regmap_register_patch can't be used to apply the probe sequence as a
patch is already registers with the regmap by
cs35l41_register_errata_patch and only a single patch can be attached to
a single regmap. The driver doesn't currently rely on a cache sync to
re-apply this probe sequence so simply switch it to a multi write.
Fixes: 6c4ddfa8b93f ("ALSA: hda: cs35l41: Add support for CS35L41 in HDA systems") Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Lucas Tanure <tanureal@opensource.cirrus.com> Link: https://lore.kernel.org/r/20220117160830.709403-1-tanureal@opensource.cirrus.com Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Sun, 16 Jan 2022 08:28:38 +0000 (09:28 +0100)]
ALSA: core: Fix SSID quirk lookup for subvendor=0
Some weird devices set the codec SSID vendor ID 0, and
snd_pci_quirk_lookup_id() loop aborts at the point although it should
still try matching with the SSID device ID. This resulted in a
missing quirk for some old Macs.
Fix the loop termination condition to check both subvendor and
subdevice.
ALSA: usb-audio: add mapping for MSI MPG X570S Carbon Max Wifi.
The USB audio device 0db0:419c based on the Realtek ALC4080 chip exposes
all playback volume controls as "PCM". This is makes distinguishing the
individual functions hard.
The added mapping distinguishes all playback volume controls as their
respective function:
- Speaker - for back panel output
- Frontpanel Headphone - for front panel output
- IEC958 - for digital output on the back panel
This clarifies the individual volume control functions for users.
ALSA: hda/realtek: fix speakers and micmute on HP 855 G8
There are several PCI ids associated with HP EliteBook 855 G8 Notebook
PC. Commit deb3688b8e8e9 ("ALSA: hda/realtek: fix mute/micmute LEDs for
HP 855 G8") covers 0x103c:0x8896, while this commit covers 0x103c:0x8895
which needs some additional work on top of the quirk from deb3688b8e8e9.
Note that the device can boot up with working speakers and micmute LED
without this patch, but the success rate would be quite low (order of
16 working boots across 709 boots) at least for the built-in drivers
scenario. This also means that there are some timing issues during early
boot and this patch is a workaround.
With this patch applied speakers and headphones are consistenly working,
as well as mute/micmute LEDs and the internal microphone.
Takashi Iwai [Fri, 7 Jan 2022 09:26:47 +0000 (10:26 +0100)]
ALSA: hda: Fix dependency on ASoC cs35l41 codec
The recently added support for CS35L41 codec unconditionally selects
CONFIG_SND_SOC_CS35L41_LIB, but this can't work unless the top-level
CONFIG_SND_SOC is enabled. This patch adds the proper dependency.
Charles Keepax [Fri, 7 Jan 2022 16:06:35 +0000 (16:06 +0000)]
ASoC: cs35l41: Update handling of test key registers
In preparation for the addition of PM runtime support move the test
key out of the register patches themselves. This is necessary to
allow the test key to be held during cache synchronisation, which is
required by the OTP settings which were unpacked from the device and
written by the driver.
Also whilst at it, the driver uses a mixture of accessing the test key
register by name and by address, consistently use the name.
Jiasheng Jiang [Fri, 7 Jan 2022 02:08:51 +0000 (10:08 +0800)]
ALSA: intel_hdmi: Check for error num after setting mask
To maintain the consistency of the code, it should be better to add the
sanity check after calling dma_set_mask_and_coherent(), like
tegra_pcm_dma_allocate() in `sound/soc/tegra/tegra_pcm.c`.
Yassine Oudjana [Tue, 4 Jan 2022 03:35:36 +0000 (03:35 +0000)]
ASoC: wcd9335: Keep a RX port value for each SLIM RX mux
Currently, rx_port_value is a single unsigned int that gets overwritten
when slim_rx_mux_put() is called for any RX mux, then the same value is
read when slim_rx_mux_get() is called for any of them. This results in
slim_rx_mux_get() reporting the last value set by slim_rx_mux_put()
regardless of which SLIM RX mux is in question.
Turn rx_port_value into an array and store a separate value for each
SLIM RX mux.
ASoC: amd: acp: acp-mach: Change default RT1019 amp dev id
RT1019 components was initially registered with i2c1 and i2c2 but
now changed to i2c0 and i2c1 in most of our AMD platforms. Change
default rt1019 components to 10EC1019:00 and 10EC1019:01 which is
aligned with most of AMD machines.
Any exception to rt1019 device ids in near future board design can
be handled using dmi based quirk for that machine.
Stefan Sauer [Thu, 6 Jan 2022 12:41:45 +0000 (13:41 +0100)]
ALSA: seq: virmidi: Add a drain operation
If a driver does not supply a drain operation for outputs, a default code
path will execute msleep(50). Especially for a virtual midi device
this severely limmits the throughput.
This implementation for the virtual midi driver simply flushes the output
workqueue.
Shengjiu Wang [Wed, 5 Jan 2022 11:08:03 +0000 (19:08 +0800)]
ASoC: fsl_asrc: refine the check of available clock divider
According to RM, the clock divider range is from 1 to 8, clock
prescaling ratio may be any power of 2 from 1 to 128.
So the supported divider is not all the value between
1 and 1024, just limited value in that range.
Create table for the supported divder and add function to
check the clock divider is available by comparing with
the table.
Hans de Goede [Thu, 6 Jan 2022 11:01:28 +0000 (12:01 +0100)]
ASoC: Intel: bytcr_rt5640: Add support for external GPIO jack-detect
Some boards have the codec IRQ hooked-up as normally, so the driver can
still do things like headset vs headphones and button-press detection,
but instead of using one of the JD pins of the codec, an external GPIO
is used to report the jack-presence switch status of the jack.
Add support for boards which have this setup and which specify which
external GPIO to use in the special Android AMCR0F28 ACPI device.
And add a quirk for the Asus TF103C tablet which uses this setup.
Hans de Goede [Thu, 6 Jan 2022 11:01:27 +0000 (12:01 +0100)]
ASoC: Intel: bytcr_rt5640: Support retrieving the codec IRQ from the AMCR0F28 ACPI dev
Some X86 tablets, which ship with Android as factory installed OS,
specify codec IRQs/GPIOS in a special Android AMCR0F28 ACPI device.
Add support for retrieving the codec IRQ from this ACPI device instead
of from the 10EC5640 device describing the codec itself and enable this
on Asus MemoPad 7 ME176C tablets.
This fixes jack-detect not working on these tablets.
Hans de Goede [Thu, 6 Jan 2022 11:01:26 +0000 (12:01 +0100)]
ASoC: rt5640: Add support for boards with an external jack-detect GPIO
Some boards have the codec IRQ hooked-up as normally, so the driver can
still do things like headset vs headphones and button-press detection,
but instead of using one of the JD pins of the codec, an external GPIO
is used to report the jack-presence switch status of the jack.
Hans de Goede [Thu, 6 Jan 2022 11:01:25 +0000 (12:01 +0100)]
ASoC: rt5640: Allow snd_soc_component_set_jack() to override the codec IRQ
On some boards where the firmware/fwnode information is in essence
read-only (x86 + ACPI boards) the i2c_client for the codec may contain
the wrong IRQ or no IRQ at all.
Since we only request the IRQ once snd_soc_component_set_jack() gets
called, allow machine drivers to override the IRQ with the proper one
through the data parameter to snd_soc_component_set_jack().
Hans de Goede [Thu, 6 Jan 2022 11:01:24 +0000 (12:01 +0100)]
ASoC: rt5640: Change jack_work to a delayed_work
Change jack_work from a struct work_struct to a struct delayed_work, this
is a preparation patch for adding support for boards where an external
GPIO is used for jack-detect, rather then one of the JD pins of the codec.
Hans de Goede [Thu, 6 Jan 2022 11:01:23 +0000 (12:01 +0100)]
ASoC: rt5640: Fix possible NULL pointer deref on resume
Commit d006f488f556 ("ASoC: rt5640: Add the HDA header support") adds
re-queuing of the jack_work on resume when rt5640->jd_src != 0.
But the jack_work will unconditionally deref rt5640->jack and that might
be NULL. E.g. the sound/soc/intel/boards/bytcr_rt5640.c machine driver
call snd_soc_component_set_jack(codec, NULL, NULL) from pre_suspend to
disable the IRQ to avoid spurious wakeups, so when rt5640_resume()
runs rt5640->jack will be NULL in this case.
Make the queueing of the work conditional on rt5640->jack instead of
on rt5640->jd_src to fix this.
Fixes: d006f488f556 ("ASoC: rt5640: Add the HDA header support") Cc: Oder Chiou <oder_chiou@realtek.com> Signed-off-by: Hans de Goede <hdegoede@redhat.com> Link: https://lore.kernel.org/r/20220106110128.66049-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie@kernel.org>
Shengjiu Wang [Tue, 4 Jan 2022 10:40:35 +0000 (18:40 +0800)]
ASoC: imx-card: improve the sound quality for low rate
According to RM, on auto mode:
For codec AK4458 and AK4497, the lowest ratio of MLCK/FS is 256
if sample rate is 8kHz-48kHz,
For codec AK5558, the lowest ratio of MLCK/FS is 512 if sample
rate is 8kHz-48kHz.
With these setting the sound quality for 8kHz-48kHz can be improved.
Shengjiu Wang [Tue, 4 Jan 2022 10:40:34 +0000 (18:40 +0800)]
ASoC: imx-card: Fix mclk calculation issue for akcodec
Transfer the refined slots and slot_width to akcodec_get_mclk_rate()
for mclk calculation, otherwise the mclk frequency does not match
with the slots and slot_width for S16_LE format, because the default
slot_width is 32.
Shengjiu Wang [Tue, 4 Jan 2022 10:40:33 +0000 (18:40 +0800)]
ASoC: imx-card: Need special setting for ak4497 on i.MX8MQ
The SAI on i.MX8MQ don't support one2one ratio for mclk:bclk, so
the mclk frequency exceeds the supported range of codec for
the case that sample rate is larger than 705kHZ and format is
S32_LE. Update the supported width for such case.
The speaker fixup that is used for the Yoga 7 14ITL5 also applies to
the IdeaPad Slim 9i 14ITL5. The attached patch applies the quirk to
initialise the amplifier on the IdeaPad Slim 9i as well.
Takashi Iwai [Wed, 5 Jan 2022 16:24:09 +0000 (17:24 +0100)]
ASoC: ak4375: Fix unused function error
A randconfig caught a compile warning that is now treated as a fatal
error:
sound/soc/codecs/ak4375.c:415:13: error: ‘ak4375_power_off’ defined but not used [-Werror=unused-function]
where ak4375_power_off() is used only from the PM handler.
As both suspend and resumes are already marked with __maybe_unused,
let's rip off the superfluous ifdef CONFIG_PM, so that the error above
can be avoided.
Mark Brown [Wed, 5 Jan 2022 15:59:21 +0000 (15:59 +0000)]
Add low power hibernation support to cs35l41
Merge series from Charles Keepax <ckeepax@opensource.cirrus.com>:
This patch series adds support for the low power hibernation feature
on cs35l41. This allows the DSP memory to be retained whilst the
device enters a very low power state.
Baole Fang [Wed, 5 Jan 2022 14:08:54 +0000 (22:08 +0800)]
ALSA: hda/realtek: Add quirk for Legion Y9000X 2020
Legion Y9000X 2020 has a speaker, but the speaker doesn't work.
This can be fixed by applying alc285_fixup_ideapad_s740_coef
to fix the speaker's coefficients.
Besides, to support the transition between the speaker and the headphone,
alc287_fixup_legion_15imhg05_speakers needs to be run.
Takashi Iwai [Wed, 5 Jan 2022 14:39:24 +0000 (15:39 +0100)]
Merge tag 'asoc-v5.17' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v5.17
Not much going on framework release this time, but a big update for
drivers especially the Intel and SOF ones.
- Refinements and cleanups around the delay() APIs.
- Wider use of dev_err_probe().
- Continuing cleanups and improvements to the SOF code.
- Support for pin switches in simple-card derived cards.
- Support for AMD Renoir ACP, Asahi Kasei Microdevices AKM4375, Intel
systems using NAU8825 and MAX98390, Mediatek MT8915, nVidia Tegra20
S/PDIF, Qualcomm systems using ALC5682I-VS and Texas Instruments
TLV320ADC3xxx.
Fabio Estevam [Tue, 4 Jan 2022 18:06:13 +0000 (15:06 -0300)]
ASoC: cs4265: Add a remove() function
When the reset_gpio GPIO is used, it is better to put the codec
back into reset state when the driver unbinds.
Add a remove() function to accomplish that.
Suggested-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Fabio Estevam <festevam@denx.de> Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com> Link: https://lore.kernel.org/r/20220104180613.639317-1-festevam@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Wed, 5 Jan 2022 11:30:24 +0000 (11:30 +0000)]
ASoC: wm_adsp: Add support for "toggle" preloaders
In the case a device can support retaining the firmware memory across
low power states it is useful for the preloader widget to only power up
whilst actually loading/unloading the core, as opposed to the normal
operation where the widget is powered for the entire time a firmware is
preloaded onto the core. Add support for this mode and a flag to enable
it.
Charles Keepax [Wed, 5 Jan 2022 11:30:23 +0000 (11:30 +0000)]
firmware: cs_dsp: Clear core reset for cache
If the Halo registers are kept in the register cache the
HALO_CORE_RESET bit will be retained as 1 after reset is triggered in
cs_dsp_halo_start_core. This will cause subsequent writes to reset
the core which is not desired. Apart from this bit the rest of the
register bits are cacheable, so for safety sake clear the bit to
ensure the cache is consistent.
Charles Keepax [Wed, 5 Jan 2022 11:30:22 +0000 (11:30 +0000)]
ASoC: cs35l41: Correct handling of some registers in the cache
It makes no sense to cache the test/user key registers, since they
require values written at specific times, mark them volatile. It is
probably best if they can't be accessed from user-space either, so
mark them precious as well.
The interrupt force, edge, polarity and debounce are all settings
applied to the IRQ rather than status bits and as such should not be
volatile.
The OTP trim values will require re-application in the event of a
cache sync and as such should not be volatile. The OTPID however
should be volatile.
The DSP scratch registers are used to read back an error/debug code
from the DSP on shutdown, as such these should be marked volatile.
Finally, add some missing defaults, add TST_FS_MON0, and allow the
DSP core control register to be cached.
Charles Keepax [Wed, 5 Jan 2022 11:30:21 +0000 (11:30 +0000)]
ASoC: cs35l41: Correct DSP power down
The wm_adsp_event should be called before the early_event on power
down, event stops the core running and early_event then powers down
the core. Additionally, the core should only be stopped if it was
actually running in the first place.
The suspend code unconditionally sets ->hp_jack_in and ->mic_jack_in
to zero but without reporting this status change to the HDA core.
To compensate for this, always assume a status change on the
first unsol event after boot or resume.
ALSA: hda/cs8409: Increase delay during jack detection
Commit d0d8008366cc reduced delays related to cs42l42 jack
detection. However, the change was too aggressive. As a result
internal speakers on DELL Inspirion 3501 are not detected.
Increase the delay in cs42l42_run_jack_detect() a bit.
ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Master after reboot from Windows
This patch addresses an issue where after rebooting from Windows into Linux
there would be no audio output.
It turns out that the Realtek Audio driver on Windows changes some coeffs
which are not being reset/reinitialized when rebooting the machine. As a
result, there is no audio output until these coeffs are being reset to
their initial state. This patch takes care of that by setting known-good
(initial) values to the coeffs.
We initially relied upon alc1220_fixup_clevo_p950() to fix some pins in the
connection list. However, it also sets coef 0x7 which does not need to be
touched. Furthermore, to prevent mixing device-specific quirks I introduced
a new alc1220_fixup_gb_x570() which is heavily based on
alc1220_fixup_clevo_p950() but does not set coeff 0x7 and fixes the coeffs
that are actually needed instead.
This new alc1220_fixup_gb_x570() is believed to also work for other boards,
like the Gigabyte X570 Aorus Extreme and the newer Gigabyte Aorus X570S
Master. However, as there is no way for me to test these I initially only
enable this new behaviour for the mainboard I have which is the Gigabyte
X570(non-S) Aorus Master.
I tested this patch on the 5.15 branch as well as on master and it is
working well for me.
Sameer Pujar [Thu, 23 Dec 2021 11:53:49 +0000 (17:23 +0530)]
ALSA: hda/tegra: Fix Tegra194 HDA reset failure
HDA regression is recently reported on Tegra194 based platforms.
This happens because "hda2codec_2x" reset does not really exist
in Tegra194 and it causes probe failure. All the HDA based audio
tests fail at the moment. This underlying issue is exposed by
commit 06ec15eb8463 ("reset: tegra-bpmp: Handle errors in BPMP
response") which now checks return code of BPMP command response.
Fix this issue by skipping unavailable reset on Tegra194.
Mark Brown [Fri, 31 Dec 2021 14:38:47 +0000 (14:38 +0000)]
ASoC: mediatek: mt8195: repair pcmif BE dai
Merge series from Trevor Wu <trevor.wu@mediatek.com>:
This series of patches repairs some problems for pcmif BE dai.
The unexpected control flow is corrected, and the missing playback
support of DPCM is added.
Lucas Tanure [Fri, 17 Dec 2021 11:57:02 +0000 (11:57 +0000)]
ASoC: cs35l41: Create shared function for errata patches
ASoC and HDA systems require the same errata patches, so
move it to the shared code using a function the correctly
applies the patches by revision
Also, move CS35L41_DSP1_CCM_CORE_CTRL write to errata
patch function as is required to be written at boot,
but not in regmap_register_patch sequence as will affect
waking up from hibernation
Lucas Tanure [Fri, 17 Dec 2021 11:56:59 +0000 (11:56 +0000)]
ASoC: cs35l41: Convert tables to shared source code
To support CS35L41 in HDA systems the HDA driver
for CS35L41 would have to duplicate some functions
that already exist on ASoC driver
So instead of duplicate the code, use the new lib
source as a shared resource for both ASoC and HDA
Also, change the way CONFIG_SND_SOC_CS35L41 is
selected, as reported by Intel Kernel test robot,
it is possible to build SND_SOC_CS35L41_SPI/I2C
without the main driver, which would lead to build
failures.
Trevor Wu [Thu, 30 Dec 2021 08:47:30 +0000 (16:47 +0800)]
ASoC: mediatek: mt8195: correct pcmif BE dai control flow
Originally, the conditions for preventing reentry are not correct.
dai->component->active is not the state specifically for pcmif dai, so it
is not a correct condition to indicate the status of pcmif dai.
On the other hand, snd_soc_dai_stream_actvie() in prepare ops for both
playback and capture possibly return true at the first entry when these
two streams are opened at the same time.
In the patch, I refer to the implementation in mt8192-dai-pcm.c.
Clock and enabling bit for PCMIF are managed by DAPM, and the condition
for prepare ops is replaced by the status of dai widget.
