Dan Murphy [Fri, 21 Feb 2020 18:13:58 +0000 (12:13 -0600)]
ASoC: tlv320adcx140: Add decimation filter support
Add decimation filter selection support.
Per Section 8.3.6.7 the Digital Decimation Filter is selectable between
a Linear Phase, Low Latency, and Ultra Low Latency filer.
Dan Murphy [Fri, 21 Feb 2020 18:13:57 +0000 (12:13 -0600)]
ASoC: tlv320adcx140: Add DRE and AGC support
The TLV320ADCx140 parts support Dynamic Range Enhancer (DRE) as defined
in Section 8.3.2 of the data sheets.
The DRE achieves a complete-channel dynamic range as high as 120 dB.
At a system level, the DRE scheme enables far-field, high-fidelity recording
of audio signals in very quiet environments and low-distortion recording in
loud environments.
There are 2 enables for DRE. The first is a global setting that enables
the DRE engine in the device and the other enable is per channel. If
the DRE is enabled globally then either DRE or AGC can be used per each
configured channel. If global DRE is disabled then even setting the DRE
enable bit in the channel config register will have no effect.
Jerome Brunet [Fri, 21 Feb 2020 15:36:06 +0000 (16:36 +0100)]
ASoC: meson: g12a: add internal DAC glue driver
Add support for the internal audio DAC glue found on the Amlogic g12a
and sm1 SoC families. This allows to connect the TDM outputs of the SoC
to the internal t9015 audio DAC.
Dan Murphy [Fri, 21 Feb 2020 12:41:51 +0000 (06:41 -0600)]
ASoC: tas2562: Add support for digital volume control
Add support for digital volume control. There is no dedicated register
for volume control but instead there are 4. The values of the registers
are determined with exponential floating point math.
So a table was created with register values for 2dB step increments
from -110dB to 0dB.
Samuel Holland [Mon, 17 Feb 2020 06:42:20 +0000 (00:42 -0600)]
ASoC: sun8i-codec: Remove unused dev from codec struct
This field is not used anywhere in the driver, so remove it.
Fixes: 36c684936fae ("ASoC: Add sun8i digital audio codec") Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Chen-Yu Tsai <wens@csie.org> Link: https://lore.kernel.org/r/20200217064250.15516-5-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families
The AIU audio controller on the Meson8 and Meson8b SoC families is
compatible with the one found in the later GXBB family. Add compatible
strings for these two older SoC families so the driver can be loaded for
them.
Instead of using the I2S divider from the AIU_CLK_CTRL_MORE register we
need to use the I2S divider from the AIU_CLK_CTRL register. This older
register is less flexible because it only supports four divider settings
(1, 2, 4, 8) compared to the AIU_CLK_CTRL_MORE register (which supports
dividers in the range 0..64).
ASoC: meson: aiu: introduce a struct for platform specific information
Introduce a struct aiu_platform_data to make the driver aware of
platform specific information. Convert the existing check for the
internal stereo audio codec (only available on GXL) to this new struct.
Support for the 32-bit SoCs will need this as well because the
AIU_CLK_CTRL_MORE register doesn't have the I2S divider bits (and we
need to use the I2S divider from AIU_CLK_CTRL instead).
ASoC: meson: aiu: Document Meson8 and Meson8b support in the dt-bindings
The AIU audio output controller on the Meson8 and Meson8b SoC families
is compatible with the one found in the GXBB family. Document the
compatible string for these two older SoCs.
Dan Murphy [Thu, 20 Feb 2020 21:07:59 +0000 (15:07 -0600)]
ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family
Add the tlv320adcx140 codec driver family.
The TLV320ADCx140 is a Burr-Brown™ highperformance, audio analog-to-digital
converter (ADC) that supports simultaneous sampling of up to four analog
channels or eight digital channels for the pulse density modulation (PDM)
microphone input. The device supports line and microphone inputs, and
allows for both single-ended and differential input configurations.
Kai Vehmanen [Thu, 20 Feb 2020 17:10:28 +0000 (19:10 +0200)]
ASoC: SOF: Intel: hda: allow operation without i915 gfx
Add support to configure the HDA controller with an external HDA
codec even if iDisp codec in i915 is not available.
