The previous fix forgot to remove the unused variable that triggers a
compile warning now:
sound/pci/hda/patch_realtek.c: In function 'alc285_fixup_hp_gpio_led':
sound/pci/hda/patch_realtek.c:4163:19: warning: unused variable 'spec' [-Wunused-variable]
Fix it.
Fixes: 404690649e ("ALSA: hda - reverse the setting value in the micmute_led_set")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Link: https://lore.kernel.org/r/20200812070256.32145-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of the DAPM widgets for DSP ASoC components reuse reg field
of the widgets for its internal calculations, however these are not
real registers. So read/writes to these numbers are not really
valid. However ASoC core will read these registers to get default
state during startup.
With recent changes to ASoC core, every register read/write
failures are reported very verbosely. Prior to this fails to reads
are totally ignored, so we never saw any error messages.
To fix this add dummy read/write function to return default value.
Fixes: e3a33673e8 ("ASoC: qdsp6: q6routing: Add q6routing driver")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200811120205.21805-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like the q6afe-dai dapm widget registers are set as "0",
which is a not correct.
As this registers will be read by ASoC core during startup
which will throw up errors, Fix this by making the registers
as SND_SOC_NOPM as these should be never used.
With recent changes to ASoC core, every register read/write
failures are reported very verbosely. Prior to this fails to reads
are totally ignored, so we never saw any error messages.
Fixes: 24c4cbcfac ("ASoC: qdsp6: q6afe: Add q6afe dai driver")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200811120205.21805-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Along with the recent unification of snd_soc_component_read*()
functions, the behavior of snd_soc_component_read() was changed
slightly; namely it returns the register read value directly, and even
if an error happens, it returns zero (but it prints an error
message). That said, the caller side can't know whether it's an error
or not any longer.
Ideally this shouldn't matter much, but in practice this seems causing
a regression, as John reported. And, grepping the tree revealed that
there are still plenty of callers that do check the error code, so
we'll need to deal with them in anyway.
As a quick band-aid over the regression, this patch changes the return
value of snd_soc_component_read() again to the negative error code.
It can't work, obviously, for 32bit register values, but it should be
enough for the known regressions, so far.
Fixes: cf6e26c71b ("ASoC: soc-component: merge snd_soc_component_read() and snd_soc_component_read32()")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200810134631.19742-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Before the micmute_led_set() is introduced, the function of
alc_gpio_micmute_update() will set the gpio value with the
!micmute_led.led_value, and the machines have the correct micmute led
status. After the micmute_led_set() is introduced, it sets the gpio
value with !!micmute_led.led_value, so the led status is not correct
anymore, we need to set micmute_led_polarity = 1 to workaround it.
Now we fix the micmute_led_set() and remove micmute_led_polarity = 1.
Fixes: 87dc36482c ("ALSA: hda/realtek - Add LED class support for micmute LED")
Reported-and-suggested-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200811122430.6546-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dsp_registers field of struct echoaduio has the volatile modifier,
but it's basically superfluous; the field is accessed only for the
base pointer of readl() and writel(), hence marking with __iomem alone
should suffice. OTOH, having the volatile prefix causes a compile
warning like:
sound/pci/echoaudio/echoaudio.c:1878:14: warning: passing argument 1 of 'iounmap' discards 'volatile' qualifier from pointer target type [-Wdiscarded-qualifiers]
So it's better to drop this superfluous modifier.
Link: https://lore.kernel.org/r/20200803143958.24324-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replacing string compare with "codec_dai->name" instead of comparing with
"codec_dai->component->name" in hw_params because,
Here the component name for codec RT1015 is "i2c-10EC5682:00"
and will never be "rt1015-aif1" as it is codec-dai->name.
So, strcmp() always compares and fails to set the
sysclk,pll,bratio for expected codec-dai="rt1015-aif1".
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20200807161046.17932-1-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The USB device (0x17aa:0x1046) that support Lenovo P620 rear panel
line-in claim to support volume control, but it doens't seem to have an
AMP, so when line-in volume lowers below 80, nothing gets recorded
anymore.
Disable the volume control to workaround the issue.
Fixes: f8c11eb7da ("ALSA: usb-audio: Add support for Lenovo ThinkStation P620")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200810133108.31580-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Further investigation of the L-R swap problem on the MS2109 reveals that
the problem isn't that the channels are swapped, but rather that they
are swapped and also out of phase by one sample. In other words, the
issue is actually that the very first frame that comes from the hardware
is a half-frame containing only the right channel, and after that
everything becomes offset.