Derek Fang [Mon, 27 Dec 2021 05:54:46 +0000 (13:54 +0800)]
ASoC: rt5682: Register wclk with its parent_hws instead of parent_data
The mclk might not be registered as a fixed clk name "mclk" on some
platforms.
In those platforms, if the mclk needed to be controlled by codec driver
and acquired by a fixed name, it would be a problem.
This patch to fix the issue that wclk becomes an orphan due to the fixed
mclk's name.
Trevor Wu [Tue, 28 Dec 2021 06:48:21 +0000 (14:48 +0800)]
ASoC: mediatek: mt8195: update control for RT5682 series
Playback pop is observed and the root cause is the reference clock
provided by MT8195 is diabled before RT5682 finishes the control flow.
To ensure the reference clock supplied to RT5682 is disabled after RT5682
finishes all register controls. We replace BCLK with MCLK for RT5682
reference clock, and makes use of set_bias_level_post to handle MCLK
which guarantees MCLK is off after all RT5682 register access.
Jiasheng Jiang [Tue, 28 Dec 2021 03:40:26 +0000 (11:40 +0800)]
ASoC: samsung: idma: Check of ioremap return value
Because of the potential failure of the ioremap(), the buf->area could
be NULL.
Therefore, we need to check it and return -ENOMEM in order to transfer
the error.
Fixes: 65a314121ee2 ("ASoC: SAMSUNG: Add I2S0 internal dma driver") Signed-off-by: Jiasheng Jiang <jiasheng@iscas.ac.cn> Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@canonical.com> Link: https://lore.kernel.org/r/20211228034026.1659385-1-jiasheng@iscas.ac.cn Signed-off-by: Mark Brown <broonie@kernel.org>
Fabio Estevam [Wed, 22 Dec 2021 14:19:19 +0000 (11:19 -0300)]
ASoC: cs4265: Fix part number ID error message
The Chip ID - Register 01h contains the following description
as per the CS4265 datasheet:
"Bits 7 through 4 are the part number ID, which is 1101b (0Dh)"
The current error message is incorrect as it prints CS4265_CHIP_ID,
which is the register number, instead of printing the expected
part number ID value.
To make it clearer, also do a shift by 4, so that the error message
would become:
[ 4.218083] cs4265 1-004f: CS4265 Part Number ID: 0x0 Expected: 0xd
Arie Geiger [Thu, 23 Dec 2021 23:28:57 +0000 (15:28 -0800)]
ALSA: hda/realtek: Add speaker fixup for some Yoga 15ITL5 devices
This patch adds another possible subsystem ID for the ALC287 used by
the Lenovo Yoga 15ITL5.
It uses the same initalization as the others.
This patch has been tested and works for my device.
Kai Vehmanen [Thu, 23 Dec 2021 07:34:23 +0000 (09:34 +0200)]
ALSA: hda: Add AlderLake-N PCI ID
Add HD Audio PCI ID for Intel AlderLake-N. Add rules to
snd_intel_dsp_find_config() to choose DSP-based SOF driver for ADL-N
systems with PCH-DMIC or Soundwire codecs, and plain HDA driver for the
rest (DSP not used).
Ville Syrjälä [Wed, 22 Dec 2021 14:53:50 +0000 (16:53 +0200)]
ALSA: hda/hdmi: Disable silent stream on GLK
The silent stream stuff recurses back into i915 audio
component .get_power() from the .pin_eld_notify() hook.
On GLK this will deadlock as i915 may already be holding
the relevant modeset locks during .pin_eld_notify() and
the GLK audio vs. CDCLK workaround will try to grab the
same locks from .get_power().
Until someone comes up with a better fix just disable the
silent stream support on GLK.
Mark Brown [Fri, 17 Dec 2021 13:02:12 +0000 (13:02 +0000)]
kselftest: alsa: Factor out check that values meet constraints
To simplify the code a bit and allow future reuse factor the checks that
values we read are valid out of test_ctl_get_value() into a separate
function which can be reused later. As part of this extend the test to
check all the values for the control, not just the first one.
