June 2020 - Alsa-devel - mailman.alsa-project.org

Re: [PATCH] ALSA: usb-audio: Add registration quirk for Kingston HyperX Cloud Flight S
by Takashi Iwai 23 Jun '20

23 Jun '20

On Fri, 19 Jun 2020 13:48:22 +0200, Christoffer Nielsen wrote: > > Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud > Alpha S (0951:0x16ea) uses two interfaces, but only the second > interface contains the capture stream. This patch delays the > registration until the second interface appears. > > > Signed-off-by: Christoffer Nielsen <segadora93(a)gmail.com> Applied now (with some correction and Cc to stable). thanks, Takashi

1 0

Re: [PATCH] ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable
by Markus Elfring 23 Jun '20

23 Jun '20

> Fix unchecked return value for clk_prepare_enable. > > And because clk_prepare_enable and clk_disable_unprepare should > check input clock parameter is NULL or not, then we don't need > to check it before calling the function. I propose to split the adjustment of two function implementations into separate update steps for a small patch series. https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/Doc… I suggest to improve the change descriptions accordingly. Regards, Markus

2 1

[PATCH v2 0/3] ALSA: compress: Document stream states and fix gapless SM
by Vinod Koul 23 Jun '20

23 Jun '20

Srini found issue with gapless implementation which prompted to look deeper into SM for compressed stream. So documenting SM was first step, so first two patches add that. Last patch fixes the issue by keeping track on partial_drain and then moving state to 'running' in snd_compr_drain_notify() for partial_drain case on success. While at it, noticed snd_compr_drain_notify() is lockless state change, so fixed that as well. I have tested this on Dragon board RB3, compressed audio works out of the box on that platform and Srini will send driver and fcplay patches for gapless soon. Changes in v2: - Added tested tag by Srini - Update compress SM with Free state and compr_stop() transitions Vinod Koul (3): ALSA: compress: document the compress audio state machine ALSA: compress: document the compress gapless audio state machine ALSA: compress: fix partial_drain completion state .../sound/designs/compress-offload.rst | 84 +++++++++++++++++++ include/sound/compress_driver.h | 12 ++- sound/core/compress_offload.c | 4 + 3 files changed, 99 insertions(+), 1 deletion(-) -- 2.26.2

3 6

[PATCH AUTOSEL 5.7 004/388] ASoC: tegra: tegra_wm8903: Support nvidia, headset property
by Sasha Levin 22 Jun '20

22 Jun '20

From: Dmitry Osipenko <digetx(a)gmail.com> [ Upstream commit 3ef9d5073b552d56bd6daf2af1e89b7e8d4df183 ] The microphone-jack state needs to be masked in a case of a 4-pin jack when microphone and ground pins are shorted. Presence of nvidia,headset tells that WM8903 CODEC driver should mask microphone's status if short circuit is detected, i.e headphones are inserted. Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> Link: https://lore.kernel.org/r/20200330204011.18465-3-digetx@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/tegra/tegra_wm8903.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 9b5651502f12..3aca354f9e08 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -177,6 +177,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_component *component = codec_dai->component; struct snd_soc_card *card = rtd->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); + int shrt = 0; if (gpio_is_valid(machine->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = machine->gpio_hp_det; @@ -189,12 +190,15 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) &tegra_wm8903_hp_jack_gpio); } + if (of_property_read_bool(card->dev->of_node, "nvidia,headset")) + shrt = SND_JACK_MICROPHONE; + snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, &tegra_wm8903_mic_jack, tegra_wm8903_mic_jack_pins, ARRAY_SIZE(tegra_wm8903_mic_jack_pins)); wm8903_mic_detect(component, &tegra_wm8903_mic_jack, SND_JACK_MICROPHONE, - 0); + shrt); snd_soc_dapm_force_enable_pin(&card->dapm, "MICBIAS"); -- 2.25.1

2 9

Re: [PATCH 1/2] dt-bindings: sound: Device tree bindings for the apq8039 sound complex
by Stephan Gerhold 22 Jun '20