So introduce a new quirk field to drop the very first 2 bytes that come
in after the format is configured and a capture stream starts. This puts
the channels in phase and in the correct order.
Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200810082400.225858-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After installing the Ubuntu Linux, the micmute led status is not
correct. Users expect that the led is on if the capture is disabled,
but with the current kernel, the led is off with the capture disabled.
We tried the old linux kernel like linux-4.15, there is no this issue.
It looks like we introduced this issue when switching to the led_cdev.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200810021659.7429-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3263"
Signed-off-by: Mirko Dietrich <buzz@l4m1.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200806124850.20334-1-buzz@l4m1.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a spelling mistake in a usb_audio_dbg debug message. Also
replace "param" with "parameter". Fix these.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20200806105134.46447-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tegra HDA has audio data buffer for upto tens of frames, this buffer
can help to avoid underflow. HW will keep issuing new data fetch
request when buffers are not full and current BDL is not done. When SW
disable DMA RUN bit for a stream, HW can't cancel the already issued data
fetch request and hence it can't stop DMA. HW has to wait for all issued
data fetch request get data returned before it stops DMA.
This HW behavior is not in sync with HDA spec which says DMA RUN bit
should be cleared within 1 audio frame. For Tegra, DMA RUN bit was
active for more than one audio frame, due to this the timeout in
snd_hdac_stream_sync function is not helping. When Stream reset set
and clear happens during DMA RUN bit active state it results in Memory
Decode error.
Unfortunately, there is no way to detect when these data accesses have
completed, but testing has shown that a 100us delay between Stream reset
set and clear operation for Tegra avoids the memory decode error.
Therefore, adding a 100us dma stop delay.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-4-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A variable dma_stop_delay is added as a new member in hdac_bus
structure to avoid memory decode error incase DMA RUN bit is not
disabled in the given timeout from snd_hdac_stream_sync function and
followed by stream reset which results in memory decode error between
reset set and clear operation.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-3-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set chip->align_buffer_size to 1 for Tegra platforms to make the buffer
alignment to be multiple of 128 bytes. This fix is applied as gstreamer
alsasink gets stuck with the default buffer-time and latency-time
parameters with 4 byte buffer alignment.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Link: https://lore.kernel.org/r/20200805095221.5476-2-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some ioctls via OSS sequencer API may race and lead to UAF when the
port create and delete are performed concurrently, as spotted by a
couple of syzkaller cases. This patch is an attempt to address it by
serializing the ioctls with the existing register_mutex.
Basically OSS sequencer API is an obsoleted interface and was designed
without much consideration of the concurrency. There are very few
applications with it, and the concurrent performance isn't asked,
hence this "big hammer" approach should be good enough.
Reported-by: syzbot+1a54a94bd32716796edd@syzkaller.appspotmail.com
Reported-by: syzbot+9d2abfef257f3e2d4713@syzkaller.appspotmail.com
Suggested-by: Hillf Danton <hdanton@sina.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200804185815.2453-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDMI on some platforms doesn't enable audio support because its Port
Connectivity [31:30] is set to AC_JACK_PORT_NONE:
Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0b000094: OUT Detect HBR HDMI DP
Pin Default 0x58560010: [N/A] Digital Out at Int HDMI
Conn = Digital, Color = Unknown
DefAssociation = 0x1, Sequence = 0x0
Pin-ctls: 0x40: OUT
Unsolicited: tag=00, enabled=0
Power states: D0 D3 EPSS
Power: setting=D0, actual=D0
Devices: 0
Connection: 3
0x02 0x03* 0x04
For now, use a quirk to force connectivity based on SSID. If there are
more platforms affected by the same issue, we can eye for a more generic
solution.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dock sold from startech (PID: ICUSBAUDIO7D) has no friendly name
and shows up currently as "USB Sound Device" in ALSA.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20200804010616.3399256-1-cujomalainey@chromium.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The variable rtd was left unused in psc_dma_free(), even unnoticed
during conversion to a new style:
sound/soc/fsl/mpc5200_dma.c:342:30: warning: unused variable 'rtd' [-Wunused-variable]
Drop the superfluous one.