Mark Brown [Fri, 24 Dec 2021 16:15:47 +0000 (16:15 +0000)]
ASoC/SoundWire: improve suspend flows and use set_stream() instead of set_tdm_slots() for HDAudio
Merge series from Bard Liao <yung-chuan.liao@linux.intel.com>:
This series contains three topics.
1. SoundWire: Intel: remove pdm support
2. ASoC/SoundWire: dai: expand 'stream' concept beyond SoundWire
3. ASoC/SOF/SoundWire: fix suspend-resume on pause with dynamic pipelines
The topics are independent but the changes are dependent. So please
allow me to send them in one series.
ASoC: amd: acp: Power on/off the speaker enable gpio pin based on DAPM callback.
Configure the speaker gpio pin based on power sequence of the DAPM
speaker events.
Enable speaker after widget power up and Disable before widget powerdown.
ASoC: Intel/SOF: use set_stream() instead of set_tdm_slots() for HDAudio
Overloading the tx_mask with a linear value is asking for trouble and
only works because the codec_dai hw_params() is called before the
cpu_dai hw_params().
Move to the more generic set_stream() API to pass the hdac_stream
information.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Link: https://lore.kernel.org/r/20211224021034.26635-6-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
The HDAudio ASoC support relies on the set_tdm_slots() helper to store
the HDaudio stream tag in the tx_mask. This only works because of the
pre-existing order in soc-pcm.c, where the hw_params() is handled for
codec_dais *before* cpu_dais. When the order is reversed, the
stream_tag is used as a mask in the codec fixup functions:
/* fixup params based on TDM slot masks */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
codec_dai->tx_mask)
soc_pcm_codec_params_fixup(&codec_params,
codec_dai->tx_mask);
As a result of this confusion, the codec_params_fixup() ends-up
generating bad channel masks, depending on what stream_tag was
allocated.
We could add a flag to state that the tx_mask is really not a mask,
but it would be quite ugly to persist in overloading concepts.
Instead, this patch suggests a more generic get/set 'stream' API based
on the existing model for SoundWire. We can expand the concept to
store 'stream' opaque information that is specific to different DAI
types. In the case of HDAudio DAIs, we only need to store a stream tag
as an unsigned char pointer. The TDM rx_ and tx_masks should really
only be used to store masks.
Rename get_sdw_stream/set_sdw_stream callbacks and helpers as
get_stream/set_stream. No functionality change beyond the rename.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@intel.com> Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com> Acked-By: Vinod Koul <vkoul@kernel.org> Link: https://lore.kernel.org/r/20211224021034.26635-5-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
This patch provides both a simplification of the suspend flows and a
better balanced operation during suspend/resume transition, as part of
the transition of Sound Open Firmware (SOF) to dynamic pipelines: the
DSP resources are only enabled when required instead of enabled on
startup.
The exiting code relies on a convoluted way of dealing with suspend
signals. Since there is no .suspend DAI callback, we used the
component .suspend and marked all the component DAI dmas as
'suspended'. The information was used in the .prepare stage to
differentiate resume operations from xrun handling, and only
reinitialize SHIM registers and DMA in the former case.
While this solution has been working reliably for about 2 years, there
is a much better solution consisting in trapping the TRIGGER_SUSPEND
in the .trigger DAI ops. The DMA is still marked in the same way for
the .prepare op to run, but in addition the callbacks sent to DSP
firmware are now balanced.
Normal operation:
hw_params -> intel_params_stream
hw_free -> intel_free_stream
This balanced operation was not required with existing SOF firmware
relying on static pipelines instantiated at every boot. With the
on-going transition to dynamic pipelines, it's however a requirement
to keep the use count for the DAI widget balanced across all
transitions.