This can happen for multiple reasons:
- internal graphics is disabled on the system
- i915 driver is not enabled in kernel or it fails to init
- i915 codec reports error in HDA codec probe
- HDA codec driver probe fails
Address all these scenarios, but keep using the existing topology.
In case failures occur, HDMI PCM nodes are created, but they will
report error if application tries to use them. No ALSA mixer controls
are created. If the external HDA codec init fails as well, SOF probe
will return error as before.
Kai Vehmanen [Thu, 20 Feb 2020 17:10:27 +0000 (19:10 +0200)]
ASoC: intel/skl/hda - add no-HDMI cases to generic HDA driver
Extend the generic HDA driver to support systems where iDisp/HDMI
audio codecs are disabled for some reason. Switch codecs to
SoC dummy in the affected DAI links. This allows to reuse
existing topologies for this case.
Charles Keepax [Thu, 20 Feb 2020 12:56:54 +0000 (12:56 +0000)]
ASoC: samsung: Update dependencies for Arizona machine drivers
Currently it is possible to get the following bad config:
WARNING: unmet direct dependencies detected for SND_SOC_WM5110
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MFD_WM5110 [=n]
commit ea00d95200d0 ("ASoC: Use imply for SND_SOC_ALL_CODECS")
commit d8dd3f92a6ba ("ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout")
After these two patches the machine drivers still selects the
SND_SOC_WM5110 symbol which doesn't take account of the dependency
added on the MFD_WM5110 symbol, fix this by also adding a dependency on
MFD_WM5110 itself.
Jerome Brunet [Wed, 19 Feb 2020 11:50:48 +0000 (12:50 +0100)]
ASoC: dpcm: remove confusing trace in dpcm_get_be()
Now that dpcm_get_be() is used in dpcm_end_walk_at_be(), it is not a error
if this function does not find a BE for the provided widget. Remove the
related dev_err() trace which is confusing since things might be working
as expected.
When called from dpcm_add_paths(), it is an error if dpcm_get_be() fails to
find a BE for the provided widget. The necessary error trace is already
done in this case.
Fixes: 027a48387183 ("ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()") Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/20200219115048.934678-1-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie@kernel.org>
Robin Murphy [Tue, 18 Feb 2020 21:31:59 +0000 (21:31 +0000)]
ASoC: rockchip: Make RK3328 GPIO_MUTE control explicit
The RK3328 reference design uses an external line driver IC as a buffer
on the analog codec output, enabled by the GPIO_MUTE pin, and such a
configuration is currently assumed in the codec driver's direct poking
of GRF_SOC_CON10 to control the GPIO_MUTE output value. However, some
boards wire up analog audio yet use that pin for some other purpose, so
that assumption doesn't always hold. Update this functionality to rely
on an explicit GPIO descriptor, such that it can be managed at the
board level.
Robin Murphy [Tue, 18 Feb 2020 21:31:58 +0000 (21:31 +0000)]
ASoC: dt-bindings: Make RK3328 codec GPIO explicit
Existing RK3328 codec drivers have overloaded the GRF phandle to assume
implicit control of the limited-function GPIO_MUTE pin, which is usually
used to enable an external audio line driver IC. Since this pin has a
proper binding of its own (see gpio/rockchip,rk3328-grf-gpio.txt), make
a GPIO explicit in the codec binding too. This will help avoid ambiguity
on boards that use that pin for some other purpose.
(and while touching the example, enforce the "don't include status" rule)
Dan Murphy [Wed, 19 Feb 2020 13:46:22 +0000 (07:46 -0600)]
ASoC: tas2562: Add support for ISENSE and VSENSE
Add additional support for ISENSE and VSENSE feature for the TAS2562.
This feature monitors the output to the loud speaker attempts to
eliminate IR drop errors due to packaging.
This feature is defined in Section 8.4.5 IV Sense of the data sheet.
This change might have been desirable to ensure the uniqueness of
the component name. It would have helped to better support linux
devices which register multiple components, something is which more
common than initially thought.