Fixes: 6d1048bc11 ("ASoC: fsl: mpc5200_dma: remove snd_pcm_ops")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803144630.9615-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_i2s.c:167:12: warning: 'tegra210_i2s_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_i2s.c:179:12: warning: 'tegra210_i2s_runtime_resume' defined but not used [-Wunused-function]
Fixes: c0bfa98349 ("ASoC: tegra: Add Tegra210 based I2S driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-6-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_dmic.c:43:12: warning: 'tegra210_dmic_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_dmic.c:55:12: warning: 'tegra210_dmic_runtime_resume' defined but not used [-Wunused-function]
Fixes: 8c8ff982e9 ("ASoC: tegra: Add Tegra210 based DMIC driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-5-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_ahub.c:567:12: warning: 'tegra_ahub_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_ahub.c:579:12: warning: 'tegra_ahub_runtime_resume' defined but not used [-Wunused-function]
Fixes: 16e1bcc2ca ("ASoC: tegra: Add Tegra210 based AHUB driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-4-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra210_admaif.c:232:12: warning: 'tegra_admaif_runtime_resume' defined but not used [-Wunused-function]
sound/soc/tegra/tegra210_ahub.c:567:12: warning: 'tegra_ahub_runtime_suspend' defined but not used [-Wunused-function]
Fixes: f74028e159 ("ASoC: tegra: Add Tegra210 based ADMAIF driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial compile warnings wrt unused functions by adding
__maybe_unused prefix:
sound/soc/tegra/tegra186_dspk.c:74:12: warning: 'tegra186_dspk_runtime_suspend' defined but not used [-Wunused-function]
sound/soc/tegra/tegra186_dspk.c:86:12: warning: 'tegra186_dspk_runtime_resume' defined but not used [-Wunused-function]
Fixes: 327ef64702 ("ASoC: tegra: Add Tegra186 based DSPK driver")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200803141850.23713-2-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Lenovo ThinkStation P620 is like other TRX40 boards, is equipped with
two USB audio cards.
USB device (17aa:104d) provides functionality for Internal Speaker and
Front Headset. It's UAC v2, so it supports insertion control (jack
detection). However, when trying to get the connector status of the
speaker, an error occurs:
[ 5.787405] usb 3-1: cannot get connectors status: req = 0x81, wValue = 0x200, wIndex = 0x1000, type = 0
Since the insertion control works perfectly for the headset, the error
for speaker is probably casued by connecting internally. So let's relax
the error for a bit if it's a speaker, and always reports it's connected.
USB device (17aa:1046) is for rear Line-in, Line-out and Microphone.
The insertion control works for all three jacks. However, there's an
Function Unit that doesn't work:
[ 5.905415] usb 3-6: cannot get ctl value: req = 0x83, wValue = 0xc00, wIndex = 0x1300, type = 4
[ 5.905418] usb 3-6: 19:0: cannot get min/max values for control 12 (id 19)
So turn off the FU to avoid the error.
Also, add specific card name for both devices, so userspace can easily
indentify both cards.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200803142612.17156-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we found an issue about the suspend and resume. If dmic is
recording the sound, and we run suspend and resume, after the resume,
the dmic can't work well anymore. we need to close the app and reopen
the app, then the dmic could record the sound again.
For example, we run "arecord -D hw:CARD=acp,DEV=0 -f S32_LE -c 2
-r 48000 test.wav", then suspend and resume, after the system resume
back, we speak to the dmic. then stop the arecord, use aplay to play
the test.wav, we could hear the sound recorded after resume is weird,
it is not what we speak to the dmic.
I found two registers are set in the dai_hw_params(), if the two
registers are set during the resume, this issue could be fixed.
Move the code of the dai_hw_params() into the pdm_dai_trigger(), then
these two registers will be set during resume since pdm_dai_trigger()
will be called during resume. And delete the empty function
dai_hw_params().
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200730123138.5659-1-hui.wang@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Removing ADDITIONAL_CONTROL_4 from the list of readable registers cause
audio distortion.
This change was sent as a comment below the --- line when submitting
commit 658bb297e3 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE"), so
it was not supposed to get merged.
Keep WM8962_ADDITIONAL_CONTROL_4 inside wm8962_readable_register() to
fix the regression.
Fixes: 658bb297e3 ("ASoC: wm8962: Do not access WM8962_GPIO_BASE")
Reported-by: Shengjiu Wang <shengjiu.wang@gmail.com>
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20200803115233.19034-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
With this case:
aplay -Dhw:x 16khz.wav 24khz.wav
There is sound distortion for 24khz.wav. The reason is that setting
PLL of WM8962 with set_bias_level function, the bias level is not
changed when 24khz.wav is played, then the PLL won't be reset, the
clock is not correct, so distortion happens.
The resolution of this issue is to remove fsl_asoc_card_set_bias_level.
