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The driver currently only supports configuring for sample rates <= 96k
and it isn't possible to setup a configuration that will support all
sample rates up to 192k.
For sample rates up to 96k MCLK is in the 12MHz group.
However, although 192k only requires an I2S clock in the 12MHz group,
the cs42l42 audio path is not natively 192k so the audio must be
resampled. But for 192k the SRC requires a 24MHz MCLK.
It is not possible to switch MCLK between 12MHz and 24MHz groups
on-the-fly. The 12MHz group supports all sample rates up to 96k.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211015133619.4698-11-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The driver can run without an interrupt so if devm_request_threaded_irq()
failed, the probe() just carried on. But if this was EPROBE_DEFER the
driver would continue without an interrupt instead of deferring to wait
for the interrupt to become available.
Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec")
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211015133619.4698-6-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Volatile registers don't need a default value.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec")
Link: https://lore.kernel.org/r/20211015133619.4698-5-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Some registers had wrong default values in cs42l42_reg_defaults[].
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec")
Link: https://lore.kernel.org/r/20211015133619.4698-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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An I2S frame always has two slots (left and right) even when sending
mono. The right channel (channel 2) of ASP TX will always have the
same bit width as the left channel and will always be on the high
phase of LRCLK.
The previous implementation always passed the field masks for both
channels to snd_soc_component_update_bits() but for mono the written value
only contained the settings for channel 1. The result was that for mono
channel 2 was set to 8-bit (which is an invalid configuration) with both
channels on the low phase of LRCLK.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Fixes: 585e7079de0e ("ASoC: cs42l42: Add Capture Support")
Link: https://lore.kernel.org/r/20211015133619.4698-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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When capture and playback substreams are both running at the same time,
cs42l42_pcm_hw_params() would be called for each direction. The first
call will configure the PLL. The second call must not write the PLL
configuration registers again if the first substream is already running,
as this could destabilize the PLL.
The DAI is marked symmetric sample bits and sample rate, so the two
directions will always have the same SCLK (I2S always has 2 channel slots
so the DAI does not need to require symmetric channels to guarantee the
same SCLK). However, since cs42l42_pll_config() is checking for an active
stream it may as well test that the requested SCLK is the same as the
currently active configuration.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211015133619.4698-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Currently errors on register read/write/update are reported with
an error code and the corresponding function but does not provide
any details on the which register number did it actually fail.
register number can give better clue and it should be easy to
locate the code and fix.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211014161330.26645-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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Enable vangogh platform machine driver build.
Signed-off-by: VIjendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20211014071714.836410-3-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add Vangogh machine driver using NAU8821 & CS35L41 Codecs.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20211014071714.836410-2-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Create platform device for Vangogh machine driver.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20211014071714.836410-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds Codec2Codec-Multi sample to audio-graph-card2-custom-sample.dtsi.
Because it can use very basic connection only for now,
it can use only
- 2channels
- S32_LE format
Test-Component driver has "IN" and "OUT" widget. Thus the route is
+--+ +-+
| | | |- Codec8 <- IN
| | <- | |- Codec9 <- IN
| | +-+
| |
| | +-+
| | -> | |- Codec10 -> OUT
| | | |- Codec11 -> OUT
+--+ +-+
One note here is that it will start works when it boot.
In other words we can't stop it so far.
We need to update driver for it in the future.
...
asoc-audio-graph-card2-custom-sample: multicodec <-> multicpu mapping ok
test-component test_codec: test_dai_startup() : test_codec.9
test-component test_codec: test_dai_startup() : test_codec.8
test-component test_codec: test_dai_startup() : test_codec.11
test-component test_codec: test_dai_startup() : test_codec.10
...
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87mtnelu2k.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds Codec2Codec-Single sample to audio-graph-card2-custom-sample.dtsi.
Because it can use very basic connection only for now,
it can use only
- 2channels
- S32_LE format
Test-Component driver has "IN" and "OUT" widget. Thus the route is
+--+
| | <-- Codec6 <-- IN
| | --> Codec7 --> OUT
+--+
One note here is that it will start works when it boot.
In other words we can't stop it so far.
We need to update driver for it in the future.
...
asoc-audio-graph-card2-custom-sample: test_codec.7 <-> test_codec.6 mapping ok
test-component test_codec: test_dai_startup() : test_codec.6
test-component test_codec: test_dai_startup() : test_codec.7
...
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o87ulu2o.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds DPCM link Multi-CPU/Codec sample to
audio-graph-card2-custom-sample.dtsi.
This sample is assuming MIXer connection.
One note is that Multi-FE is not supported on ASoC
FE BE
**** +-+
CPU5 -- * * -- | | -- Codec4
CPU6 -- * * | | -- Codec5
**** +-+
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pmsalu2s.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds DPCM link Single-CPU/Codec sample to
audio-graph-card2-custom-sample.dtsi.
This sample is assuming MIXer connection.
FE BE
****
CPU3 -- * * -- Codec3
CPU4 -- * *
****
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r1cqlu2w.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds Normal link Multi-CPU/Codec sample to
audio-graph-card2-custom-sample.dtsi.
+-+ +-+
CPU1 --| | <---> | | -- Codec1
CPU2 --| | | | -- Codec2
+-+ +-+
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87sfx6lu30.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Audio Graph Card2 settings is a little bit difficult for beginner,
and Customizing it also difficult/confusable too.
So, this patch adds sample for it.
You can easily use it by adding below line on your DT file,
and select CONFIGs to your .config.
#include "../../../../../sound/soc/generic/audio-graph-card2-custom-sample.dtsi"
CONFIG_SND_AUDIO_GRAPH_CARD2
CONFIG_SND_AUDIO_GRAPH_CARD2_CUSTOM_SAMPLE
CONFIG_SND_TEST_COMPONENT
This patch uses audio-graph-card2 base custom sample driver.
You can directly use audio-graph-card2 instead of custom sample driver
by modifing compatible.
- compatible = "audio-graph-card2-custom-sample";
+ compatible = "audio-graph-card2";
Sample custom driver will indicate customized print.
It is using Test-Component driver for CPU/Codec.
It can indicate more detail print of each behavior if user want to.
In such case, you need to update compatible to "xxx-nv" or "xxx-vv".
- compatible = "test-cpu";
+ compatible = "test-cpu-verbose";
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87tuhmlu35.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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audio-graph-card2 has customizing support.
This means user can re-use audio-graph-card2 DT parsing, and possible
to expand to own special handling.
This patch adds Audio Graph Card2 Customize Sample Driver.
It can re-use audio-graph-card2 parsing by calling
audio_graph2_parse_of(...), and user can expand each functions by
using hooks.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v922lu3c.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds Codec2Codec support to audio-graph-card2.
It can use Codec2Codec but very simple case only for now.
It doesn't have "SWITCH" control yet, thus it start automatically
when it was probed, and can't stop, so far.
Thus it needs to be updated around widgets/routing handling,
and you need to understand that it is under experimental.
Codec has SND_SOC_DAPM_INPUT() (= IN) / SND_SOC_DAPM_OUTPUT(= OUT)
widgets in below case.
It is assuming 2channel, S32_LE format for now.
It needs to be updated, too.
It needs "codec2codec" node (= B), needs to have routing (= A),
need to indicate CPU side at links (= X).
ports@0 is for CPU side (= X), port@1 is Codec side (= Y).
It needs to have "rate" (= C)
+--+
| |<-- Codec0 <-- IN
| |--> Codec1 --> OUT
+--+
sound {
compatible = "audio-graph-card2";
(A) routing = "OUT" ,"DAI1 Playback",
"DAI0 Capture", "IN";
(X) links = <&c2c>;
(B) codec2codec {
ports {
(C) rate = <48000>;
(X) c2c: port@0 { c2cf_ep: endpoint { remote-endpoint = <&codec0_ep>; }; };
(Y) port@1 { c2cb_ep: endpoint { remote-endpoint = <&codec1_ep>; }; };
};
};
Codec {
ports {
port@0 {
bitclock-master;
frame-master;
codec0_ep: endpoint { remote-endpoint = <&c2cf_ep>; }; };
port@1 { codec1_ep: endpoint { remote-endpoint = <&c2cb_ep>; }; };
};
};
Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87y26ylu4a.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds DPCM support to audio-graph-card2.
It uses "dpcm" node (= D), needs to have routing (= A),
need to indicate both FE/BE at links (= B, C).
dpcm ports@0 is for FE (= B), port@1 is for BE (= C).
remote-endpoint can use both Single/Multi connection.
DSP
************
PCM0 <--> * fe0 be0 * <--> DAI0: Codec Headset
PCM1 <--> * fe1 be1 * <--> DAI1: Codec Speakers
PCM2 <--> * fe2 be2 * <--> DAI2: MODEM
PCM3 <--> * fe3 be3 * <--> DAI3: BT
* be4 * <--> DAI4: DMIC
* be5 * <--> DAI5: FM
************
sound {
compatible = "audio-graph-card2";
// indicate routing
(A) routing = "xxx Playback", "xxx Playback",
"xxx Playback", "xxx Playback",
"xxx Playback", "xxx Playback";
// indicate all Front-End, Back-End in DPCM case
(B) links = <&fe0, &fe1, ...
(C) &be0, &be1, ...
(D) dpcm {
// Front-End
ports@0 {
(B) fe0: port@0 { fe0_ep: endpoint { remote-endpoint = <&pcm0_ep>; }; };
(B) fe1: port@1 { fe1_ep: endpoint { remote-endpoint = <&pcm1_ep>; }; };
...
};
// Back-End
ports@1 {
(C) be0: port@0 { be0_ep: endpoint { remote-endpoint = <&dai0_ep>; }; };
(C) be1: port@1 { be1_ep: endpoint { remote-endpoint = <&dai1_ep>; }; };
...
};
};
};
CPU {
ports {
bitclock-master;
frame-master;
port@0 { pcm0_ep: endpoint { remote-endpoint = <&fe0_ep>; }; };
port@1 { pcm1_ep: endpoint { remote-endpoint = <&fe1_ep>; }; };
...
};
};
Codec {
ports {
port@0 { dai0_ep: endpoint { remote-endpoint = <&be0_ep>; }; };
port@1 { dai1_ep: endpoint { remote-endpoint = <&be1_ep>; }; };
...
};
};
Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87zgrelu4v.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch adds Multi CPU/Codec support to audio-graph-card2.
Multi CPU/Codec will have connection part (= X) and CPU/Codec list part (= Y).
links indicates connection part of CPU side (= A).
+-+ (A) +-+
CPU1 --(Y) | | <-(X)--(X)-> | | (Y)-- Codec1
CPU2 --(Y) | | | | (Y)-- Codec2
+-+ +-+
sound {
compatible = "audio-graph-card2";
(A) links = <&mcpu>;
multi {
ports@0 {
(X) (A) mcpu: port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; };
(Y) port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; };
(Y) port@1 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; };
};
ports@1 {
(X) port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; };
(Y) port@0 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; };
(Y) port@1 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; };
};
};
};
CPU {
ports {
bitclock-master;
frame-master;
port@0 { cpu1_ep: endpoint { remote-endpoint = <&mcpu1_ep>; }; };
port@1 { cpu2_ep: endpoint { remote-endpoint = <&mcpu2_ep>; }; };
};
};
Codec {
ports {
port@0 { codec1_ep: endpoint { remote-endpoint = <&mcodec1_ep>; }; };
port@1 { codec2_ep: endpoint { remote-endpoint = <&mcodec2_ep>; }; };
};
};
Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/20210804171748.GC26252@sirena.org.uk
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/871r4qn8pk.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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We already have audio-graph-card which is Of-graph base of general
sound card driver.
It is supporting DPCM connection, but was forcibly expanded.
Thus, it is very difficult to add new features on it, for example
Multi CPU/Codec support, Codec2Codec support, etc.
This patch adds more flexible new Audio Graph Card2 driver for it.
audio-graph-card and audio-graph-card2 are similar, but don't have
full compatibility.
Audio Graph Card2 supports very generic connection, but some users
want to have its own settings, for example PLL settings, etc.
For such case, it has customizing support.
In users own driver, it can use Audio Graph Card2 parsing by using
audio_graph2_parse_of(), and doing its own customizing.
Because Audio Graph Card2 is still under experimental stage,
it will indicate such warning when probing, and the DT syntax
might be changed.
Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/871r8u4s6q.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/87a6mhwyqn.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/87tuitusy4.wl-kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/87a6jn56x0.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8735p6n8q1.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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codec2codec needs snd_soc_pcm_stream settings.
This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874k9mn8qy.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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audio-graph-card2 will support DPCM/Multi/Codec2Codec,
and these will use almost same DT settings which uses
ports0 and ports1.
This patch adds asoc_graph_is_ports0() which checks
port is under port0 or not.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/875yu2n8ra.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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We already have dummy-codec, dummy-platform.
But its issues are
1) we don't have dummy-cpu,
2) we can't select it via DeviceTree
3) It do nothing
Sometimes we want to have Dummy Sound Component for debugging,
for testing, for learning Framework behavior, etc, etc...
This patch adds Test-Component driver for it.
User can select CPU Component by using "test-cpu" compatible,
and can select Codec Component by using "test-codec" compatible.
It doesn't support Platform so far, but is easy to add.
We can verbose print to know its progress if user selected
xxx-verbose compatible driver.
for example,
test-cpu : silent Component, silent DAI
test-cpu-verbose-component : verbose Component, silent DAI
test-cpu-verbose-dai : silent Component, verbose DAI
test-cpu-verbose : verbose Component, verbose DAI
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/877dein8rx.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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In laptop 'HP Spectre x360 Convertible 15-eb1xxx/8811' both front and
rear speakers are silent, this patch fixes that by overriding the pin
layout and by initializing the amplifier which needs a GPIO pin to be
set to 1 then 0, similar to the existing HP Spectre x360 14 model.
In order to have volume control, both front and rear speakers were
forced to use the DAC1.
This patch also correctly map the mute LED but since there is no
microphone on/off switch exposed by the alsa subsystem it never turns
on by itself.
There are still known audio issues in this laptop: headset microphone
doesn't work, the button to mute/unmute microphone is not yet mapped,
the LED of the mute/unmute speakers doesn't seems to be exposed via
GPIO and never turns on.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213953
Signed-off-by: Davide Baldo <davide@baldo.me>
Link: https://lore.kernel.org/r/20211015072121.5287-1-davide@baldo.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As per discussion at: https://github.com/szszoke/sennheiser-gsp670-pulseaudio-profile/issues/13
The GSP670 has 2 playback and 1 recording device that by default are
detected in an incompatible order for alsa. This may have been done to make
it compatible for the console by the manufacturer and only affects the
latest firmware which uses its own ID.
This quirk will resolve this by reordering the channels.
Signed-off-by: Brendan Grieve <brendan@grieve.com.au>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211015025335.196592-1-brendan@grieve.com.au
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Both snd_pcm_delay() and snd_pcm_hwsync() do the almost same thing.
The only difference is that the former calculate the delay, so unify
them as a code cleanup, and treat NULL delay argument only for hwsync
operation.
Also, the patch does a slight code refactoring in snd_pcm_delay().
The initialization of the delay value is done in the caller side now.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20211014145323.26506-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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So far we used to read the current value of the mixer element
dynamically at the first access, and the error from a GET_CUR message
is treated as a fatal error (unless QUIRK_IGNORE_CTL_ERROR is set).
It's rather inconvenient, as most of GET_CUR errors are no fatal, and
we can continue operation with assumption of some fixed value.
This patch makes the USB-audio driver to change the behavior at probe
time; now it tries to initialize the current value of each mixer
element that is built from a feature unit (those for typically for
mixer volumes and switches). When a read failure happens, it tries to
set the known minimum value. After that point, a cached value is used
always, hence we won't hit GET_CUR message error any longer.
The error from GET_CUR message is still shown as a warning normally,
but only once at the probe time, and it'll keep operating. If the
message is confirmed to be harmless, it can be shut up by
QUIRK_IGNORE_CTL_ERROR quirk flag, too.
Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20211014130636.17860-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The error from snd_usb_lock_shutdown() indicates that the device is
disconnected, hence it makes no sense to show any further control
message error in get_ctl_value_v2(). Return the error directly
instead.
Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20211014130636.17860-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The error message in get_ctl_value_v2() (for UAC2/3) is shown via
KERN_ERR level but it was intended to be rather a debug message as
seen in get_ctl_value_v1() (for UAC1). This patch downgrade the
printk level.
Tested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20211014130636.17860-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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A back-merge of 5.15 branch into 5.16-devel branch for further
development of USB and ALSA core stuff that depends on 5.15 fixes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Apply existing PCI quirk to the Clevo PC50HS and related models to fix
audio output on the built in speakers.
Signed-off-by: Steven Clarkson <sc@lambdal.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211014133554.1326741-1-sc@lambdal.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Shciit Hel device responds to the ctl message for the mic capture
switch with a timeout of -EPIPE:
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
usb 7-2.2: cannot get ctl value: req = 0x81, wValue = 0x100, wIndex = 0x1100, type = 1
This seems safe to ignore as the device works properly with the control
message quirk, so add it to the quirk table so all is good.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-usb@vger.kernel.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/YWgR3nOI1osvr5Yo@kroah.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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A little pop can be heard obviously from HP while playing a silent.
This patch fixes it by using two functions:
1. Enable HP 1bit output mode.
2. Change the charge pump switch size during playback on and off.
Signed-off-by: Derek Fang <derek.fang@realtek.com>
Link: https://lore.kernel.org/r/20211014094054.811-1-derek.fang@realtek.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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There are several things the patch adding the support for 'I2S Reference'
got wrong:
- "None" selection is in fact equals to last selected reference
- The custom put overrides RX/TX len, TDM slot sizes, etc
- the enum is useless in most part for the reference tracking
- there is no need for EXT control as there is a single bit in
RT1011_TDM1_SET_1 register (bit 7) which selects the reference
- it was using ucontrol->value.integer.value[0] in the put/get callbacks
which causesed access to 'I2S Reference' enum with alsamixer to fail
Complements: c3de683c4d1d ("ASoC: rt1011: Fix 'I2S Reference' enum control caused error")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20211013123300.11095-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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There is a noise issue for 8kHz sample rate on slave mode.
Compared with master mode, the difference is the DACDIV
setting, after correcting the DACDIV, the noise is gone.
There is no noise issue for 48kHz sample rate, because
the default value of DACDIV is correct for 48kHz.
So wm8960_configure_clocking() should be functional for
ADC and DAC function even if it is slave mode.
In order to be compatible for old use case, just add
condition for checking that sysclk is zero with
slave mode.
Fixes: 0e50b51aa22f ("ASoC: wm8960: Let wm8960 driver configure its bit clock and frame clock")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/1634102224-3922-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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ChiYuan Huang <cy_huang@richtek.com>:
From: ChiYuan Huang <cy_huang@richtek.com>
This patch series Add the Richtek RT9120 support.
In v4:
- Add 'classd_tlv' for 'SPK Gain Volume' control item.
- Unify the tlv declaration to the postfix '_tlv'.
- Fix 'digital_tlv' mute as 1 to declare the minimum is muted.
In v3:
- Add dvdd regulator binding to check the dvdd voltage domain.
- Refine sdo_select_text.
- Use switch case in 'internal_power_event' function.
- Remove the volume and mute initially write in component probe.
- Remove the mute API. It's no need by HW design.
In v2:
- Add missing #sound-dai-cells property.
ChiYuan Huang (2):
ASoC: dt-bindings: rt9120: Add initial bindings
ASoC: rt9120: Add rt9210 audio amplifier support
.../devicetree/bindings/sound/richtek,rt9120.yaml | 59 +++
sound/soc/codecs/Kconfig | 10 +
sound/soc/codecs/Makefile | 2 +
sound/soc/codecs/rt9120.c | 495 +++++++++++++++++++++
4 files changed, 566 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/richtek,rt9120.yaml
create mode 100644 sound/soc/codecs/rt9120.c
--
2.7.4
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Add Richtek rt9120 audio amplifier support.
Signed-off-by: ChiYuan Huang <cy_huang@richtek.com>
Link: https://lore.kernel.org/r/1634088519-995-3-git-send-email-u0084500@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The only usage of acp5x_i2s_dai_ops is to assign its address to the ops
field in the snd_soc_dai_driver struct, which is a pointer to const.
Make it const to allow the compiler to put it in read-only memory.
Signed-off-by: Rikard Falkeborn <rikard.falkeborn@gmail.com>
Link: https://lore.kernel.org/r/20211012211506.21159-1-rikard.falkeborn@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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These are only assigned to the ops field in the snd_soc_dai_link struct
which is a pointer to const struct snd_soc_ops. Make them const to allow
the compiler to put them in read-only memory.
Signed-off-by: Rikard Falkeborn <rikard.falkeborn@gmail.com>
Link: https://lore.kernel.org/r/20211012205521.14098-1-rikard.falkeborn@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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This drops the rt9210 support due to a race with a new version being
sent out for some incremental changes.
Signed-off-by: Mark Brown <broonie@kernel.org>
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The device advertises 8 formats, but only a rate of 48kHz is honored
by the hardware and 24 bits give chopped audio, so only report the
one working combination. This fixes out-of-the-box audio experience
with PipeWire which otherwise attempts to choose S24_3LE (while
PulseAudio defaulted to S16_LE).
Signed-off-by: Jonas Hahnfeld <hahnjo@hahnjo.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211012200906.3492-1-hahnjo@hahnjo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The snd_hdac_bus_reset_link() contains logic to clear STATESTS register
before performing controller reset. This code dates back to an old
bugfix in commit e8a7f136f5ed ("[ALSA] hda-intel - Improve HD-audio
codec probing robustness"). Originally the code was added to
azx_reset().
The code was moved around in commit a41d122449be ("ALSA: hda - Embed bus
into controller object") and ended up to snd_hdac_bus_reset_link() and
called primarily via snd_hdac_bus_init_chip().
The logic to clear STATESTS is correct when snd_hdac_bus_init_chip() is
called when controller is not in reset. In this case, STATESTS can be
cleared. This can be useful e.g. when forcing a controller reset to retry
codec probe. A normal non-power-on reset will not clear the bits.
However, this old logic is problematic when controller is already in
reset. The HDA specification states that controller must be taken out of
reset before writing to registers other than GCTL.CRST (1.0a spec,
3.3.7). The write to STATESTS in snd_hdac_bus_reset_link() will be lost
if the controller is already in reset per the HDA specification mentioned.
This has been harmless on older hardware. On newer generation of Intel
PCIe based HDA controllers, if configured to report issues, this write
will emit an unsupported request error. If ACPI Platform Error Interface
(APEI) is enabled in kernel, this will end up to kernel log.
Fix the code in snd_hdac_bus_reset_link() to only clear the STATESTS if
the function is called when controller is not in reset. Otherwise
clearing the bits is not possible and should be skipped.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20211012142935.3731820-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The recent support for the improved low-latency playback mode applied
the SNDRV_PCM_INFO_EXPLICIT_SYNC flag for the target streams, but this
was a slight overkill. The use of the flag above disables effectively
both PCM status and control mmaps, while basically what we want to
track is only about the appl_ptr update.
For less restriction, use a more proper flag,
SNDRV_PCM_INFO_SYNC_APPLPTR instead, which disables only the control
mmap.
Fixes: d5f871f89e21 ("ALSA: usb-audio: Improved lowlatency playback support")
Link: https://lore.kernel.org/r/20211011103650.10182-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We need to define the codec pin 0x1b to be the mic, but somehow
the mic doesn't support hot plugging detection, and Windows also has
this issue, so we set it to phantom headset-mic.
Also the determine_headset_type() often returns the omtp type by a
mistake when we plug a ctia headset, this makes the mic can't record
sound at all. Because most of the headset are ctia type nowadays and
some machines have the fixed ctia type audio jack, it is possible this
machine has the fixed ctia jack too. Here we set this mic jack to
fixed ctia type, this could avoid the mic type detection mistake and
make the ctia headset work stable.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214537
Reported-and-tested-by: msd <msd.mmq@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20211012114748.5238-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Access to 'I2S Reference' enum causes alsamixer to fail to load:
$ alsamixer
cannot load mixer controls: Invalid argument
cml_rt1011_rt5682 cml_rt1011_rt5682: control 2:0:0:TL I2S Reference:0: access overflow
The reason is that the original patch adding the code was using
ucontrol->value.integer.value[0]
instead the correct
ucontrol->value.enumerated.item[0]
for an ENUM control.
Fixes: 87f40af26c262 ("ASoC: rt1011: add i2s reference control for rt1011")
Reported-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20211011144518.2518-1-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Ensure the default 0dB playback path is always used.
The code that set FULL_SCALE_VOL based on LOAD_DET_RCSTAT was
spurious, and resulted in a -6dB attenuation being accidentally
inserted into the playback path.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20211011144903.28915-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The small set of cleanups against bytcr_rt5651 board file.
In v2:
- added commit message to patch 2 (Joe, Pierre)
- added cover letter (Pierre)
- added Hans to Cc list (Hans)
Andy Shevchenko (4):
ASoC: Intel: bytcr_rt5651: Get platform data via dev_get_platdata()
ASoC: Intel: bytcr_rt5651: Use temporary variable for struct device
ASoC: Intel: bytcr_rt5651: use devm_clk_get_optional() for mclk
ASoC: Intel: bytcr_rt5651: Utilize dev_err_probe() to avoid log
saturation
sound/soc/intel/boards/bytcr_rt5651.c | 118 +++++++++++---------------
1 file changed, 50 insertions(+), 68 deletions(-)
--
2.33.0
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Michael Forney reported an incorrect padding type that was defined in
the commit 80fe7430c708 ("ALSA: add new 32-bit layout for
snd_pcm_mmap_status/control") for PCM control mmap data.
His analysis is correct, and this caused the misplacements of PCM
control data on 32bit arch and 32bit compat mode.
The bug is that the __pad2 definition in __snd_pcm_mmap_control64
struct was wrongly with __pad_before_uframe, which should have been
__pad_after_uframe instead. This struct is used in SYNC_PTR ioctl and
control mmap. Basically this bug leads to two problems:
- The offset of avail_min field becomes wrong, it's placed right after
appl_ptr without padding on little-endian
- When appl_ptr and avail_min are read as 64bit values in kernel side,
the values become either zero or corrupted (mixed up)
One good news is that, because both user-space and kernel
misunderstand the wrong offset, at least, 32bit application running on
32bit kernel works as is. Also, 64bit applications are unaffected
because the padding size is zero. The remaining problem is the 32bit
compat mode; as mentioned in the above, avail_min is placed right
after appl_ptr on little-endian archs, 64bit kernel reads bogus values
for appl_ptr updates, which may lead to streaming bugs like jumping,
XRUN or whatever unexpected.
(However, we haven't heard any serious bug reports due to this over
years, so practically seen, it's fairly safe to assume that the impact
by this bug is limited.)
Ideally speaking, we should correct the wrong mmap status control
definition. But this would cause again incompatibility with the
existing binaries, and fixing it (e.g. by renumbering ioctls) would be
really messy.
So, as of this patch, we only correct the behavior of 32bit compat
mode and keep the rest as is. Namely, the SYNC_PTR ioctl is now
handled differently in compat mode to read/write the 32bit values at
the right offsets. The control mmap of 32bit apps on 64bit kernels
has been already disabled (which is likely rather an overlook, but
this worked fine at this time :), so covering SYNC_PTR ioctl should
suffice as a fallback.
Fixes: 80fe7430c708 ("ALSA: add new 32-bit layout for snd_pcm_mmap_status/control")
Reported-by: Michael Forney <mforney@mforney.org>
Reviewed-by: Arnd Bergmann <arnd@arndb.de>
Cc: <stable@vger.kernel.org>
Cc: Rich Felker <dalias@libc.org>
Link: https://lore.kernel.org/r/29QBMJU8DE71E.2YZSH8IHT5HMH@mforney.org
Link: https://lore.kernel.org/r/20211010075546.23220-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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dev_err_probe() avoids printing into log when the deferred probe is invoked.
This is possible when clock provider is pending to appear.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20211007170250.27997-5-andriy.shevchenko@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Richard Fitzgerald
Srinivas Kandagatla
Vijendar Mukunda
Kuninori Morimoto
Davide Baldo
Brendan Grieve
Takashi Iwai
Steven Clarkson
Greg Kroah-Hartman
Derek Fang
Peter Ujfalusi
Shengjiu Wang
Mark Brown
ChiYuan Huang
Rikard Falkeborn
Jonas Hahnfeld
Kai Vehmanen
Hui Wang
Stefan Binding
Andy Shevchenko