22 Jun '20

Hi Pantelis, On Fri, Jun 19, 2020 at 10:38:30PM +0300, Pantelis Antoniou wrote: > Add a yaml device binding for the QCOM apq8039 sound complex driver. > Nice to see some activity to get sound working on another SoC! Thanks for documenting all these properties. > Signed-off-by: Pantelis Antoniou <pantelis.antoniou(a)linaro.org> > --- > .../bindings/sound/qcom,apq8039.yaml | 370 ++++++++++++++++++ > 1 file changed, 370 insertions(+) > create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8039.yaml > > diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8039.yaml b/Documentation/devicetree/bindings/sound/qcom,apq8039.yaml > new file mode 100644 > index 000000000000..f1c4fb99ccbb > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/qcom,apq8039.yaml > @@ -0,0 +1,370 @@ > +# SPDX-License-Identifier: GPL-2.0 > +%YAML 1.2 > +--- > +$id: http://devicetree.org/schemas/sound/qcom,apq8039.yaml# > +$schema: http://devicetree.org/meta-schemas/core.yaml# > + > +title: Qualcomm Technologies APQ8039 ASoC sound card > + > +maintainers: > + - Pantelis Antoniou <pantelis.antoniou(a)linaro.org> > + > +properties: > + compatible: > + items: > + - const: qcom,apq8039-sndcard > + > + pinctrl-0: > + $ref: /schemas/types.yaml#/definitions/phandle-array > + description: | > + Should specify pin control groups used for this controller matching > + the first entry in pinctrl-names. > + > + pinctrl-1: > + description: | > + Should specify pin control groups used for this controller matching > + the second entry in pinctrl-names. > + > + pinctrl-names: > + minItems: 1 > + items: > + - const: default > + - const: sleep > + description: > + Names for the pin configuration(s); may be "default" or "sleep", > + where the "sleep" configuration may describe the state > + the pins should be in during system suspend. > + > + reg: > + description: Must contain an address for each entry in "reg-names". > + minItems: 2 > + maxItems: 2 > + > + reg-names: > + items: > + - const: mic-iomux > + - const: spkr-iomux > + > + qcom,model: > + $ref: /schemas/types.yaml#/definitions/string > + description: The user-visible name of the sound complex. > + > + qcom,audio-routing: > + $ref: /schemas/types.yaml#/definitions/non-unique-string-array > + description: | > + List of the connections between audio components; each entry is a > + pair of strings, the first being the connection's sink, the second > + being the connection's source; valid names could be power supplies > + and MicBias of msm8916-analoc-wcd codec. > + > + function-definition: > + type: object > + description: | > + Functional configuration for the sound complex via a > + simple control. allows fixed and dynamically constructed > + function selection. > + > + properties: > + mixer-control: > + $ref: /schemas/types.yaml#/definitions/string > + description: | > + Name of the exported alsa mix control. > + > + function-list: > + $ref: /schemas/types.yaml#/definitions/phandle-array > + description: | > + phandle(s) of the functions which the sound complex > + exposes via the control. > + > + system-list: > + $ref: /schemas/types.yaml#/definitions/phandle-array > + description: | > + phandle(s) of the default, init and shutdown functions > + Must be one of the declared ones in the function property. > + The default function is the one selected by default on > + startup (after the init function's sequence is executed). > + On shutdown the shutdown function sequence will be executed. > + Typically init and shutdown are the same and it's purpose > + is to initialize the sound complex mixer controls to the > + all off state, and be ready for a regular function selection. > + > + patternProperties: > + "^[A-Za-z_][A-Aa-z0-9_]*$": > + type: object > + description: > + Function description subnodes. The name of the function > + is simply the name of the subnode, so restrictions apply > + to the valid node names. > + > + The function definition of each subnode is either a cooked > + function (i.e. which is not dependent on state inputs), or > + a function that is selecting a cooked function based on the > + state inputs and the generated state vector. > + > + oneOf: > + # non-cooked function > + - properties: > + enable: > + $ref: /schemas/types.yaml#/definitions/non-unique-string-array > + description: | > + Sequence of alsa mixer controls to apply when this state is to > + be enabled. > + > + disable: > + $ref: /schemas/types.yaml#/definitions/non-unique-string-array > + description: | > + Sequence of alsa mixer controls to apply when this state is to > + be disabled. > + > + required: > + - enable > + > + # cooked function > + - properties: > + state-inputs: > + description: | > + A list of state inputs to be used in constructing a state > + vector. > + type: array > + uniqueItems: true > + minItems: 1 > + items: > + anyOf: > + - const: JACK_HEADPHONE > + - const: JACK_MICROPHONE > + > + state-input-bits: > + $ref: /schemas/types.yaml#/definitions/uint32-array > + description: | > + Number of bits to use for each state-input in the > + state vector creation. For now only the value 1 is > + supported for JACK_HEADPHONE and JACK_MICROPHONE. > + > + state-input-defaults: > + $ref: /schemas/types.yaml#/definitions/uint32-array > + description: | > + The default value to use as a state input at startup. > + > + state-map: > + $ref: /schemas/types.yaml#/definitions/phandle-array > + description: | > + The mapping of this function to a cooked function. The > + format used is a sequence of phandle to a state, the mask > + to apply to the state vector, and the equality value. > + > + Take the example's configuration > + > + state-inputs = "JACK_HEADPHONE", "JACK_MICROPHONE"; > + state-input-bits = <1>, <1>; > + state-input-defaults = <0>, <0>; > + > + state-map = <&speaker 0x1 0x0>, > + <&headphones 0x3 0x1>, > + <&headset 0x3 0x3>; > + > + is decoded as follows. > + > + There are 3 possible cooked functions to be selected. > + speaker, headphone and headset. The state-inputs are > + the JACK_HEADPHONE and JACK_MICROPHONE, which are single > + bit values, being placed at bit 0 and bit 1 of the > + constructed vector. > + > + The 4 possible state vectors are: > + MICROPHONE=0, HEADPHONE=0, 0 > + MICROPHONE=0, HEADPHONE=1, 1 > + MICROPHONE=1, HEADPHONE=0, 2 > + MICROPHONE=1, HEADPHONE=1, 3 > + > + The speaker function is selected when HEADPHONE=0 because > + both (0 & 1) == (2 & 1) == 0. > + > + The headphones function is selected when HEADPHONE=1 and > + MICROPHONE=0 because (1 & 3) == 1. > + > + The headset function is selected when both HEADPHONE=1 and > + MICROPHONE=1 because (3 & 3) == 3. > + > + required: > + - state-inputs > + - state-input-bits > + - state-input-defaults > + - state-map > + > +patternProperties: > + "^.*dai-link-[0-9]+$": > + type: object > + description: |- > + cpu and codec child nodes: > + Container for cpu and codec dai sub-nodes. > + One cpu and one codec sub-node must exist. > + > + properties: > + link-name: > + description: The link name > + > + cpu: > + type: object > + properties: > + > + sound-dai: > + $ref: /schemas/types.yaml#/definitions/phandle-array > + description: phandle(s) of the CPU DAI(s) > + > + required: > + - sound-dai > + > + codec: > + type: object > + properties: > + > + sound-dai: > + $ref: /schemas/types.yaml#/definitions/phandle-array > + description: phandle(s) of the codec DAI(s) > + > + required: > + - sound-dai > + > + required: > + - link-name > + - cpu > + - codec > + > +required: > + - compatible > + - reg > + - reg-names > + - qcom,model > + > +additionalProperties: false > + > +examples: > + - | > + #include <dt-bindings/sound/apq8016-lpass.h> > + > + sound: sound@7702000 { > + compatible = "qcom,apq8039-sndcard"; > + reg = <0x07702000 0x4>, <0x07702004 0x4>; > + reg-names = "mic-iomux", "spkr-iomux"; > + > + status = "okay"; > + pinctrl-0 = <&cdc_pdm_lines_act>; > + pinctrl-1 = <&cdc_pdm_lines_sus>; > + pinctrl-names = "default", "sleep"; > + qcom,model = "APQ8039"; > + qcom,audio-routing = "AMIC2", "MIC BIAS Internal2"; > + > + internal-codec-playback-dai-link-0 { > + link-name = "WCD"; > + cpu { > + sound-dai = <&lpass MI2S_PRIMARY>; > + }; > + codec { > + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; > + }; > + }; > + > + internal-codec-capture-dai-link-0 { > + link-name = "WCD-Capture"; > + cpu { > + sound-dai = <&lpass MI2S_TERTIARY>; > + }; > + codec { > + sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; > + }; > + }; > + > + function-definition { > + > + mixer-control = "Jack Function"; > + function-list = <&auto &headphones &headset &speaker &off>; > + system-list = <&auto &off &off>; // default, init, shutdown > + > + auto: Automatic { > + // Headphone presence bit 0 (1) - H > + // Microphone presence bit 1 (2) - M > + state-inputs = "JACK_HEADPHONE", "JACK_MICROPHONE"; > + state-input-bits = <1>, <1>; > + state-input-defaults = <0>, <0>; > + > + // HM & MASK > + state-map = > + <&speaker 0x1 0x0>, // no headphone -> speaker > + <&headphones 0x3 0x1>, // headphone but no mic -> headphones > + <&headset 0x3 0x3>; // headphone & mic -> headset > + }; > + headphones: Headphones { > + enable = > + "RX1 MIX1 INP1", "RX1", > + "RX2 MIX1 INP1", "RX2", > + "RDAC2 MUX", "RX2", > + "RX1 Digital Volume", "128", > + "RX2 Digital Volume", "128", > + "HPHL", "Switch", > + "HPHR", "Switch"; > + > + disable = > + "RX1 Digital Volume", "0", > + "RX2 Digital Volume", "0", > + "HPHL", "ZERO", > + "HPHR", "ZERO", > + "RDAC2 MUX", "RX1", > + "RX1 MIX1 INP1", "ZERO", > + "RX2 MIX1 INP1", "ZERO"; > + }; > + headset: Headset { > + enable = > + "RX1 MIX1 INP1", "RX1", > + "RX2 MIX1 INP1", "RX2", > + "RDAC2 MUX", "RX2", > + "RX1 Digital Volume", "128", > + "RX2 Digital Volume", "128", > + "DEC1 MUX", "ADC2", > + "CIC1 MUX", "AMIC", > + "ADC2 Volume", "8", > + "ADC2 MUX", "INP2", > + "HPHL", "Switch", > + "HPHR", "Switch"; > + > + disable = > + "RX1 Digital Volume", "0", > + "RX2 Digital Volume", "0", > + "HPHL", "ZERO", > + "HPHR", "ZERO", > + "RDAC2 MUX", "RX1", > + "RX1 MIX1 INP1", "ZERO", > + "RX2 MIX1 INP1", "ZERO", > + "ADC2 MUX", "ZERO", > + "ADC2 Volume", "0", > + "DEC1 MUX", "ZERO"; > + }; > + speaker: Speaker { > + enable = > + "SPK DAC Switch", "1", > + "RX3 MIX1 INP1", "RX1", > + "RX3 MIX1 INP2", "RX2", > + "RX3 Digital Volume", "128"; > + > + disable = > + "SPK DAC Switch", "0", > + "RX3 MIX1 INP1", "ZERO", > + "RX3 MIX1 INP2", "ZERO"; > + }; > + off: Off { > + enable = > + "RX1 Digital Volume", "0", > + "RX2 Digital Volume", "0", > + "HPHL", "ZERO", > + "HPHR", "ZERO", > + "RDAC2 MUX", "RX1", > + "RX1 MIX1 INP1", "ZERO", > + "RX2 MIX1 INP1", "ZERO", > + "ADC2 MUX", "ZERO", > + "ADC2 Volume", "0", > + "DEC1 MUX", "ZERO", > + "SPK DAC Switch", "0", > + "RX3 MIX1 INP1", "ZERO", > + "RX3 MIX1 INP2", "ZERO"; > + }; > + }; This looks much like a replacement for ALSA UCM and userspace audio jack detection coded into the device tree. While I personally think this is an interesting idea (We have the device tree to describe the hardware, why can we not also describe necessary audio routing to enable a particular output?) this is also not really specific to the APQ8039 hardware, is it? In fact, without all the code to handle the mixer enable/disable sequences the machine driver looks almost identical to the existing apq8016-sbc. If you want to discuss ways to integrate mixer enable/disable sequences into the device tree, I suggest that you post your ideas separately as [RFC] with a more generic subject. That will make it more easy for everyone interested to share their thoughts. Right now it's quite hidden in a patch set where the subjects suggest that it's just a simple machine driver to glue some codecs together. Thanks, Stephan

3 7

[PATCH 00/16] ASoC: merge snd_soc_component_read() and snd_soc_component_read32()
by Kuninori Morimoto 22 Jun '20

22 Jun '20

Hi Mark Because we can use "read" function which is using "regmap" or "driver", current ALSA SoC drivers are using both snd_soc_component_read() // for regmap snd_soc_component_read32() // for driver callback These are similar but needs different parameter. This patch aggressively merge snd_soc_component_read() and _read32(), and makes snd_soc_component_read/write() as generally style. New read doesn't return error if it failed, thus, it can't keep compatibility, but assuming it is not a big problem. Because 1) it will indicate error message, 2) it can do nothing anyway if it fails "read". [02/16] patch is not directly connected to read function merging, but need to be apply after [01/16]. Kuninori Morimoto (16): ASoC: soc-component: merge snd_soc_component_read() and snd_soc_component_read32() ASoC: soc-component: use io_mutex correctly ASoC: pxa: rename to snd_soc_component_read() ASoC: atmel: rename to snd_soc_component_read() ASoC: codecs: rename to snd_soc_component_read() ASoC: codecs: wcd*: rename to snd_soc_component_read() ASoC: codecs: tlv*: rename to snd_soc_component_read() ASoC: codecs: max*: rename to snd_soc_component_read() ASoC: codecs: msm*: rename to snd_soc_component_read() ASoC: codecs: alc*: rename to snd_soc_component_read() ASoC: codecs: wm*: rename to snd_soc_component_read() ASoC: codecs: rt*: rename to snd_soc_component_read() ASoC: codecs: da*: rename to snd_soc_component_read() ASoC: codecs: cs*: rename to snd_soc_component_read() ASoC: codecs: ak*: rename to snd_soc_component_read() ASoC: remove snd_soc_component_read32() include/sound/soc-component.h | 4 +- sound/soc/atmel/atmel-pdmic.c | 4 +- sound/soc/codecs/88pm860x-codec.c | 14 ++--- sound/soc/codecs/ab8500-codec.c | 8 +-- sound/soc/codecs/ad1980.c | 4 +- sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak4535.c | 4 +- sound/soc/codecs/ak4613.c | 4 +- sound/soc/codecs/ak4671.c | 8 +-- sound/soc/codecs/alc5623.c | 6 +- sound/soc/codecs/alc5632.c | 6 +- sound/soc/codecs/arizona.c | 18 +++--- sound/soc/codecs/cs4270.c | 10 +-- sound/soc/codecs/cs42l42.c | 2 +- sound/soc/codecs/cs42l51.c | 8 +-- sound/soc/codecs/cs42l73.c | 4 +- sound/soc/codecs/cs47l35.c | 10 +-- sound/soc/codecs/cs47l85.c | 10 +-- sound/soc/codecs/da7210.c | 24 ++++---- sound/soc/codecs/da7213.c | 24 ++++---- sound/soc/codecs/da7218.c | 34 +++++------ sound/soc/codecs/da7219-aad.c | 16 ++--- sound/soc/codecs/da7219.c | 20 +++--- sound/soc/codecs/da732x.c | 18 +++--- sound/soc/codecs/da9055.c | 14 ++--- sound/soc/codecs/inno_rk3036.c | 6 +- sound/soc/codecs/madera.c | 49 +++------------ sound/soc/codecs/max98088.c | 12 ++-- sound/soc/codecs/max98090.c | 20 +++--- sound/soc/codecs/max98095.c | 16 ++--- sound/soc/codecs/max9850.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 14 ++--- sound/soc/codecs/msm8916-wcd-digital.c | 16 ++--- sound/soc/codecs/nau8822.c | 6 +- sound/soc/codecs/rt1305.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5631.c | 32 +++++----- sound/soc/codecs/rt5640.c | 10 +-- sound/soc/codecs/rt5645.c | 16 ++--- sound/soc/codecs/rt5651.c | 6 +- sound/soc/codecs/rt5659.c | 14 ++--- sound/soc/codecs/rt5660.c | 2 +- sound/soc/codecs/rt5663.c | 34 +++++------ sound/soc/codecs/rt5665.c | 16 ++--- sound/soc/codecs/rt5668.c | 16 ++--- sound/soc/codecs/rt5670.c | 18 +++--- sound/soc/codecs/rt5682-i2c.c | 2 +- sound/soc/codecs/rt5682.c | 19 +++--- sound/soc/codecs/sgtl5000.c | 16 ++--- sound/soc/codecs/sta32x.c | 4 +- sound/soc/codecs/tas2552.c | 4 +- sound/soc/codecs/tas5720.c | 4 +- sound/soc/codecs/tda7419.c | 9 +-- sound/soc/codecs/tlv320aic23.c | 14 ++--- sound/soc/codecs/tlv320aic26.c | 4 +- sound/soc/codecs/tlv320aic32x4.c | 16 ++--- sound/soc/codecs/tlv320aic3x.c | 14 ++--- sound/soc/codecs/tscs42xx.c | 4 +- sound/soc/codecs/tscs454.c | 24 ++------ sound/soc/codecs/wcd-clsh-v2.c | 2 +- sound/soc/codecs/wcd9335.c | 48 +++++++-------- sound/soc/codecs/wcd934x.c | 52 ++++++++-------- sound/soc/codecs/wm2200.c | 4 +- sound/soc/codecs/wm5100.c | 18 +++--- sound/soc/codecs/wm5110.c | 6 +- sound/soc/codecs/wm8350.c | 32 +++++----- sound/soc/codecs/wm8400.c | 50 +++++++-------- sound/soc/codecs/wm8510.c | 28 ++++----- sound/soc/codecs/wm8523.c | 6 +- sound/soc/codecs/wm8580.c | 12 ++-- sound/soc/codecs/wm8711.c | 8 +-- sound/soc/codecs/wm8728.c | 10 +-- sound/soc/codecs/wm8731.c | 6 +- sound/soc/codecs/wm8750.c | 8 +-- sound/soc/codecs/wm8753.c | 42 ++++++------- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 22 +++---- sound/soc/codecs/wm8903.c | 20 +++--- sound/soc/codecs/wm8904.c | 16 ++--- sound/soc/codecs/wm8940.c | 32 +++++----- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8958-dsp2.c | 18 +++--- sound/soc/codecs/wm8960.c | 20 +++--- sound/soc/codecs/wm8961.c | 58 +++++++++--------- sound/soc/codecs/wm8962.c | 31 +++++----- sound/soc/codecs/wm8971.c | 8 +-- sound/soc/codecs/wm8974.c | 24 ++++---- sound/soc/codecs/wm8978.c | 12 ++-- sound/soc/codecs/wm8983.c | 8 +-- sound/soc/codecs/wm8985.c | 8 +-- sound/soc/codecs/wm8988.c | 12 ++-- sound/soc/codecs/wm8990.c | 18 +++--- sound/soc/codecs/wm8991.c | 38 ++++++------ sound/soc/codecs/wm8993.c | 28 ++++----- sound/soc/codecs/wm8994.c | 60 +++++++++--------- sound/soc/codecs/wm8995.c | 16 ++--- sound/soc/codecs/wm8996.c | 30 ++++----- sound/soc/codecs/wm8998.c | 8 +-- sound/soc/codecs/wm9081.c | 36 +++++------ sound/soc/codecs/wm9090.c | 4 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_hubs.c | 30 ++++----- sound/soc/fsl/fsl_audmix.c | 10 +-- sound/soc/fsl/fsl_easrc.c | 5 +- sound/soc/meson/aiu-encoder-i2s.c | 3 +- sound/soc/meson/aiu-fifo-i2s.c | 3 +- sound/soc/meson/aiu-fifo.c | 3 +- sound/soc/pxa/mioa701_wm9713.c | 8 +-- sound/soc/soc-ac97.c | 7 +-- sound/soc/soc-component.c | 84 ++++++++++++++------------ sound/soc/soc-dapm.c | 31 ++++------ sound/soc/soc-ops.c | 43 +++---------- 115 files changed, 854 insertions(+), 963 deletions(-) -- 2.25.1

5 23

[PATCH] ASoC: wm_adsp: Add controls for calibration and diagnostic FW
by David Rhodes 22 Jun '20

22 Jun '20

From: Vlad Karpovich <Vlad.Karpovich(a)cirrus.com> Exposed additional mixer controls to select calibration or diagnostic firmware. 'Calibration' --> chip-dsp<id>-spk-cali.wmfw (.bin) 'Diagnostic' --> chip-dsp<id>-spk-diag.wmfw (.bin) Signed-off-by: Vlad Karpovich <Vlad.Karpovich(a)cirrus.com> Signed-off-by: David Rhodes <david.rhodes(a)cirrus.com> --- sound/soc/codecs/wm_adsp.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4e63defa6857..d9d2bb5a1945 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -363,9 +363,11 @@ static void wm_adsp_buf_free(struct list_head *list) #define WM_ADSP_FW_ASR 7 #define WM_ADSP_FW_TRACE 8 #define WM_ADSP_FW_SPK_PROT 9 -#define WM_ADSP_FW_MISC 10 +#define WM_ADSP_FW_SPK_CALI 10 +#define WM_ADSP_FW_SPK_DIAG 11 +#define WM_ADSP_FW_MISC 12 -#define WM_ADSP_NUM_FW 11 +#define WM_ADSP_NUM_FW 13 static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { [WM_ADSP_FW_MBC_VSS] = "MBC/VSS", @@ -378,6 +380,8 @@ static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { [WM_ADSP_FW_ASR] = "ASR Assist", [WM_ADSP_FW_TRACE] = "Dbg Trace", [WM_ADSP_FW_SPK_PROT] = "Protection", + [WM_ADSP_FW_SPK_CALI] = "Calibration", + [WM_ADSP_FW_SPK_DIAG] = "Diagnostic", [WM_ADSP_FW_MISC] = "Misc", }; @@ -612,6 +616,8 @@ static const struct { .caps = trace_caps, }, [WM_ADSP_FW_SPK_PROT] = { .file = "spk-prot" }, + [WM_ADSP_FW_SPK_CALI] = { .file = "spk-cali" }, + [WM_ADSP_FW_SPK_DIAG] = { .file = "spk-diag" }, [WM_ADSP_FW_MISC] = { .file = "misc" }, }; -- 2.25.1

3 2

GQuark is not exported.
by GitHub issues - edited 22 Jun '20

22 Jun '20

alsa-project/alsa-gobject issue #47 was edited from takaswie: Although each library in alsa-gobject uses GQuark to propagate error information, the symbol of GQuark is not exported yet by symbol mapping file of each library: * src/ctl/alsactl.map * src/timer/alsatimer.map * src/seq/alsaseq.map * src/hwdep/alsahwdep.map * src/rawmidi/alsarawmidi.map This issue brings inconvenience for [Rust bindings](https://github.com/alsa-project/alsa-gobject-rs) to implement [glib::error::ErrorDomain](https://gtk-rs.org/docs/glib/error/trait.ErrorDom… for each GQuark. The glib::error::ErrorDomain trait requires domain function implementation to return GQuark itself, however it's not exported and unable to access. Issue URL : https://github.com/alsa-project/alsa-gobject/issues/47 Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