Move PLL configuration to hw_params and hw_free.
After removing fsl_asoc_card_set_bias_level, also test WM8960 case,
it can work.
Fixes: 708b4351f0 ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/1596420811-16690-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The biggest changes here one again come from Mormioto-san who has
continued his dilligent work cleaning up long standing issues in the
APIs, it's particularly nice to see the transition from digital_mute()
to mute_stream() finally completed. There's also been a lot of work on
the x86 code again, this time a big focus has been on cleaning up some
issues identified by various static tests, and on the Freescale systems.
Otherwise the biggest thing has been a lot of driver additions:
- Convert users of digital_mute() to mute_stream().
- Simplify I/O helper functions.
- Add a helper for getting the RTD from a substream.
- Many, many fixes and cleanups to the x86 code.
- New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim
MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia
Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some
of the first phones I worked on!) and TI J721e EVM.
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Merge tag 'asoc-v5.9' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.9
The biggest changes here one again come from Mormioto-san who has
continued his dilligent work cleaning up long standing issues in the
APIs, it's particularly nice to see the transition from digital_mute()
to mute_stream() finally completed. There's also been a lot of work on
the x86 code again, this time a big focus has been on cleaning up some
issues identified by various static tests, and on the Freescale systems.
Otherwise the biggest thing has been a lot of driver additions:
- Convert users of digital_mute() to mute_stream().
- Simplify I/O helper functions.
- Add a helper for getting the RTD from a substream.
- Many, many fixes and cleanups to the x86 code.
- New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim
MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia
Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some
of the first phones I worked on!) and TI J721e EVM.
This reverts commit 9a6418487b ("ALSA: hda: call runtime_allow()
for all hda controllers").
The reverted patch already introduced some regressions on some
machines:
- on gemini-lake machines, the error of "azx_get_response timeout"
happens in the hda driver.
- on the machines with alc662 codec, the audio jack detection doesn't
work anymore.
Fixes: 9a6418487b ("ALSA: hda: call runtime_allow() for all hda controllers")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208511
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200803064638.6139-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ca0113 command had the wrong group_id, 0x48 when it should've been
0x30. The front microphone selection should now work.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-3-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the ZxR headphone gain control was added, the ca0132_switch_get
function was not updated, which meant that the changes to the control
state were not saved when entering/exiting alsamixer.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-1-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are several Loongson-3 based laptops produced by CZC or Lemote,
they use alc269/alc662 codecs and need specific pin-tables, this patch
add their pin-tables.
Signed-off-by: Huacai Chen <chenhc@lemote.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1596360400-32425-1-git-send-email-chenhc@lemote.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Provide a mechanism for true two-step component registration. This
mimics device registration flow where initialization is the first step
while addition goes as second in line. Drivers may choose to modify
component's fields before registering component to ASoC subsystem via
snd_soc_add_component.
Patchset achieves status quo - behavior of snd_soc_register_component
remains unchanged.
Cezary Rojewski (3):
ASoC: core: Relocate and expose snd_soc_component_initialize
ASoC: core: Simplify snd_soc_component_initialize declaration
ASoC: core: Two step component registration
include/sound/soc-component.h | 3 --
include/sound/soc.h | 11 +++---
sound/soc/soc-component.c | 16 ---------
sound/soc/soc-core.c | 52 +++++++++++++++++----------
sound/soc/soc-generic-dmaengine-pcm.c | 14 +++++---
sound/soc/stm/stm32_adfsdm.c | 9 +++--
6 files changed, 55 insertions(+), 50 deletions(-)
--
2.17.1
Modify snd_soc_add_component so it calls snd_soc_component_initialize
no longer and thus providing true two-step registration. Drivers may
choose to change component's fields before actually adding it to ASoC
subsystem.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200731144146.6678-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Move 'name' field initialization responsibility back to
snd_soc_component_initialize to prepare snd_soc_add_component function
for being called separatelly as a second registration step.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200731144146.6678-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To allow for two-step component registration, expose
snd_soc_component_initialize function and move it back to soc-core.c.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200731144146.6678-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Enabling a whole subsystem from a single driver 'select' is frowned
upon and won't be accepted in new drivers, that need to use 'depends on'
instead. Existing selection of DMADEVICES will then cause circular
dependencies. Replace them with a dependency.
Signed-off-by: Laurent Pinchart <laurent.pinchart@ideasonboard.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Link: https://lore.kernel.org/r/20200731152433.1297-3-laurent.pinchart@ideasonboard.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai