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

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

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[PATCH 0/3] ALSA: compress: Document stream states and fix gaplless SM
by Vinod Koul 22 Jun '20

22 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. 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

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[GIT PULL] ASoC fixes for v5.8-rc2
by broonie＠kernel.org 22 Jun '20

22 Jun '20

The following changes since commit 44ce45f866c1606f03a411fb2d2c0db4626403a6: Merge series "ASoC: Fix dailink checks for DPCM" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: (2020-06-09 15:46:20 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.8-rc2 for you to fetch changes up to f141a422159a199f4c8dedb7e0df55b3b2cf16cd: ASoC: rockchip: Fix a reference count leak. (2020-06-18 17:21:58 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.8 This is a collection of mostly small fixes, mostly fixing fallout from some of the DPCM changes that went in last time around which shook out some issues on i.MX and Qualcomm platforms. The addition of a managed version of snd_soc_register_dai() is to fix resource leaks. There's also a few new device IDs for x86 systems. ---------------------------------------------------------------- Brent Lu (1): ASoC: SOF: Intel: hda: Clear RIRB status before reading WP Jack Yu (1): ASoC: rt1015: Update rt1015 default register value according to spec modification. Mark Brown (2): Merge series "ASoC: topology: fix use-after-free when removing components" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: SOF: Intel: update PCI IDs" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Pierre-Louis Bossart (7): ASoC: soc-devres: add devm_snd_soc_register_dai() ASoC: soc-topology: use devm_snd_soc_register_dai() ASoC: soc-pcm: fix checks for multi-cpu FE dailinks ASoC: Intel: SOF: merge COMETLAKE_LP and COMETLAKE_H ASoC: SOF: Intel: add PCI ID for CometLake-S ASoC: SOF: Intel: add PCI IDs for ICL-H and TGL-H ASoC: hdac_hda: fix memleak with regmap not freed on remove Qiushi Wu (1): ASoC: rockchip: Fix a reference count leak. Ravulapati Vishnu vardhan rao (1): ASoC: amd: closing specific instance. Shengjiu Wang (5): ASoC: soc-card: export snd_soc_lookup_component_nolocked ASoC: dmaengine_pcm: export soc_component_to_pcm ASoC: fsl_asrc_dma: Reuse the dma channel if available in Back-End ASoC: fsl_asrc_dma: Fix data copying speed issue with EDMA ASoC: fsl_ssi: Fix bclk calculation for mono channel Srinivas Kandagatla (3): ASoC: q6asm: handle EOS correctly ASoc: q6afe: add support to get port direction ASoC: qcom: common: set correct directions for dailinks Steve Lee (1): ASoC: max98390: Update regmap readable reg and volatile derek.fang (1): ASoC: rt5682: Let dai clks be registered whether mclk exists or not include/sound/dmaengine_pcm.h | 11 +++ include/sound/soc.h | 6 ++ sound/hda/intel-dsp-config.c | 4 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 30 ++++++-- sound/soc/codecs/hdac_hda.c | 16 +++-- sound/soc/codecs/max98390.c | 6 +- sound/soc/codecs/rt1015.c | 124 ++++++++++++++++++++-------------- sound/soc/codecs/rt1015.h | 15 +++- sound/soc/codecs/rt5682.c | 11 +-- sound/soc/fsl/fsl_asrc_common.h | 2 + sound/soc/fsl/fsl_asrc_dma.c | 47 +++++++++---- sound/soc/fsl/fsl_ssi.c | 13 ++-- sound/soc/intel/boards/Kconfig | 4 +- sound/soc/qcom/common.c | 14 +++- sound/soc/qcom/qdsp6/q6afe.c | 8 +++ sound/soc/qcom/qdsp6/q6afe.h | 1 + sound/soc/qcom/qdsp6/q6asm.c | 7 +- sound/soc/rockchip/rockchip_pdm.c | 4 +- sound/soc/soc-core.c | 3 +- sound/soc/soc-devres.c | 37 ++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 12 ---- sound/soc/soc-pcm.c | 6 +- sound/soc/soc-topology.c | 3 +- sound/soc/sof/intel/Kconfig | 29 +++----- sound/soc/sof/intel/hda-stream.c | 9 ++- sound/soc/sof/sof-pci-dev.c | 24 ++++--- 26 files changed, 295 insertions(+), 151 deletions(-)

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[PATCH AUTOSEL 5.7 130/388] ASoC: Fix wrong dependency of da7210 and wm8983
by Sasha Levin 22 Jun '20

22 Jun '20

From: Wei Li <liwei391(a)huawei.com> [ Upstream commit c1c050ee74d67aeb879fd38e3a07139d7fdb79f4 ] As these two drivers support I2C and SPI, we should add the SND_SOC_I2C_AND_SPI dependency instead. Fixes: ce0c97f8a2936 ("ASoC: Fix SND_SOC_ALL_CODECS imply SPI fallout") Signed-off-by: Wei Li <liwei391(a)huawei.com> Link: https://lore.kernel.org/r/20200420202410.47327-3-liwei391@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8cdc68c141dc..7d2cbed55a9d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -717,7 +717,7 @@ config SND_SOC_L3 config SND_SOC_DA7210 tristate - depends on I2C + depends on SND_SOC_I2C_AND_SPI config SND_SOC_DA7213 tristate "Dialog DA7213 CODEC" @@ -1569,7 +1569,7 @@ config SND_SOC_WM8978 config SND_SOC_WM8983 tristate - depends on I2C + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8985 tristate "Wolfson Microelectronics WM8985 and WM8758 codec driver" -- 2.25.1

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[PATCH] soundwire: qcom: Constify static structs
by Rikard Falkeborn 22 Jun '20

22 Jun '20

qcom_swrm_port_ops and qcom_swrm_ops are not modified and can be made const to allow the compiler to put them in read-only memory. Before: text data bss dec hex filename 18266 3056 256 21578 544a drivers/soundwire/qcom.o After: text data bss dec hex filename 18426 2896 256 21578 544a drivers/soundwire/qcom.o Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- drivers/soundwire/qcom.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index a1c2a44a3b4d..915c2cf0c274 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -406,13 +406,13 @@ static int qcom_swrm_port_enable(struct sdw_bus *bus, return ctrl->reg_write(ctrl, reg, val); } -static struct sdw_master_port_ops qcom_swrm_port_ops = { +static const struct sdw_master_port_ops qcom_swrm_port_ops = { .dpn_set_port_params = qcom_swrm_port_params, .dpn_set_port_transport_params = qcom_swrm_transport_params, .dpn_port_enable_ch = qcom_swrm_port_enable, }; -static struct sdw_master_ops qcom_swrm_ops = { +static const struct sdw_master_ops qcom_swrm_ops = { .xfer_msg = qcom_swrm_xfer_msg, .pre_bank_switch = qcom_swrm_pre_bank_switch, }; -- 2.27.0

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