However, some card driver are directly using dev_name() to fill the
component names of the dai_link which is a problem if want to change
the way ASoC generates the component names.
Until we figure out the appropriate way to deal with this, revert the
change and keep the names as they were. There might be a couple of warning
related to debugfs (which were already present before the change) but it
is still better than breaking working audio cards.
ASoC: soc-pcm: merge playback/cature_active into stream_active
DAI has playback_active and capture_active to care usage count.
OTOH, we have SNDRV_PCM_STREAM_PLAYBACK/CAPTURE.
But because of this kind of implementation mismatch,
ALSA SoC has many verbose code.
To solve this issue, this patch merge playback_active/capture_active
into stream_active[2];
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/875zg5botu.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
When we use some kind of lock, we need to do unlock.
In that time, multi unlock/return is not good implementation.
This patch add label and use goto at dpcm_fe_dai_open()
to reduce such code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/877e0lboty.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds common snd_soc_dpcm_check_state(), and use it from
snd_soc_dpcm_can_be_free_stop() / snd_soc_dpcm_can_be_params().
It can reduce duplicate code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/878sl1bou2.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
No one is using snd_soc_dpcm_be_get/set_state().
If it exists only by assumption that "it may be necessary someday",
let's remove it now. Otherwise code maintenance will be difficult.
We can revive it when we really needed it.
Let's remove it, so far.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/87a75hbou7.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
No one is using soc_dpcm_be_digital_mute().
If it exists only by assumption that "it may be necessary someday",
let's remove it now. Otherwise code maintenance will be difficult.
We can revive it when we really needed it.
Let's remove it, so far.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/87blpxbouc.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be()
dpcm_end_walk_at_be() has very duplicate code.
dpcm_end_walk_at_be() {
...
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
(1) /* code for Playback */
} else {
(2) /* code for Capture */
}
}
The difference between Playback (1) and Capture (2) code is pointer only
(= "playback_widget" or "caputre_widget").
OTOH, now we already has dai_get_widget() for it.
This means we can merge (1) and (2).
This patch do it and remove duplicated code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/87eeutboul.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be()
dpcm_get_be() has very duplicate code.
dpcm_get_be() {
...
if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
(1) /* code for Playback */
} else {
(2) /* code for Capture */
}
}
The difference between Playback (1) and Capture (2) code is pointer only
(= "playback_widget" or "caputre_widget").
OTOH, now we already has dai_get_widget() for it.
This means we can merge (1) and (2).
This patch do it and remove duplicated code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/87ftf9bouq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
Derek Fang [Tue, 18 Feb 2020 13:51:51 +0000 (21:51 +0800)]
ASoC: rt5682: Add CCF usage for providing I2S clks
There is a need to use RT5682 as DAI clock master for other codecs
within a platform, which means that the DAI clocks are required to
remain, regardless of whether the RT5682 is actually running
playback/capture.
The RT5682 CCF basic functions are implemented almost by the existing
internal functions and asoc apis. It needs a clk provider (rt5682 mclk)
to generate the bclk and wclk outputs.
The RT5682 CCF supports and restricts as below:
1. Fmt of DAI-AIF1 must be configured to master before using CCF.
2. Only accept a 48MHz clk as the clk provider.
3. Only provide a 48kHz wclk and a set of multiples of wclk as bclk.
There are some temporary limitations in this patch until a better
implementation.
Cezary Rojewski [Tue, 18 Feb 2020 14:39:23 +0000 (15:39 +0100)]
ASoC: SOF: Provide probe debugfs support
Define debugfs subdirectory delegated for IPC communication with DSP.
Input format: uint,uint,(...) which are later translated into DWORDS
sequence and further into instances of struct of interest given the IPC
type.
For Extractor probes, following have been enabled:
- PROBE_POINT_ADD (echo <..> probe_points)
- PROBE_POINT_REMOVE (echo <..> probe_points_remove)
- PROBE_POINT_INFO (cat probe_points)
Cezary Rojewski [Tue, 18 Feb 2020 14:39:22 +0000 (15:39 +0100)]
ASoC: SOF: Intel: Probe compress operations
Add HDA handlers for soc_compr_ops and snd_compr_ops which cover probe
related operations. Implementation supports both connection purposes.
These merely define stream setups as core flow is covered by SOF
compress core.
Cezary Rojewski [Tue, 18 Feb 2020 14:39:21 +0000 (15:39 +0100)]
ASoC: SOF: Intel: Expose SDnFMT helpers
Hda stream is setup in similar fashion for compress as it is for pcm
operations. To reuse existing code in compress path, expose SDnFMT
helper routines.
Cezary Rojewski [Tue, 18 Feb 2020 14:39:20 +0000 (15:39 +0100)]
ASoC: SOF: Generic probe compress operations
Define system-agnostic probe compress flow which serves as a base for
actual, hardware-dependent implementations.
As per firmware spec, maximum of one extraction stream is allowed, while
for injection, there can be plenty.
Apart from probe_pointer, all probe compress operations are mandatory.
Copy operation is defined as unified as its flow should be shared across
all SOF systems.
Cezary Rojewski [Tue, 18 Feb 2020 14:39:19 +0000 (15:39 +0100)]
ASoC: SOF: Implement Probe IPC API
Add all required types and methods to support each and every request
that driver could sent to firmware. Probe is one of SOF firmware
features which allows for data extraction and injection directly from
or to DMA stream.
Exposes eight IPCs:
- addition and removal of injection DMAs
- addition and removal of probe points
- info retrieval of injection DMAs and probe points
- probe initialization and cleanup
Cezary Rojewski [Tue, 18 Feb 2020 14:39:17 +0000 (15:39 +0100)]
ALSA: core: Implement compress page allocation and free routines
Add simple malloc and free methods for memory management for compress
streams. Based on snd_pcm_lib_malloc_pages and snd_pcm_lib_free_pages
implementation.
Cezary Rojewski [Tue, 18 Feb 2020 14:39:16 +0000 (15:39 +0100)]
ALSA: core: Expand DMA buffer information
Update DMA buffer definition for snd_compr_runtime so it is represented
similarly as in snd_pcm_runtime. While at it, modify
snd_compr_set_runtime_buffer to account for newly added members.
Stephan Gerhold [Tue, 18 Feb 2020 10:38:24 +0000 (11:38 +0100)]
ASoC: soc-pcm: fix regression in soc_new_pcm()
Commit af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai")
swapped the SNDRV_PCM_STREAM_* parameter in the
snd_soc_dai_stream_valid(cpu_dai, ...) checks. But that works only
for codec2codec links. For normal links it breaks registration of
playback/capture-only PCM devices.
E.g. on qcom/apq8016_sbc there is usually one playback-only and one
capture-only PCM device, but they disappeared after the commit.
The codec2codec case was added in commit a342031cdd08
("ASoC: create pcm for codec2codec links as well") as an extra check
(e.g. `playback = playback && cpu_playback->channels_min`).
We should be able to simplify the code by checking directly for
the correct stream type in the loop.
This also fixes the regression because we check for PLAYBACK for
both codec and cpu dai again when codec2codec is not used.
Fixes: af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai") Signed-off-by: Stephan Gerhold <stephan@gerhold.net> Tested-by: Jerome Brunet <jbrunet@baylibre.com> Reviewed-by: Jerome Brunet <jbrunet@baylibre.com> Cc: Jerome Brunet <jbrunet@baylibre.com> Cc: Sameer Pujar <spujar@nvidia.com> Link: https://lore.kernel.org/r/20200218103824.26708-1-stephan@gerhold.net Signed-off-by: Mark Brown <broonie@kernel.org>
Jerome Brunet [Mon, 17 Feb 2020 09:20:19 +0000 (10:20 +0100)]
ASoC: meson: aiu: simplify component addition
Now that the component name is unique within ASoC, there is no need to
hack the debugfs prefix to add more than one ASoC component to a linux
device. Remove the unnecessary function and use
snd_soc_register_component() directly.
Tzung-Bi Shih [Mon, 17 Feb 2020 03:16:53 +0000 (11:16 +0800)]
drm/mediatek: fix race condition for HDMI jack status reporting
hdmi_conn_detect and mtk_hdmi_audio_hook_plugged_cb would be called
by different threads.
Imaging the following calling sequence:
Thread A Thread B
--------------------------------------------------------------------
mtk_hdmi_audio_hook_plugged_cb()
mtk_cec_hpd_high() -> disconnected
hdmi_conn_detect()
mtk_cec_hpd_high() -> connected
plugged_cb(connected)
plugged_cb(disconnected)
The latest disconnected is false reported. Makes mtk_cec_hpd_high
and plugged_cb atomic to fix.
Also uses the same lock to protect read/write of plugged_cb and codec_dev.
Jerome Brunet [Fri, 14 Feb 2020 13:13:48 +0000 (14:13 +0100)]
ASoC: meson: aiu: fix irq registration
The aiu stored the irq in an unsigned integer which may have discarded an
error returned by platform_get_irq_byname(). This is incorrect and should
have been a signed integer.
Also drop the irq error traces from the probe function as this is already
done by platform_get_irq_byname().
Fixes: 6ae9ca9ce986 ("ASoC: meson: aiu: add i2s and spdif support") Reported-by: kbuild test robot <lkp@intel.com> Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20200214131350.337968-4-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie@kernel.org>
Jerome Brunet [Fri, 14 Feb 2020 13:47:04 +0000 (14:47 +0100)]
ASoC: core: ensure component names are unique
Make sure each ASoC component is registered with a unique name.
The component is derived from the device name. If a device registers more
than one component, the component names will be the same.
This usually brings up a warning about the debugfs directory creation of
the component since directory already exists.
In such case, start numbering the component of the device so the names
don't collide anymore.
Jerome Brunet [Thu, 13 Feb 2020 15:51:57 +0000 (16:51 +0100)]
ASoC: meson: axg: extract sound card utils
This prepares the addition of the GX SoC family sound card driver.
The GX sound card, while slightly different, will be similar to the
AXG one. The purpose of this change is to share the utils common to
both sound card driver.
Jerome Brunet [Thu, 13 Feb 2020 15:51:55 +0000 (16:51 +0100)]
ASoC: meson: aiu: add hdmi codec control support
Add the codec to codec component which handles the routing between
the audio producers (PCM and I2S) and the synopsys hdmi controller
on the amlogic GX SoC family
Jerome Brunet [Thu, 13 Feb 2020 15:51:53 +0000 (16:51 +0100)]
ASoC: meson: aiu: add audio output dt-bindings
Add the dt-bindings and documentation of the AIU audio controller.
This component provides most of the audio outputs found on the Amlogic
Gx SoC family.
Jerome Brunet [Thu, 13 Feb 2020 15:51:52 +0000 (16:51 +0100)]
ASoC: meson: g12a: extract codec-to-codec utils
The hdmi routing mechanism used on g12a hdmi is also used:
* other Amlogic SoC types
* for the internal DAC path
Each of these codec glues are slightly different but the idea
behind it remains the same. This change extract some helper functions
from the g12a-tohdmitx driver to make them available for other Amlogic
codecs.
Jerome Brunet [Thu, 13 Feb 2020 15:51:51 +0000 (16:51 +0100)]
ASoC: core: allow a dt node to provide several components
At the moment, querying the dai_name will stop of the first component
matching the dt node. This does not allow a device (single dt node) to
provide several ASoC components which could then be used through DT.
This change let the search go on if the xlate function of the component
returns an error, giving the possibility to another component to match
and return the dai_name.
Reported-by: Randy Dunlap <rdunlap@infradead.org> Fixes: ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS") Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org> Link: https://lore.kernel.org/r/20200212145650.4602-4-geert@linux-m68k.org Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/...: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int]
drivers/base/regmap/regmap-i2c.c: In function ‘regmap_smbus_byte_reg_read’:
drivers/base/regmap/regmap-i2c.c:25:8: error: implicit declaration of function ‘i2c_smbus_read_byte_data’; did you mean ‘i2c_set_adapdata’? [-Werror=implicit-function-declaration]
Reported-by: Randy Dunlap <rdunlap@infradead.org> Fixes: ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS") Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org> Acked-by: Randy Dunlap <rdunlap@infradead.org> # build-tested Link: https://lore.kernel.org/r/20200212145650.4602-2-geert@linux-m68k.org Signed-off-by: Mark Brown <broonie@kernel.org>
soc_pcm_open() operation order is not good.