GQuark is not exported.
by GitHub issues - opened 22 Jun '20

22 Jun '20

alsa-project/alsa-gobject issue #47 was opened from takaswie: Although each library in alsa-gobject uses GQuark to propagate error information, the symbol of GQuark is not exported yet by symbol mapping file of each library: * src/ctl/alsactl.map * src/timer/alsatimer.map * src/seq/alsaseq.map * src/hwdep/alsahwdep.map * src/rawmidi/alsarawmidi.map This issue brings inconvenience for [Rust bindings](https://github.com/alsa-project/alsa-gobject-rs) to implement [glib::error::ErrorDomain](https://gtk-rs.org/docs/glib/error/trait.ErrorDom… for each GQuark. Issue URL : https://github.com/alsa-project/alsa-gobject/issues/47 Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ASoC: amd: add logic to check dmic hardware runtime
by Vijendar Mukunda 22 Jun '20

22 Jun '20

Add logic to check DMIC hardware exists or not on the platform at runtime. Add module param for overriding DMIC hardware check at runtime. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 29 +++++++++++++++++++++++++++++ sound/soc/amd/renoir/rn_acp3x.h | 7 +++++++ 2 files changed, 36 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 859ed67..ef6eeba 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -5,6 +5,7 @@ //Copyright 2020 Advanced Micro Devices, Inc. #include <linux/pci.h> +#include <linux/acpi.h> #include <linux/module.h> #include <linux/io.h> #include <linux/delay.h> @@ -18,6 +19,16 @@ static int acp_power_gating; module_param(acp_power_gating, int, 0644); MODULE_PARM_DESC(acp_power_gating, "Enable acp power gating"); +/** + * dmic_acpi_check = 0 - Skips DMIC device creation and returns probe failure + * = 1 - Assumes that platform has DMIC support and skips ACPI + * method check + * = 2 - Checks ACPI method to know DMIC hardware status runtime + */ +static int dmic_acpi_check = ACP_DMIC_AUTO; +module_param(dmic_acpi_check, int, 0644); +MODULE_PARM_DESC(dmic_acpi_check, "checks Dmic hardware runtime"); + struct acp_dev_data { void __iomem *acp_base; struct resource *res; @@ -157,6 +168,8 @@ static int snd_rn_acp_probe(struct pci_dev *pci, { struct acp_dev_data *adata; struct platform_device_info pdevinfo[ACP_DEVS]; + acpi_handle handle; + acpi_integer dmic_status; unsigned int irqflags; int ret, index; u32 addr; @@ -201,6 +214,22 @@ static int snd_rn_acp_probe(struct pci_dev *pci, if (ret) goto disable_msi; + if (!dmic_acpi_check) { + ret = -ENODEV; + goto de_init; + } else if (dmic_acpi_check == ACP_DMIC_AUTO) { + handle = ACPI_HANDLE(&pci->dev); + ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); + if (ACPI_FAILURE(ret)) { + ret = -EINVAL; + goto de_init; + } + if (!dmic_status) { + ret = -ENODEV; + goto de_init; + } + } + adata->res = devm_kzalloc(&pci->dev, sizeof(struct resource) * 2, GFP_KERNEL); diff --git a/sound/soc/amd/renoir/rn_acp3x.h b/sound/soc/amd/renoir/rn_acp3x.h index 75228e3..fcc5dca 100644 --- a/sound/soc/amd/renoir/rn_acp3x.h +++ b/sound/soc/amd/renoir/rn_acp3x.h @@ -55,6 +55,13 @@ #define MAX_BUFFER (CAPTURE_MAX_PERIOD_SIZE * CAPTURE_MAX_NUM_PERIODS) #define MIN_BUFFER MAX_BUFFER + +enum acp_dmic_audio { + ACP_DMIC_DISABLE = 0, + ACP_DMIC_ENABLE, + ACP_DMIC_AUTO +}; + struct pdm_dev_data { u32 pdm_irq; void __iomem *acp_base; -- 2.7.4

3 2