The component suspend is not removed but instead modified to deal with
a corner case: when a substream is PAUSED, the ALSA core does not
throw the TRIGGER_SUSPEND. This is problematic since the refcount for
all pipelines and widgets is not balanced, leading to issues on
resume. The trigger callback keeps track of the 'paused' state with a
new flag, which is tested during the component suspend called later to
release the remaining DSP resources. These resources will be
re-enabled in the .prepare step.
The IPC used in the TRIGGER_SUSPEND to release DSP resources is not a
problem since the BE dailink is already marked as non-atomic.
ASoC/soundwire: intel: simplify callbacks for params/hw_free
We don't really need to pass a substream to the callback, we only need
the direction. No functionality change, only simplification to enable
improve suspend with paused streams.
Mark Brown [Thu, 23 Dec 2021 19:38:13 +0000 (19:38 +0000)]
ASoC: qcom: Parse "pin-switches" and "widgets" from DT
Merge series from Stephan Gerhold <stephan@gerhold.net>:
Some sound card setups might require extra pin switches to allow
turning off certain audio components. simple-card supports this
already using the "pin-switches" and "widgets" device tree property.
This series makes it possible to use the same properties for the Qcom
sound cards.
To implement that, the function that parses the "pin-switches" property
in simple-card-utils.c is first moved into the ASoC core. Then two
simple function calls are added to the common Qcom sound card DT parser.
Finally there is a small patch for the msm8916-wcd-analog codec to make
it possible to model sound card setups used in some MSM8916 smartphones.
(See PATCH 2/4 for an explanation of some real example use cases.)
Using pin switches rather than patching codec drivers with switches was
originally suggested by Mark Brown on a patch for the tfa989x codec:
https://lore.kernel.org/alsa-devel/YXaMVHo9drCIuD3u@sirena.org.uk/
Stephan Gerhold [Tue, 14 Dec 2021 14:20:49 +0000 (15:20 +0100)]
ASoC: msm8916-wcd-analog: Use separate outputs for HPH_L/HPH_R
The analog codec has separate output paths for the left headphone channel
(HPH_L) and the right headphone channel (HPH_R). While they are usually
used together for actual headphones output, some devices also have an
analog speaker amplifier connected to one of the headphone channels.
To allow modelling that properly (and to avoid powering on the unneeded
output path), HPH_L and HPH_R should be represented by separate outputs
rather than a shared HEADPHONE output that always activates both paths.
Stephan Gerhold [Tue, 14 Dec 2021 14:20:48 +0000 (15:20 +0100)]
ASoC: qcom: common: Parse "pin-switches" and "widgets" from DT
Use the DT helpers in the ASoC core to parse the "pin-switches" and
"widgets" properties from the device tree. This allows adding extra
mixers to disable e.g. an extra speaker amplifier that would be
normally powered on automatically because it is connected to a shared
output pin.
Stephan Gerhold [Tue, 14 Dec 2021 14:20:47 +0000 (15:20 +0100)]
ASoC: dt-bindings: qcom: sm8250: Document "pin-switches" and "widgets"
Some sound card setups might require extra pin switches to allow
turning off certain audio components. There are two real examples for
this in smartphones/tablets based on MSM8916:
1. Analog speaker amplifiers connected to headphone outputs.
The MSM8916 analog codec does not have a separate "Line Out" port
so some devices have an analog speaker amplifier connected to one
of the headphone outputs. A pin switch is necessary to allow
playback on headphones without also activating the speaker.
2. External speaker codec also used as earpiece.
Some smartphones have two front-facing (stereo) speakers that can
be also configured to act as an earpiece during voice calls. A pin
switch is needed to allow disabling the second speaker during
voice calls.
There are existing bindings that allow setting up such pin switches in
simple-card.yaml. Document the same for Qcom sound cards.
One variant of example 1 above is added to the examples in the DT
schema: There is an analog speaker amplifier connected to the HPH_R
(right headphone channel) output. Adding a "Speaker" pin switch and
widget allows turning off the speaker when audio should be only played
via the connected headphones.
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