At first, soc_pcm_open() operation order is
1) CPU DAI startup
2) Component open
3) Codec DAI startup
4) rtd startup
But here, 2) will call try_module_get() if component has
module_get_upon_open flags. This means 1) CPU DAI startup
will be operated *before* its module was loaded.
DAI should be called *after* Component.
Second, soc_pcm_close() operation order is
1) CPU DAI shutdown
2) Codec DAI shutdown
3) rtd shutdown
4) Component close
soc_pcm_open() and soc_pcm_close() are paired function,
but, its operation order is unbalance.
This patch tidyup soc_pcm_open() order to Component -> rtd -> DAI.
This is one of prepare for cleanup soc-pcm-open()
ASoC: soc-pcm: call snd_soc_component_open/close() once
Current soc_pcm_open() calls snd_soc_component_open() under loop.
Thus, it needs to care about opened/not-yet-opened Component.
But, if soc-component.c is handling it, soc-pcm.c don't need to care
about it.
This patch adds opened flag to soc-component.h, and simplify soc-pcm.c.
This is one of prepare for cleanup soc-pcm-open()
Tzung-Bi Shih [Wed, 12 Feb 2020 05:55:17 +0000 (13:55 +0800)]
ASoC: mediatek: mt8183-da7219: add speaker switch
Da7219 and max98357a share the same I2S lines. When writing audio data
to the I2S, both codecs generate sound.
Da7219 already has a separate control "Headphone Switch". Adds a new
control "Speakers Switch" for turning on/off max98357a. Userspace
program can decide to turn on/off which codecs by different use cases.
Tzung-Bi Shih [Wed, 12 Feb 2020 05:55:16 +0000 (13:55 +0800)]
ASoC: max98357a: move control of SD_MODE to DAPM
Some machine may share the same I2S lines for multiple codecs. For
example, mediatek/mt8183/mt8183-da7219-max98357 shares the same lines
between max98357a and da7219. When writing audio data through the I2S
lines, all codecs on the lines would try to generate sound if they
accepts DO line. As a result, multiple codecs generate sound at a
time.
Moves control of SD_MODE to DAPM so that machine drivers have chances
to manipulate DAPM widget to turn on/off max98357a.
ASoC: wm0010: Replace zero-length array with flexible-array member
The current codebase makes use of the zero-length array language
extension to the C90 standard, but the preferred mechanism to declare
variable-length types such as these ones is a flexible array member[1][2],
introduced in C99:
struct foo {
int stuff;
struct boo array[];
};
By making use of the mechanism above, we will get a compiler warning
in case the flexible array does not occur last in the structure, which
will help us prevent some kind of undefined behavior bugs from being
inadvertenly introduced[3] to the codebase from now on.
Peter Ujfalusi [Mon, 10 Feb 2020 15:33:36 +0000 (17:33 +0200)]
ALSA: dmaengine_pcm: Consider DMA cache caused delay in pointer callback
Some DMA engines can have big FIFOs which adds to the latency.
The DMAengine framework can report the FIFO utilization in bytes. Use this
information for the delay reporting.
ASoC: soc-pcm: call snd_soc_dai_startup()/shutdown() once
Current soc_pcm_open() calls snd_soc_dai_startup() under loop.
Thus, it needs to care about started/not-yet-started codec DAI.
But, if soc-dai.c is handling it, soc-pcm.c don't need to care
about it.
This patch adds started flag to soc-dai.h, and simplify soc-pcm.c.
This is one of prepare for cleanup soc-pcm-open()
ret |= snd_soc_component_close(component, substream);
ret |= snd_soc_component_hw_free(component, substream);
The driver may return arbitrary error codes so they can conflict.
The bit-OR'ed error works only if the return code is always consistent.
This patch fixup it, and use *last* ret value.
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