn't trigger.
Fixes: 5c0769af4c ("ASoC: soc-dai: add snd_soc_dai_bespoke_trigger()")
Fixes: 95aef35533 ("ASoC: soc-dai: add snd_soc_dai_trigger()")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20190923142257.GB31251@mwanda
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of WM1811 device there are currently being registered controls
referring to registers not existing on that device.
It has been noticed when getting values of "AIF1ADC2 Volume", "AIF1DAC2
Volume" controls was failing during ALSA state restoring at boot time:
"amixer: Mixer hw:0 load error: Device or resource busy"
Reading some registers through I2C was failing with EBUSY error and
indeed these registers were not available according to the datasheet.
To fix this controls not available on WM1811 are moved to a separate
array and registered only for WM8994 and WM8958.
There are some further differences between WM8994 and WM1811,
e.g. registers 603h, 604h, 605h, which are not covered in this patch.
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Link: https://lore.kernel.org/r/20190920130218.32690-2-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Ensure there is no OF node references kept when the driver
is removed/unbound.
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Link: https://lore.kernel.org/r/20190920130218.32690-3-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A small smattering of ASoC fixes for v5.4 - nothing too exciting
here, all small standalone things.
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Merge tag 'asoc-fix-v5.4-rc1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.4
A small smattering of ASoC fixes for v5.4 - nothing too exciting
here, all small standalone things.
