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[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

[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

3 11

[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(-)

2 1

[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

2 4

[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

2 1

[PATCH v2 0/6] soundwire: intel: transition to 3 steps initialization
by Bard Liao 22 Jun '20

22 Jun '20

This series is to split the original "soundwire: intel: transition to 3 steps initialization" patch into different patches for better review. It also address comments from Vinod. Pierre-Louis Bossart (6): soundwire: intel: cleanups for indirections/logs soundwire: intel: clarify drvdata and remove more indirections soundwire: intel_init: remove useless test soundwire: intel_init: use devm_ allocation soundwire: intel_init: pass link information as platform data soundwire: intel: transition to 3 steps initialization drivers/soundwire/intel.c | 138 +++++++++-------- drivers/soundwire/intel.h | 13 ++ drivers/soundwire/intel_init.c | 261 ++++++++++++++++++++++++++------- 3 files changed, 295 insertions(+), 117 deletions(-) -- 2.17.1

2 7

[PATCH] soundwire: intel: fix memory leak with devm_kasprintf
by Pierre-Louis Bossart 22 Jun '20

22 Jun '20

The dais are allocated with devm_kcalloc() but their name isn't resourced managed and never freed. Fix by also using devm_ for the dai names as well. Fixes: c46302ec554c5 ('soundwire: intel: Add audio DAI ops') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 4cfdd074e310..c7422740edd4 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -930,8 +930,9 @@ static int intel_create_dai(struct sdw_cdns *cdns, /* TODO: Read supported rates/formats from hardware */ for (i = off; i < (off + num); i++) { - dais[i].name = kasprintf(GFP_KERNEL, "SDW%d Pin%d", - cdns->instance, i); + dais[i].name = devm_kasprintf(cdns->dev, GFP_KERNEL, + "SDW%d Pin%d", + cdns->instance, i); if (!dais[i].name) return -ENOMEM; -- 2.20.1

2 1

[PATCH] soundwire: bus: clock_stop: don't deal with UNATTACHED Slave devices
by Bard Liao 22 Jun '20

22 Jun '20

We don't need to do anything for the slave if it is unattached during clock stop prepare and exit sequences. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 23 +++++++++++++++++------ 1 file changed, 17 insertions(+), 6 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 2289c2ac8c5a..405335fb790a 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -863,13 +863,13 @@ int sdw_bus_prep_clk_stop(struct sdw_bus *bus) if (!slave->dev_num) continue; - /* Identify if Slave(s) are available on Bus */ - is_slave = true; - if (slave->status != SDW_SLAVE_ATTACHED && slave->status != SDW_SLAVE_ALERT) continue; + /* Identify if Slave(s) are available on Bus */ + is_slave = true; + slave_mode = sdw_get_clk_stop_mode(slave); slave->curr_clk_stop_mode = slave_mode; @@ -900,6 +900,10 @@ int sdw_bus_prep_clk_stop(struct sdw_bus *bus) return ret; } + /* Don't need to inform slaves if there is no slave attached */ + if (!is_slave) + return ret; + /* Inform slaves that prep is done */ list_for_each_entry(slave, &bus->slaves, node) { if (!slave->dev_num) @@ -985,13 +989,13 @@ int sdw_bus_exit_clk_stop(struct sdw_bus *bus) if (!slave->dev_num) continue; - /* Identify if Slave(s) are available on Bus */ - is_slave = true; - if (slave->status != SDW_SLAVE_ATTACHED && slave->status != SDW_SLAVE_ALERT) continue; + /* Identify if Slave(s) are available on Bus */ + is_slave = true; + mode = slave->curr_clk_stop_mode; if (mode == SDW_CLK_STOP_MODE1) { @@ -1016,6 +1020,13 @@ int sdw_bus_exit_clk_stop(struct sdw_bus *bus) if (is_slave && !simple_clk_stop) sdw_bus_wait_for_clk_prep_deprep(bus, SDW_BROADCAST_DEV_NUM); + /* + * Don't need to call slave callback function if there is no slave + * attached + */ + if (!is_slave) + return 0; + list_for_each_entry(slave, &bus->slaves, node) { if (!slave->dev_num) continue; -- 2.17.1

2 1

[PATCH alsa-lib 1/2] pcm: dmix: Fix semaphore usage
by Takashi Iwai 22 Jun '20

22 Jun '20

As Maarten Baert recently reported, the current dmix code applies the semaphore unnecessarily around mixing streams even when the lockless mix operation is used on x86. This was rather introduced mistakenly at the commit 267d7c728196 ("Add support of little-endian on i386/x86_64 dmix") where the generic dmix code was included on x86, too. For achieving the original performance back, this patch changes the semaphore handling to be checked at run time instead of statically at compile time. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.h | 1 + src/pcm/pcm_dmix.c | 18 +++++++++++------- src/pcm/pcm_dmix_generic.c | 2 +- src/pcm/pcm_dmix_i386.c | 1 + src/pcm/pcm_dmix_x86_64.c | 1 + 5 files changed, 15 insertions(+), 8 deletions(-) diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 8a236970a3a1..2150bce15449 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -186,6 +186,7 @@ struct snd_pcm_direct { mix_areas_32_t *remix_areas_32; mix_areas_24_t *remix_areas_24; mix_areas_u8_t *remix_areas_u8; + unsigned int use_sem; } dmix; struct { unsigned long long chn_mask; diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 843fa3168756..e9343b19a536 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -292,13 +292,17 @@ static void remix_areas(snd_pcm_direct_t *dmix, * the area via semaphore */ #ifndef DOC_HIDDEN -#ifdef NO_CONCURRENT_ACCESS -#define dmix_down_sem(dmix) snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT) -#define dmix_up_sem(dmix) snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT) -#else -#define dmix_down_sem(dmix) -#define dmix_up_sem(dmix) -#endif +static void dmix_down_sem(snd_pcm_direct_t *dmix) +{ + if (dmix->u.dmix.use_sem) + snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); +} + +static void dmix_up_sem(snd_pcm_direct_t *dmix) +{ + if (dmix->u.dmix.use_sem) + snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); +} #endif /* diff --git a/src/pcm/pcm_dmix_generic.c b/src/pcm/pcm_dmix_generic.c index 40c08747a74a..8a5b6f148556 100644 --- a/src/pcm/pcm_dmix_generic.c +++ b/src/pcm/pcm_dmix_generic.c @@ -43,7 +43,6 @@ static inline unsigned long __cmpxchg(volatile void *ptr, unsigned long old, #ifndef ARCH_ADD #define ARCH_ADD(p,a) (*(p) += (a)) #define ARCH_CMPXCHG(p,a,b) (*(p)) /* fake */ -#define NO_CONCURRENT_ACCESS /* use semaphore to avoid race */ #define IS_CONCURRENT 0 /* no race check */ #endif @@ -530,6 +529,7 @@ static void generic_mix_select_callbacks(snd_pcm_direct_t *dmix) dmix->u.dmix.mix_areas_u8 = generic_mix_areas_u8; dmix->u.dmix.remix_areas_24 = generic_remix_areas_24; dmix->u.dmix.remix_areas_u8 = generic_remix_areas_u8; + dmix->u.dmix.use_sem = 1; } #endif diff --git a/src/pcm/pcm_dmix_i386.c b/src/pcm/pcm_dmix_i386.c index 1ab983a8a373..82a91c5c2897 100644 --- a/src/pcm/pcm_dmix_i386.c +++ b/src/pcm/pcm_dmix_i386.c @@ -135,4 +135,5 @@ static void mix_select_callbacks(snd_pcm_direct_t *dmix) dmix->u.dmix.mix_areas_24 = smp > 1 ? mix_areas_24_smp: mix_areas_24; dmix->u.dmix.remix_areas_24 = smp > 1 ? remix_areas_24_smp: remix_areas_24; } + dmix->u.dmix.use_sem = 0; } diff --git a/src/pcm/pcm_dmix_x86_64.c b/src/pcm/pcm_dmix_x86_64.c index 34c40d4e9d1d..4d882bfd01bf 100644 --- a/src/pcm/pcm_dmix_x86_64.c +++ b/src/pcm/pcm_dmix_x86_64.c @@ -102,4 +102,5 @@ static void mix_select_callbacks(snd_pcm_direct_t *dmix) dmix->u.dmix.remix_areas_32 = smp > 1 ? remix_areas_32_smp : remix_areas_32; dmix->u.dmix.mix_areas_24 = smp > 1 ? mix_areas_24_smp : mix_areas_24; dmix->u.dmix.remix_areas_24 = smp > 1 ? remix_areas_24_smp : remix_areas_24; + dmix->u.dmix.use_sem = 0; } -- 2.16.4

3 3

alsa-gobject: bump version up to 0.1.0 for official release
by GitHub pull_request - opened 22 Jun '20

22 Jun '20

alsa-project/alsa-gobject pull request #46 was opened from takaswie: This patch is a preparation for official v0.1.0 release. After merged, I'm going to push a tag 'v0.1.0' with release note. Request URL : https://github.com/alsa-project/alsa-gobject/pull/46 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/46.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable
by Shengjiu Wang 22 Jun '20

22 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. Fixes: 9e28f6532c61 ("ASoC: fsl_mqs: Add MQS component driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_mqs.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index 0c813a45bba7..69aeb0e71844 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -265,12 +265,20 @@ static int fsl_mqs_remove(struct platform_device *pdev) static int fsl_mqs_runtime_resume(struct device *dev) { struct fsl_mqs *mqs_priv = dev_get_drvdata(dev); + int ret; - if (mqs_priv->ipg) - clk_prepare_enable(mqs_priv->ipg); + ret = clk_prepare_enable(mqs_priv->ipg); + if (ret) { + dev_err(dev, "failed to enable ipg clock\n"); + return ret; + } - if (mqs_priv->mclk) - clk_prepare_enable(mqs_priv->mclk); + ret = clk_prepare_enable(mqs_priv->mclk); + if (ret) { + dev_err(dev, "failed to enable mclk clock\n"); + clk_disable_unprepare(mqs_priv->ipg); + return ret; + } if (mqs_priv->use_gpr) regmap_write(mqs_priv->regmap, IOMUXC_GPR2, @@ -292,11 +300,8 @@ static int fsl_mqs_runtime_suspend(struct device *dev) regmap_read(mqs_priv->regmap, REG_MQS_CTRL, &mqs_priv->reg_mqs_ctrl); - if (mqs_priv->mclk) - clk_disable_unprepare(mqs_priv->mclk); - - if (mqs_priv->ipg) - clk_disable_unprepare(mqs_priv->ipg); + clk_disable_unprepare(mqs_priv->mclk); + clk_disable_unprepare(mqs_priv->ipg); return 0; } -- 2.21.0

1 0

alsa-gobject: reorder documentation sections
by GitHub pull_request - opened 21 Jun '20

21 Jun '20

alsa-project/alsa-gobject pull request #45 was opened from takaswie: For better documentation, this patchset reorders sections of ALSATimer/ALSASeq documentation. Request URL : https://github.com/alsa-project/alsa-gobject/pull/45 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/45.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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alsa-gobject: add API to get current protocol version in each interface
by GitHub pull_request - opened 21 Jun '20

21 Jun '20

alsa-project/alsa-gobject pull request #44 was opened from takaswie: In UAPI of Linux sound subsystem, interfaces support protocol versioning. This patchset adds API to get current protocol version in each interface. Request URL : https://github.com/alsa-project/alsa-gobject/pull/44 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/44.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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alsa-gobject: fix indentation
by GitHub pull_request - opened 21 Jun '20

21 Jun '20

alsa-project/alsa-gobject pull request #43 was opened from takaswie: This patchset is trivial to apply fixes for 4-spaces indentation. Request URL : https://github.com/alsa-project/alsa-gobject/pull/43 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/43.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[Resend PATCH] Add implicit feedback quirk for SSL2.
by Laurence Tratt 21 Jun '20

21 Jun '20

As expected, this requires the same quirk as the SSL2+ in order for the clock to sync. This was suggested by, and tested on an SSL2, by Dmitry. Suggested-by: Dmitry <dpavlushko(a)gmail.com> Signed-off-by: Laurence Tratt <laurie(a)tratt.net> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git sound/usb/pcm.c sound/usb/pcm.c index 84c0ae4319..dc1608bdf6 100644 --- sound/usb/pcm.c +++ sound/usb/pcm.c @@ -367,6 +367,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 0; goto add_sync_ep_from_ifnum; case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ ep = 0x81; ifnum = 2; -- 2.27.0

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[HDAudio][ALC295] speakers not working on Acer Travelmate P614
by Stanislav Kazmin 21 Jun '20

21 Jun '20

Hello, I have the following problem on my Acer TravelMate P614-51T-G2: I have all the audio sinks correctly identified (like HDMI, DMic, Headphones) but the internal speakers do not produce any sound. What I have tried so far: - hard/soft shutdown on Windows and reboot - disable/re-enable speakers and microphone in BIOS - uninstall pulseaudio and test alsa alone (so it is defeitely **not** a pulseaudio issue) - switch from sof-hda-dsp to snd-hda-intel driver (without Dmic support) - retask pins 0x14, 0x16, 0x1b to "Internal Speakers" through hdajackretask (only basic, without advanced features) - removed/reconnected the headphones `alsamixer` shows all needed sinks and nothing is muted. I already discussed the issue at sof github https://github.com/thesofproject/ sof/issues/3058 but since the same issue occurs on `snd-hda-intel` legacy river, I was advised to communicate with alsa-devel team. The alsa-info.sh result are linked at http://alsa-project.org/db/? f=252f92c7a1df3c755d16ee69353b26d2535a4d81 I have tested the kernel 5.8-rc1 but it does not make any difference. Let me know if I can do anything else to test the issue.

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Lock-free dmix considered harmful
by Maarten Baert 20 Jun '20

20 Jun '20

I'm in the process of adding monitoring support to dmix (i.e. the ability for applications to capture the sound being played back through the speakers) - expect patches for this soon. However while reading the dmix source code I came across something weird. Dmix uses a lock-free mixing implementation based on (architecture-dependent) atomic operations such as atomic addition and compare-exchange. However at the same time dmix also uses a semaphore to prevent multiple processes from accessing the shared memory buffers simultaneously. This is redundant and makes the lock-free implementation rather pointless. Out of curiosity I decided to measure the time it takes to execute snd_pcm_dmix_sync_area for these three cases: - A simple locked implementation, which acquires the semaphore and then does mixing using the non-concurrent code from pcm_dmix_generic.c. - The current (redundant) implementation, which acquires the semaphore and then does atomic mixing. - A lock-free implementation, which does not acquire the semaphore and does atomic mixing. The resulting sound is identical in all three cases. (As a sanity check I also tested the 4th case, non-concurrent code without locking, which as expected produces audio with glitches.) I tested each case with 1, 4 and 20 simultaneous playback streams (multiple instances of 'aplay') on a system with 4 physical CPU cores (Intel Core i7-4770). You can see the results here: https://misc.maartenbaert.be/dmix_perf_locked_atomic.png The results are quite surprising: - The locked implementation is 3 to 6 times *faster* than the current redundant implementation. - The lock-free implementation is usually about as fast as the current redundant implementation, but has extreme outliers that are up to 3 times *slower* than the redundant implementation when there are multiple threads. It seems that the overhead of the atomic operations is so high that it completely negates the theoretical advantage of being able to mix from multiple threads simultaneously. This may be the result of 'false sharing' when multiple threads are accessing different words in the same cache line (which is almost impossible to avoid when mixing audio). On a related note, I believe that I have also found a bug in the lock-free implementation. The logic which is used to clear the 'sum' buffer is as follows: sample = *src; old_sample = *sum; if (ARCH_CMPXCHG(dst, 0, 1) == 0) sample -= old_sample; ARCH_ADD(sum, sample); This takes advantage of the fact that the hardware driver clears the playback buffer after reading it. However it does not account for the possibility that 'dst' might be zero for other reasons, such as: - An application plays back silence. - An application plays back sound, but then rewinds it. - Multiple applications play back sound which just happens to sum to zero occasionally. These all result in situations where 'dst' is changed from 0 to 1 by the compare-exchange operation, but then changed back to zero later, resulting in the classic 'ABA problem'. However this problem is currently hidden by the fact that the lock-free implementation is in fact still locked. Because of these reasons I think it would be better to drop the lock-free implementation entirely and just use the existing non-concurrent architecture-independent implementation from pcm_dmix_generic.c. Aside from being faster, it would also eliminate a lot of architecture-dependent code and inline assembly. Should I submit a patch for this? Best regards, Maarten Baert

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[PATCH] snd-usb-audio: add quirk for MacroSilicon MS2109
by Hector Martin 20 Jun '20

20 Jun '20

These devices claim to be 96kHz mono, but actually are 48kHz stereo with swapped channels and unaligned transfers. Signed-off-by: Hector Martin <marcan(a)marcan.st> Cc: stable(a)vger.kernel.org --- sound/usb/quirks-table.h | 52 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 52 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 4ec491011b19..9092cc0aa807 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3633,4 +3633,56 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ } }, +/* + * MacroSilicon MS2109 based HDMI capture cards + * + * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. + * They also need QUIRK_AUDIO_ALIGN_TRANSFER, which makes one wonder if + * they pretend to be 96kHz mono as a workaround for stereo being broken + * by that... + * + * They also have swapped L-R channels, but that's for userspace to deal + * with. + */ +{ + USB_DEVICE(0x534d, 0x2109), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MacroSilicon", + .product_name = "MS2109", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + }, + { + .ifnum = -1 + } + } + } +}, + #undef USB_DEVICE_VENDOR_SPEC -- 2.27.0

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How to populate required string as audio endpoint name in gnome audio devices tab
by Mukunda,Vijendar 20 Jun '20

20 Jun '20

Renoir platform has onboard dmic + south bridge HD Audio endpoint support. For onboard Dmic, Audio endpoint name is populated as "Analog Input" in Audio input devices. How can we populate name as "Internal Mic" or "Dmic" in audio input devices tab in sound settings? Alsa info link: https://pastebin.ubuntu.com/p/4kprhDZYbg/ Pluse audio info output: https://pastebin.ubuntu.com/p/sdx9Xs234C/ Thanks, Vijendar

3 3

Re: [PATCH 2/2] ASoC: qcom: add apq8039 sound card support
by kernel test robot 20 Jun '20

20 Jun '20

Hi Pantelis, I love your patch! Perhaps something to improve: [auto build test WARNING on asoc/for-next] [also build test WARNING on sound/for-next v5.8-rc1 next-20200618] [If your patch is applied to the wrong git tree, kindly drop us a note. And when submitting patch, we suggest to use as documented in https://git-scm.com/docs/git-format-patch] url: https://github.com/0day-ci/linux/commits/Pantelis-Antoniou/ASoC-qcom-add-ap… base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next config: i386-allyesconfig (attached as .config) compiler: gcc-9 (Debian 9.3.0-13) 9.3.0 reproduce (this is a W=1 build): # save the attached .config to linux build tree make W=1 ARCH=i386 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>, old ones prefixed by <<): In file included from arch/x86/include/asm/page_32.h:35, from arch/x86/include/asm/page.h:14, from arch/x86/include/asm/thread_info.h:12, from include/linux/thread_info.h:38, from arch/x86/include/asm/preempt.h:7, from include/linux/preempt.h:78, from include/linux/rcupdate.h:27, from include/linux/rculist.h:11, from include/linux/pid.h:5, from include/linux/sched.h:14, from include/linux/ratelimit.h:6, from include/linux/dev_printk.h:16, from include/linux/device.h:15, from sound/soc/qcom/apq8039.c:5: In function 'strncpy', inlined from 'snd_soc_card_ctl_getset.constprop' at sound/soc/qcom/apq8039.c:176:5: >> include/linux/string.h:297:30: warning: '__builtin_strncpy' destination unchanged after copying no bytes [-Wstringop-truncation] 297 | #define __underlying_strncpy __builtin_strncpy | ^ include/linux/string.h:307:9: note: in expansion of macro '__underlying_strncpy' 307 | return __underlying_strncpy(p, q, size); | ^~~~~~~~~~~~~~~~~~~~ vim +/__builtin_strncpy +297 include/linux/string.h 47227d27e2fcb0 Daniel Axtens 2020-06-03 275 47227d27e2fcb0 Daniel Axtens 2020-06-03 276 #ifdef CONFIG_KASAN 47227d27e2fcb0 Daniel Axtens 2020-06-03 277 extern void *__underlying_memchr(const void *p, int c, __kernel_size_t size) __RENAME(memchr); 47227d27e2fcb0 Daniel Axtens 2020-06-03 278 extern int __underlying_memcmp(const void *p, const void *q, __kernel_size_t size) __RENAME(memcmp); 47227d27e2fcb0 Daniel Axtens 2020-06-03 279 extern void *__underlying_memcpy(void *p, const void *q, __kernel_size_t size) __RENAME(memcpy); 47227d27e2fcb0 Daniel Axtens 2020-06-03 280 extern void *__underlying_memmove(void *p, const void *q, __kernel_size_t size) __RENAME(memmove); 47227d27e2fcb0 Daniel Axtens 2020-06-03 281 extern void *__underlying_memset(void *p, int c, __kernel_size_t size) __RENAME(memset); 47227d27e2fcb0 Daniel Axtens 2020-06-03 282 extern char *__underlying_strcat(char *p, const char *q) __RENAME(strcat); 47227d27e2fcb0 Daniel Axtens 2020-06-03 283 extern char *__underlying_strcpy(char *p, const char *q) __RENAME(strcpy); 47227d27e2fcb0 Daniel Axtens 2020-06-03 284 extern __kernel_size_t __underlying_strlen(const char *p) __RENAME(strlen); 47227d27e2fcb0 Daniel Axtens 2020-06-03 285 extern char *__underlying_strncat(char *p, const char *q, __kernel_size_t count) __RENAME(strncat); 47227d27e2fcb0 Daniel Axtens 2020-06-03 286 extern char *__underlying_strncpy(char *p, const char *q, __kernel_size_t size) __RENAME(strncpy); 47227d27e2fcb0 Daniel Axtens 2020-06-03 287 #else 47227d27e2fcb0 Daniel Axtens 2020-06-03 288 #define __underlying_memchr __builtin_memchr 47227d27e2fcb0 Daniel Axtens 2020-06-03 289 #define __underlying_memcmp __builtin_memcmp 47227d27e2fcb0 Daniel Axtens 2020-06-03 290 #define __underlying_memcpy __builtin_memcpy 47227d27e2fcb0 Daniel Axtens 2020-06-03 291 #define __underlying_memmove __builtin_memmove 47227d27e2fcb0 Daniel Axtens 2020-06-03 292 #define __underlying_memset __builtin_memset 47227d27e2fcb0 Daniel Axtens 2020-06-03 293 #define __underlying_strcat __builtin_strcat 47227d27e2fcb0 Daniel Axtens 2020-06-03 294 #define __underlying_strcpy __builtin_strcpy 47227d27e2fcb0 Daniel Axtens 2020-06-03 295 #define __underlying_strlen __builtin_strlen 47227d27e2fcb0 Daniel Axtens 2020-06-03 296 #define __underlying_strncat __builtin_strncat 47227d27e2fcb0 Daniel Axtens 2020-06-03 @297 #define __underlying_strncpy __builtin_strncpy 47227d27e2fcb0 Daniel Axtens 2020-06-03 298 #endif 47227d27e2fcb0 Daniel Axtens 2020-06-03 299 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

Re: [PATCH 2/2] ASoC: qcom: add apq8039 sound card support
by kernel test robot 20 Jun '20

20 Jun '20

Hi Pantelis, I love your patch! Perhaps something to improve: [auto build test WARNING on asoc/for-next] [also build test WARNING on sound/for-next v5.8-rc1 next-20200618] [If your patch is applied to the wrong git tree, kindly drop us a note. And when submitting patch, we suggest to use as documented in https://git-scm.com/docs/git-format-patch] url: https://github.com/0day-ci/linux/commits/Pantelis-Antoniou/ASoC-qcom-add-ap… base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next config: parisc-allyesconfig (attached as .config) compiler: hppa-linux-gcc (GCC) 9.3.0 reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross ARCH=parisc If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>, old ones prefixed by <<): sound/soc/qcom/apq8039.c: In function 'snd_soc_card_ctl_getset.constprop': >> sound/soc/qcom/apq8039.c:176:5: warning: 'strncpy' destination unchanged after copying no bytes [-Wstringop-truncation] 176 | strncpy(orig, uinfo->value.enumerated.name, | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 177 | origsz); | ~~~~~~~ vim +/strncpy +176 sound/soc/qcom/apq8039.c 67 68 static int 69 snd_soc_card_ctl_getset(struct snd_soc_card *card, 70 const char *name, 71 char *orig, size_t origsz, const char *value) 72 { 73 struct snd_kcontrol *ctl; 74 struct snd_ctl_elem_info *uinfo = NULL; 75 struct snd_ctl_elem_value *uctl = NULL; 76 unsigned int i, count; 77 int ret; 78 79 /* get the control */ 80 ctl = snd_soc_card_ctl_find(card, name, NULL); 81 if (!ctl) 82 return -ENOENT; 83 84 /* allocate info and value */ 85 uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); 86 uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); 87 if (!uinfo || !uctl) { 88 ret = -ENOMEM; 89 goto out_free; 90 } 91 92 ret = ctl->info(ctl, uinfo); 93 if (ret) 94 goto out_free; 95 if (uinfo->count != 1) { 96 ret = -ENOTSUPP; 97 goto out_free; 98 } 99 100 ret = ctl->get(ctl, uctl); 101 if (ret) 102 goto out_free; 103 104 ret = 0; 105 106 switch (uinfo->type) { 107 108 case SNDRV_CTL_ELEM_TYPE_NONE: 109 break; 110 111 case SNDRV_CTL_ELEM_TYPE_BOOLEAN: 112 if (orig && origsz > 0) 113 snprintf(orig, origsz, "%s", 114 uctl->value.integer.value[0] ? 115 "true" : "false"); 116 117 if (value) { 118 bool bval; 119 120 ret = kstrtobool(value, &bval); 121 if (ret) 122 goto out_free; 123 uctl->value.integer.value[0] = !!bval; 124 125 } 126 break; 127 128 case SNDRV_CTL_ELEM_TYPE_INTEGER: 129 if (orig && origsz > 0) 130 snprintf(orig, origsz, "%ld", 131 uctl->value.integer.value[0]); 132 133 if (value) { 134 ret = kstrtol(value, 10, 135 &uctl->value.integer.value[0]); 136 if (ret) 137 goto out_free; 138 } 139 break; 140 141 case SNDRV_CTL_ELEM_TYPE_ENUMERATED: 142 count = uinfo->value.enumerated.items; 143 144 if (value) { /* set */ 145 for (i = 0; i < count; i++) { 146 uinfo->value.enumerated.item = i; 147 ret = ctl->info(ctl, uinfo); 148 if (ret) 149 goto out_free; 150 151 if (!strcmp(value, 152 uinfo->value.enumerated.name)) 153 break; 154 } 155 156 /* setting without finding the enum */ 157 if (i >= count) { 158 ret = -EINVAL; 159 goto out_free; 160 } 161 uctl->value.enumerated.item[0] = i; 162 } else { /* get */ 163 uinfo->value.enumerated.item = 164 uctl->value.enumerated.item[0]; 165 ret = ctl->info(ctl, uinfo); 166 if (ret) 167 goto out_free; 168 169 if (orig && origsz) { 170 if (origsz < 171 strlen(uinfo->value.enumerated.name) 172 + 1) { 173 ret = -ENOSPC; 174 goto out_free; 175 } > 176 strncpy(orig, uinfo->value.enumerated.name, 177 origsz); 178 } 179 } 180 break; 181 182 case SNDRV_CTL_ELEM_TYPE_INTEGER64: 183 if (orig && origsz > 0) 184 snprintf(orig, origsz, "%lld", 185 uctl->value.integer64.value[0]); 186 187 if (value) { 188 ret = kstrtoll(value, 10, 189 &uctl->value.integer64.value[0]); 190 if (ret) 191 goto out_free; 192 } 193 break; 194 195 case SNDRV_CTL_ELEM_TYPE_BYTES: 196 case SNDRV_CTL_ELEM_TYPE_IEC958: 197 ret = -ENOTSUPP; 198 goto out_free; 199 default: 200 ret = -EINVAL; 201 goto out_free; 202 } 203 204 if (value) { 205 ret = ctl->put(ctl, uctl); 206 if (ret < 0) 207 goto out_free; 208 209 /* if it changed, report change to user space */ 210 if (ret > 0) 211 snd_ctl_notify(card->snd_card, 212 SNDRV_CTL_EVENT_MASK_VALUE, 213 &uctl->id); 214 } 215 216 out_free: 217 kfree(uctl); 218 kfree(uinfo); 219 220 if (ret < 0) 221 dev_err(card->dev, "%s: %s operation failed for \"%s\"", 222 __func__, value ? "set" : "get", name); 223 224 return ret; 225 } 226 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[PATCH] ASoC: Intel: haswell: Power transition refactor
by Cezary Rojewski 20 Jun '20

20 Jun '20

Update D0 <-> D3 sequence to correctly transition hardware and DSP core from and to D3. On top of that, set SHIM registers to their recommended defaults during D0 and D3 proceduces as HW does not reset registers for us. Connected to: [alsa-devel][BUG] bdw-rt5650 DSP boot timeout https://mailman.alsa-project.org/pipermail/alsa-devel/2019-July/153098.html Github issue ticket reference: https://github.com/thesofproject/linux/pull/1842 Tested on: - BDW-Y RVP with rt286 - SAMUS with rt5677 Proposed solution (both in July 2019 and on github): 'Revert "ASoC: Intel: Work around to fix HW d3 potential crash issue"' is NAKed as it only covers the problem up and actually brings back the undefined behavior: some registers (e.g.: APLLSE) are describing LPT offsets rather than WPT ones. In consequence, during power-transitions driver issues incorrect writes and leaves the regs of interest alone. Existing patch - the non-revert - does not resolve the HW D3 issue at all as it ignores the recommended sequence and does not initialize hardware registers as expected. And thus, leaving things as are is also unacceptable. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-dsp.c | 185 ++++++++++++---------- 1 file changed, 104 insertions(+), 81 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c index 88c3f63bded9..de80e19454c1 100644 --- a/sound/soc/intel/haswell/sst-haswell-dsp.c +++ b/sound/soc/intel/haswell/sst-haswell-dsp.c @@ -243,45 +243,92 @@ static irqreturn_t hsw_irq(int irq, void *context) return ret; } +#define CSR_DEFAULT_VALUE 0x8480040E +#define ISC_DEFAULT_VALUE 0x0 +#define ISD_DEFAULT_VALUE 0x0 +#define IMC_DEFAULT_VALUE 0x7FFF0003 +#define IMD_DEFAULT_VALUE 0x7FFF0003 +#define IPCC_DEFAULT_VALUE 0x0 +#define IPCD_DEFAULT_VALUE 0x0 +#define CLKCTL_DEFAULT_VALUE 0x7FF +#define CSR2_DEFAULT_VALUE 0x0 +#define LTR_CTRL_DEFAULT_VALUE 0x0 +#define HMD_CTRL_DEFAULT_VALUE 0x0 + +static void hsw_set_shim_defaults(struct sst_dsp *sst) +{ + sst_dsp_shim_write_unlocked(sst, SST_CSR, CSR_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_ISRX, ISC_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_ISRD, ISD_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_IMRX, IMC_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_IMRD, IMD_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_IPCX, IPCC_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_IPCD, IPCD_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_CLKCTL, CLKCTL_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_CSR2, CSR2_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_LTRC, LTR_CTRL_DEFAULT_VALUE); + sst_dsp_shim_write_unlocked(sst, SST_HMDC, HMD_CTRL_DEFAULT_VALUE); +} + +/* all clock-gating minus DCLCGE and DTCGE */ +#define SST_VDRTCL2_CG_OTHER 0xB7D + static void hsw_set_dsp_D3(struct sst_dsp *sst) { - u32 val; u32 reg; - /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + /* disable clock core gating */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); + reg &= ~(SST_VDRTCL2_DCLCGE); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* enable power gating and switch off DRAM & IRAM blocks */ - val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - val |= SST_VDRTCL0_DSRAMPGE_MASK | - SST_VDRTCL0_ISRAMPGE_MASK; - val &= ~(SST_VDRTCL0_D3PGD | SST_VDRTCL0_D3SRAMPGD); - writel(val, sst->addr.pci_cfg + SST_VDRTCTL0); + /* stall, reset and set 24MHz XOSC */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST, + SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST); - /* switch off audio PLL */ - val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - val |= SST_VDRTCL2_APLLSE_MASK; - writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); + /* DRAM power gating all */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_ISRAMPGE_MASK | + SST_VDRTCL0_DSRAMPGE_MASK; + reg &= ~(SST_VDRTCL0_D3SRAMPGD); + reg |= SST_VDRTCL0_D3PGD; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + udelay(50); - /* disable MCLK(clkctl.smos = 0) */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, - SST_CLKCTL_MASK, 0); + /* PLL shutdown enable */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_APLLSE_MASK; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* Set D3 state, delay 50 us */ - val = readl(sst->addr.pci_cfg + SST_PMCS); - val |= SST_PMCS_PS_MASK; - writel(val, sst->addr.pci_cfg + SST_PMCS); - udelay(50); + /* disable MCLK */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, + SST_CLKCTL_MASK, 0); - /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + /* switch clock gating */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_CG_OTHER; + reg &= ~(SST_VDRTCL2_DTCGE); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + /* enable DTCGE separatelly */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; + reg |= SST_VDRTCL2_DTCGE; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + /* set shim defaults */ + hsw_set_shim_defaults(sst); + + /* set D3 */ + reg = readl(sst->addr.pci_cfg + SST_PMCS); + reg |= SST_PMCS_PS_MASK; + writel(reg, sst->addr.pci_cfg + SST_PMCS); udelay(50); + /* enable clock core gating */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_DCLCGE; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + udelay(50); } static void hsw_reset(struct sst_dsp *sst) @@ -299,75 +346,62 @@ static void hsw_reset(struct sst_dsp *sst) SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); } +/* recommended CSR state for power-up */ +#define SST_CSR_D0_MASK (0x18A09C0C | SST_CSR_DCS_MASK) + static int hsw_set_dsp_D0(struct sst_dsp *sst) { - int tries = 10; - u32 reg, fw_dump_bit; + u32 reg; - /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ + /* disable clock core gating */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); + reg &= ~(SST_VDRTCL2_DCLCGE); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* Disable D3PG (VDRTCTL0.D3PGD = 1) */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg |= SST_VDRTCL0_D3PGD; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + /* switch clock gating */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + reg |= SST_VDRTCL2_CG_OTHER; + reg &= ~(SST_VDRTCL2_DTCGE); + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* Set D0 state */ + /* set D0 */ reg = readl(sst->addr.pci_cfg + SST_PMCS); - reg &= ~SST_PMCS_PS_MASK; + reg &= ~(SST_PMCS_PS_MASK); writel(reg, sst->addr.pci_cfg + SST_PMCS); - /* check that ADSP shim is enabled */ - while (tries--) { - reg = readl(sst->addr.pci_cfg + SST_PMCS) & SST_PMCS_PS_MASK; - if (reg == 0) - goto finish; - - msleep(1); - } - - return -ENODEV; - -finish: - /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); + /* DRAM power gating none */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg &= ~(SST_VDRTCL0_ISRAMPGE_MASK | + SST_VDRTCL0_DSRAMPGE_MASK); + reg |= SST_VDRTCL0_D3SRAMPGD; + reg |= SST_VDRTCL0_D3PGD; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); + mdelay(10); - /* stall DSP core, set clk to 192/96Mhz */ - sst_dsp_shim_update_bits_unlocked(sst, - SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK, - SST_CSR_STALL | SST_CSR_DCS(4)); + /* set shim defaults */ + hsw_set_shim_defaults(sst); - /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */ + /* restore MCLK */ sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, - SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, - SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); + SST_CLKCTL_MASK, SST_CLKCTL_MASK); - /* Stall and reset core, set CSR */ - hsw_reset(sst); - - /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ + /* PLL shutdown disable */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; + reg &= ~(SST_VDRTCL2_APLLSE_MASK); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_D0_MASK, SST_CSR_SBCS0 | SST_CSR_SBCS1 | + SST_CSR_STALL | SST_CSR_DCS(4)); udelay(50); - /* switch on audio PLL */ + /* enable clock core gating */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~SST_VDRTCL2_APLLSE_MASK; + reg |= SST_VDRTCL2_DCLCGE; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* set default power gating control, enable power gating control for all blocks. that is, - can't be accessed, please enable each block before accessing. */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; - /* for D0, always enable the block(DSRAM[0]) used for FW dump */ - fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT; - writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0); - + /* clear reset */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_RST, 0); /* disable DMA finish function for SSP0 & SSP1 */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, @@ -384,12 +418,6 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst) sst_dsp_shim_update_bits(sst, SST_IMRD, (SST_IMRD_DONE | SST_IMRD_BUSY | SST_IMRD_SSP0 | SST_IMRD_DMAC), 0x0); - /* clear IPC registers */ - sst_dsp_shim_write(sst, SST_IPCX, 0x0); - sst_dsp_shim_write(sst, SST_IPCD, 0x0); - sst_dsp_shim_write(sst, 0x80, 0x6); - sst_dsp_shim_write(sst, 0xe0, 0x300a); - return 0; } @@ -415,11 +443,6 @@ static void hsw_sleep(struct sst_dsp *sst) { dev_dbg(sst->dev, "HSW_PM dsp runtime suspend\n"); - /* put DSP into reset and stall */ - sst_dsp_shim_update_bits(sst, SST_CSR, - SST_CSR_24MHZ_LPCS | SST_CSR_RST | SST_CSR_STALL, - SST_CSR_RST | SST_CSR_STALL | SST_CSR_24MHZ_LPCS); - hsw_set_dsp_D3(sst); dev_dbg(sst->dev, "HSW_PM dsp runtime suspend exit\n"); } -- 2.17.1

5 11

[PATCH alsa-lib] conf: Add /usr/etc to the default search paths
by Takashi Iwai 19 Jun '20

19 Jun '20

There's been some recent action to move the distribution-default setup from /etc to /usr/etc, and we should follow that, too. This patch adds /usr/etc/alsa.conf.d and /usr/etc/alsa.conf to the default search paths. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/conf/alsa.conf | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/conf/alsa.conf b/src/conf/alsa.conf index 18427ec6c4c6..aea1fe24f73c 100644 --- a/src/conf/alsa.conf +++ b/src/conf/alsa.conf @@ -8,6 +8,8 @@ { func load files [ + "/usr/etc/alsa/conf.d" + "/usr/etc/asound.conf" "/etc/alsa/conf.d" "/etc/asound.conf" "~/.asoundrc" -- 2.16.4

1 0

[PATCH AUTOSEL 4.19 047/172] ASoC: meson: add missing free_irq() in error path
by Sasha Levin 19 Jun '20

19 Jun '20

From: "Pavel Machek (CIP)" <pavel(a)denx.de> [ Upstream commit 3b8a299a58b2afce464ae11324b59dcf0f1d10a7 ] free_irq() is missing in case of error, fix that. Signed-off-by: Pavel Machek (CIP) <pavel(a)denx.de> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200606153103.GA17905@amd Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/axg-fifo.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 0e4f65e654c4..b229c182b7c3 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -209,7 +209,7 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) /* Enable pclk to access registers and clock the fifo ip */ ret = clk_prepare_enable(fifo->pclk); if (ret) - return ret; + goto free_irq; /* Setup status2 so it reports the memory pointer */ regmap_update_bits(fifo->map, FIFO_CTRL1, @@ -229,8 +229,14 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) /* Take memory arbitror out of reset */ ret = reset_control_deassert(fifo->arb); if (ret) - clk_disable_unprepare(fifo->pclk); + goto free_clk; + + return 0; +free_clk: + clk_disable_unprepare(fifo->pclk); +free_irq: + free_irq(fifo->irq, ss); return ret; } -- 2.25.1

2 1

[PATCH v2] ASoC: fsl_spdif: Add pm runtime function
by Shengjiu Wang 19 Jun '20

19 Jun '20

Add pm runtime support and move clock handling there. Close the clocks at suspend to reduce the power consumption. fsl_spdif_suspend is replaced by pm_runtime_force_suspend. fsl_spdif_resume is replaced by pm_runtime_force_resume. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- changes in v2 - remove goto in startup() - remove goto disable_spba_clk - Add Acked-by: Nicolin Chen sound/soc/fsl/fsl_spdif.c | 117 ++++++++++++++++++++++---------------- 1 file changed, 67 insertions(+), 50 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 5bc0e4729341..5b2689ae63d4 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -16,6 +16,7 @@ #include <linux/of_device.h> #include <linux/of_irq.h> #include <linux/regmap.h> +#include <linux/pm_runtime.h> #include <sound/asoundef.h> #include <sound/dmaengine_pcm.h> @@ -495,29 +496,14 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, struct platform_device *pdev = spdif_priv->pdev; struct regmap *regmap = spdif_priv->regmap; u32 scr, mask; - int i; int ret; /* Reset module and interrupts only for first initialization */ if (!snd_soc_dai_active(cpu_dai)) { - ret = clk_prepare_enable(spdif_priv->coreclk); - if (ret) { - dev_err(&pdev->dev, "failed to enable core clock\n"); - return ret; - } - - if (!IS_ERR(spdif_priv->spbaclk)) { - ret = clk_prepare_enable(spdif_priv->spbaclk); - if (ret) { - dev_err(&pdev->dev, "failed to enable spba clock\n"); - goto err_spbaclk; - } - } - ret = spdif_softreset(spdif_priv); if (ret) { dev_err(&pdev->dev, "failed to soft reset\n"); - goto err; + return ret; } /* Disable all the interrupts */ @@ -531,18 +517,10 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK | SCR_TXSEL_MASK | SCR_USRC_SEL_MASK | SCR_TXFIFO_FSEL_MASK; - for (i = 0; i < SPDIF_TXRATE_MAX; i++) { - ret = clk_prepare_enable(spdif_priv->txclk[i]); - if (ret) - goto disable_txclk; - } } else { scr = SCR_RXFIFO_FSEL_IF8 | SCR_RXFIFO_AUTOSYNC; mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK| SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK; - ret = clk_prepare_enable(spdif_priv->rxclk); - if (ret) - goto err; } regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); @@ -550,17 +528,6 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, 0); return 0; - -disable_txclk: - for (i--; i >= 0; i--) - clk_disable_unprepare(spdif_priv->txclk[i]); -err: - if (!IS_ERR(spdif_priv->spbaclk)) - clk_disable_unprepare(spdif_priv->spbaclk); -err_spbaclk: - clk_disable_unprepare(spdif_priv->coreclk); - - return ret; } static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, @@ -569,20 +536,17 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct regmap *regmap = spdif_priv->regmap; - u32 scr, mask, i; + u32 scr, mask; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { scr = 0; mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK | SCR_TXSEL_MASK | SCR_USRC_SEL_MASK | SCR_TXFIFO_FSEL_MASK; - for (i = 0; i < SPDIF_TXRATE_MAX; i++) - clk_disable_unprepare(spdif_priv->txclk[i]); } else { scr = SCR_RXFIFO_OFF | SCR_RXFIFO_CTL_ZERO; mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK| SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK; - clk_disable_unprepare(spdif_priv->rxclk); } regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); @@ -591,9 +555,6 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, spdif_intr_status_clear(spdif_priv); regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, SCR_LOW_POWER); - if (!IS_ERR(spdif_priv->spbaclk)) - clk_disable_unprepare(spdif_priv->spbaclk); - clk_disable_unprepare(spdif_priv->coreclk); } } @@ -1350,6 +1311,8 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Register with ASoC */ dev_set_drvdata(&pdev->dev, spdif_priv); + pm_runtime_enable(&pdev->dev); + regcache_cache_only(spdif_priv->regmap, true); ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component, &spdif_priv->cpu_dai_drv, 1); @@ -1365,36 +1328,90 @@ static int fsl_spdif_probe(struct platform_device *pdev) return ret; } -#ifdef CONFIG_PM_SLEEP -static int fsl_spdif_suspend(struct device *dev) +#ifdef CONFIG_PM +static int fsl_spdif_runtime_suspend(struct device *dev) { struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev); + int i; regmap_read(spdif_priv->regmap, REG_SPDIF_SRPC, &spdif_priv->regcache_srpc); - regcache_cache_only(spdif_priv->regmap, true); - regcache_mark_dirty(spdif_priv->regmap); + + clk_disable_unprepare(spdif_priv->rxclk); + + for (i = 0; i < SPDIF_TXRATE_MAX; i++) + clk_disable_unprepare(spdif_priv->txclk[i]); + + if (!IS_ERR(spdif_priv->spbaclk)) + clk_disable_unprepare(spdif_priv->spbaclk); + clk_disable_unprepare(spdif_priv->coreclk); return 0; } -static int fsl_spdif_resume(struct device *dev) +static int fsl_spdif_runtime_resume(struct device *dev) { struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev); + int ret; + int i; + + ret = clk_prepare_enable(spdif_priv->coreclk); + if (ret) { + dev_err(dev, "failed to enable core clock\n"); + return ret; + } + + if (!IS_ERR(spdif_priv->spbaclk)) { + ret = clk_prepare_enable(spdif_priv->spbaclk); + if (ret) { + dev_err(dev, "failed to enable spba clock\n"); + goto disable_core_clk; + } + } + + for (i = 0; i < SPDIF_TXRATE_MAX; i++) { + ret = clk_prepare_enable(spdif_priv->txclk[i]); + if (ret) + goto disable_tx_clk; + } + + ret = clk_prepare_enable(spdif_priv->rxclk); + if (ret) + goto disable_tx_clk; regcache_cache_only(spdif_priv->regmap, false); + regcache_mark_dirty(spdif_priv->regmap); regmap_update_bits(spdif_priv->regmap, REG_SPDIF_SRPC, SRPC_CLKSRC_SEL_MASK | SRPC_GAINSEL_MASK, spdif_priv->regcache_srpc); - return regcache_sync(spdif_priv->regmap); + ret = regcache_sync(spdif_priv->regmap); + if (ret) + goto disable_rx_clk; + + return 0; + +disable_rx_clk: + clk_disable_unprepare(spdif_priv->rxclk); +disable_tx_clk: + for (i--; i >= 0; i--) + clk_disable_unprepare(spdif_priv->txclk[i]); + if (!IS_ERR(spdif_priv->spbaclk)) + clk_disable_unprepare(spdif_priv->spbaclk); +disable_core_clk: + clk_disable_unprepare(spdif_priv->coreclk); + + return ret; } -#endif /* CONFIG_PM_SLEEP */ +#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_spdif_pm = { - SET_SYSTEM_SLEEP_PM_OPS(fsl_spdif_suspend, fsl_spdif_resume) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) + SET_RUNTIME_PM_OPS(fsl_spdif_runtime_suspend, fsl_spdif_runtime_resume, + NULL) }; static const struct of_device_id fsl_spdif_dt_ids[] = { -- 2.21.0

2 1

[PATCH v3] ASoC: qcom: Kconfig: Tweak dependencies on SND_SOC_SDM845
by John Stultz 19 Jun '20

19 Jun '20

CROS_EC isn't strictly required for audio to work on other SDM845 platforms (like the Dragonboard 845c). So lets remove the dependency and select the related CROS_EC options via imply. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Rohit kumar <rohitkr(a)codeaurora.org> Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: John Stultz <john.stultz(a)linaro.org> --- v2: Switch to using imply as suggested by Srinivas --- sound/soc/qcom/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index f51b28d1b94d..0ea4cde9f4f0 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -99,12 +99,12 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR && CROS_EC && I2C && SOUNDWIRE + depends on QCOM_APR && I2C && SOUNDWIRE select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 select SND_SOC_MAX98927 - select SND_SOC_CROS_EC_CODEC + imply SND_SOC_CROS_EC_CODEC help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.17.1

3 2

ALSA 1.2.3 release
by Jaroslav Kysela 19 Jun '20

19 Jun '20

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly from the FTP server ftp://ftp.alsa-project.org/pub . Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.2_v1.2.3 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

1 1

alsa-gobject: supplement to use guint8 for the numerical ID of client/queue
by GitHub pull_request - edited 19 Jun '20

19 Jun '20

alsa-project/alsa-gobject pull request #42 was edited from takaswie: Some APIs are missing in #41. This patchset has supplements for them. Request URL : https://github.com/alsa-project/alsa-gobject/pull/42 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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alsa-gobject:
by GitHub pull_request - opened 19 Jun '20

19 Jun '20

alsa-project/alsa-gobject pull request #42 was opened from takaswie: Some APIs are missing in #41. This patchset has supplements for them. Request URL : https://github.com/alsa-project/alsa-gobject/pull/42 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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alsa-gobject: seq: use guint8 for the numerical ID of client/port/queue
by GitHub pull_request - opened 19 Jun '20

19 Jun '20

alsa-project/alsa-gobject pull request #41 was opened from takaswie: In ALSA Sequencer core, the number of client/port/queue is defined as below per each: - SNDRV_SEQ_MAX_CLIENTS (=192) - SNDRV_SEQ_MAX_PORTS (=254) - SNDRV_SEQ_MAX_QUEUES (=32) Following to it, in UAPI of Linux sound subsystem, several structures have `unsigned char` type of members: - struct snd_seq_addr.client - struct snd_seq_addr.port - struct snd_seq_event.queue On the other hand, within the UAPI, 'int' type is also used for the numerical ID of client/port/queue. Following to the 'int' type, current API of ALSASeq includes API with 'gint' or 'guint' type of argument. However, it's better to use 'guint8' with consideration about the range of value. This patchset declares the arguments to have 'guint8' type. Request URL : https://github.com/alsa-project/alsa-gobject/pull/41 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/41.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH] ASoC: fsl_spdif: Add pm runtime function
by Shengjiu Wang 19 Jun '20

19 Jun '20

Add pm runtime support and move clock handling there. Close the clocks at suspend to reduce the power consumption. fsl_spdif_suspend is replaced by pm_runtime_force_suspend. fsl_spdif_resume is replaced by pm_runtime_force_resume. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 113 ++++++++++++++++++++++---------------- 1 file changed, 67 insertions(+), 46 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 5bc0e4729341..46719fd2f1ec 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -16,6 +16,7 @@ #include <linux/of_device.h> #include <linux/of_irq.h> #include <linux/regmap.h> +#include <linux/pm_runtime.h> #include <sound/asoundef.h> #include <sound/dmaengine_pcm.h> @@ -495,25 +496,10 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, struct platform_device *pdev = spdif_priv->pdev; struct regmap *regmap = spdif_priv->regmap; u32 scr, mask; - int i; int ret; /* Reset module and interrupts only for first initialization */ if (!snd_soc_dai_active(cpu_dai)) { - ret = clk_prepare_enable(spdif_priv->coreclk); - if (ret) { - dev_err(&pdev->dev, "failed to enable core clock\n"); - return ret; - } - - if (!IS_ERR(spdif_priv->spbaclk)) { - ret = clk_prepare_enable(spdif_priv->spbaclk); - if (ret) { - dev_err(&pdev->dev, "failed to enable spba clock\n"); - goto err_spbaclk; - } - } - ret = spdif_softreset(spdif_priv); if (ret) { dev_err(&pdev->dev, "failed to soft reset\n"); @@ -531,18 +517,10 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK | SCR_TXSEL_MASK | SCR_USRC_SEL_MASK | SCR_TXFIFO_FSEL_MASK; - for (i = 0; i < SPDIF_TXRATE_MAX; i++) { - ret = clk_prepare_enable(spdif_priv->txclk[i]); - if (ret) - goto disable_txclk; - } } else { scr = SCR_RXFIFO_FSEL_IF8 | SCR_RXFIFO_AUTOSYNC; mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK| SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK; - ret = clk_prepare_enable(spdif_priv->rxclk); - if (ret) - goto err; } regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); @@ -551,15 +529,7 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, return 0; -disable_txclk: - for (i--; i >= 0; i--) - clk_disable_unprepare(spdif_priv->txclk[i]); err: - if (!IS_ERR(spdif_priv->spbaclk)) - clk_disable_unprepare(spdif_priv->spbaclk); -err_spbaclk: - clk_disable_unprepare(spdif_priv->coreclk); - return ret; } @@ -569,20 +539,17 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); struct regmap *regmap = spdif_priv->regmap; - u32 scr, mask, i; + u32 scr, mask; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { scr = 0; mask = SCR_TXFIFO_AUTOSYNC_MASK | SCR_TXFIFO_CTRL_MASK | SCR_TXSEL_MASK | SCR_USRC_SEL_MASK | SCR_TXFIFO_FSEL_MASK; - for (i = 0; i < SPDIF_TXRATE_MAX; i++) - clk_disable_unprepare(spdif_priv->txclk[i]); } else { scr = SCR_RXFIFO_OFF | SCR_RXFIFO_CTL_ZERO; mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK| SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK; - clk_disable_unprepare(spdif_priv->rxclk); } regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); @@ -591,9 +558,6 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, spdif_intr_status_clear(spdif_priv); regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, SCR_LOW_POWER); - if (!IS_ERR(spdif_priv->spbaclk)) - clk_disable_unprepare(spdif_priv->spbaclk); - clk_disable_unprepare(spdif_priv->coreclk); } } @@ -1350,6 +1314,8 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Register with ASoC */ dev_set_drvdata(&pdev->dev, spdif_priv); + pm_runtime_enable(&pdev->dev); + regcache_cache_only(spdif_priv->regmap, true); ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component, &spdif_priv->cpu_dai_drv, 1); @@ -1365,36 +1331,91 @@ static int fsl_spdif_probe(struct platform_device *pdev) return ret; } -#ifdef CONFIG_PM_SLEEP -static int fsl_spdif_suspend(struct device *dev) +#ifdef CONFIG_PM +static int fsl_spdif_runtime_suspend(struct device *dev) { struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev); + int i; regmap_read(spdif_priv->regmap, REG_SPDIF_SRPC, &spdif_priv->regcache_srpc); - regcache_cache_only(spdif_priv->regmap, true); - regcache_mark_dirty(spdif_priv->regmap); + + clk_disable_unprepare(spdif_priv->rxclk); + + for (i = 0; i < SPDIF_TXRATE_MAX; i++) + clk_disable_unprepare(spdif_priv->txclk[i]); + + if (!IS_ERR(spdif_priv->spbaclk)) + clk_disable_unprepare(spdif_priv->spbaclk); + clk_disable_unprepare(spdif_priv->coreclk); return 0; } -static int fsl_spdif_resume(struct device *dev) +static int fsl_spdif_runtime_resume(struct device *dev) { struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev); + int ret; + int i; + + ret = clk_prepare_enable(spdif_priv->coreclk); + if (ret) { + dev_err(dev, "failed to enable core clock\n"); + return ret; + } + + if (!IS_ERR(spdif_priv->spbaclk)) { + ret = clk_prepare_enable(spdif_priv->spbaclk); + if (ret) { + dev_err(dev, "failed to enable spba clock\n"); + goto disable_core_clk; + } + } + + for (i = 0; i < SPDIF_TXRATE_MAX; i++) { + ret = clk_prepare_enable(spdif_priv->txclk[i]); + if (ret) + goto disable_spba_clk; + } + + ret = clk_prepare_enable(spdif_priv->rxclk); + if (ret) + goto disable_tx_clk; regcache_cache_only(spdif_priv->regmap, false); + regcache_mark_dirty(spdif_priv->regmap); regmap_update_bits(spdif_priv->regmap, REG_SPDIF_SRPC, SRPC_CLKSRC_SEL_MASK | SRPC_GAINSEL_MASK, spdif_priv->regcache_srpc); - return regcache_sync(spdif_priv->regmap); + ret = regcache_sync(spdif_priv->regmap); + if (ret) + goto disable_rx_clk; + + return 0; + +disable_rx_clk: + clk_disable_unprepare(spdif_priv->rxclk); +disable_tx_clk: +disable_spba_clk: + for (i--; i >= 0; i--) + clk_disable_unprepare(spdif_priv->txclk[i]); + if (!IS_ERR(spdif_priv->spbaclk)) + clk_disable_unprepare(spdif_priv->spbaclk); +disable_core_clk: + clk_disable_unprepare(spdif_priv->coreclk); + + return ret; } -#endif /* CONFIG_PM_SLEEP */ +#endif static const struct dev_pm_ops fsl_spdif_pm = { - SET_SYSTEM_SLEEP_PM_OPS(fsl_spdif_suspend, fsl_spdif_resume) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) + SET_RUNTIME_PM_OPS(fsl_spdif_runtime_suspend, fsl_spdif_runtime_resume, + NULL) }; static const struct of_device_id fsl_spdif_dt_ids[] = { -- 2.21.0

3 2

[PATCH] ASoC: dt-bindings: ak4642: switch to yaml base Documentation
by Kuninori Morimoto 19 Jun '20

19 Jun '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/ak4642.txt | 37 ------------ .../devicetree/bindings/sound/ak4642.yaml | 57 +++++++++++++++++++ 2 files changed, 57 insertions(+), 37 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ak4642.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4642.yaml diff --git a/Documentation/devicetree/bindings/sound/ak4642.txt b/Documentation/devicetree/bindings/sound/ak4642.txt deleted file mode 100644 index 58e48ee97175..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4642.txt +++ /dev/null @@ -1,37 +0,0 @@ -AK4642 I2C transmitter - -This device supports I2C mode only. - -Required properties: - - - compatible : "asahi-kasei,ak4642" or "asahi-kasei,ak4643" or "asahi-kasei,ak4648" - - reg : The chip select number on the I2C bus - -Optional properties: - - - #clock-cells : common clock binding; shall be set to 0 - - clocks : common clock binding; MCKI clock - - clock-frequency : common clock binding; frequency of MCKO - - clock-output-names : common clock binding; MCKO clock name - -Example 1: - -&i2c { - ak4648: ak4648@12 { - compatible = "asahi-kasei,ak4642"; - reg = <0x12>; - }; -}; - -Example 2: - -&i2c { - ak4643: codec@12 { - compatible = "asahi-kasei,ak4643"; - reg = <0x12>; - #clock-cells = <0>; - clocks = <&audio_clock>; - clock-frequency = <12288000>; - clock-output-names = "ak4643_mcko"; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ak4642.yaml b/Documentation/devicetree/bindings/sound/ak4642.yaml new file mode 100644 index 000000000000..c8ffc86fd2d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4642.yaml @@ -0,0 +1,57 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4642.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4642 I2C transmitter Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + compatible: + enum: + - asahi-kasei,ak4642 + - asahi-kasei,ak4643 + - asahi-kasei,ak4648 + + reg: + maxItems: 1 + + "#clock-cells": + const: 0 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-frequency: + description: frequency of MCKO + + clock-output-names: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4643: codec@12 { + compatible = "asahi-kasei,ak4643"; + #sound-dai-cells = <0>; + reg = <0x12>; + #clock-cells = <0>; + clocks = <&audio_clock>; + clock-frequency = <12288000>; + clock-output-names = "ak4643_mcko"; + }; + }; -- 2.25.1

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[PATCH] ASoC: dt-bindings: ak4613: switch to yaml base Documentation
by Kuninori Morimoto 19 Jun '20

19 Jun '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - use patternProperties .../devicetree/bindings/sound/ak4613.txt | 27 ---------- .../devicetree/bindings/sound/ak4613.yaml | 49 +++++++++++++++++++ 2 files changed, 49 insertions(+), 27 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4613.yaml diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt deleted file mode 100644 index 49a2e74fd9cb..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4613.txt +++ /dev/null @@ -1,27 +0,0 @@ -AK4613 I2C transmitter - -This device supports I2C mode only. - -Required properties: - -- compatible : "asahi-kasei,ak4613" -- reg : The chip select number on the I2C bus - -Optional properties: -- asahi-kasei,in1-single-end : Boolean. Indicate input / output pins are single-ended. -- asahi-kasei,in2-single-end rather than differential. -- asahi-kasei,out1-single-end -- asahi-kasei,out2-single-end -- asahi-kasei,out3-single-end -- asahi-kasei,out4-single-end -- asahi-kasei,out5-single-end -- asahi-kasei,out6-single-end - -Example: - -&i2c { - ak4613: ak4613@10 { - compatible = "asahi-kasei,ak4613"; - reg = <0x10>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ak4613.yaml b/Documentation/devicetree/bindings/sound/ak4613.yaml new file mode 100644 index 000000000000..eb915377bc5d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4613.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4613 I2C transmitter Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + compatible: + const: asahi-kasei,ak4613 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +patternProperties: + "^asahi-kasei,in[1-2]-single-end$": + description: Input Pin 1 - 2. + $ref: /schemas/types.yaml#/definitions/flag + + "^asahi-kasei,out[1-6]-single-end$": + description: Output Pin 1 - 6. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4613: ak4613@10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; + }; -- 2.25.1

1 0

[PATCH] ASoC: dt-bindings: renesas, fsi: use patternProperties for FSI-A/B
by Kuninori Morimoto 19 Jun '20

19 Jun '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> FSI has FSI-A and FSI-B, and has fsia-xxx/fsib-xxx properties. This patch uses patternProperties, and reduce verbose settings. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../bindings/sound/renesas,fsi.yaml | 19 ++++--------------- 1 file changed, 4 insertions(+), 15 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml index 8a4406be387a..0dd3f7361399 100644 --- a/Documentation/devicetree/bindings/sound/renesas,fsi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,fsi.yaml @@ -43,30 +43,19 @@ properties: '#sound-dai-cells': const: 1 - fsia,spdif-connection: +patternProperties: + "^fsi(a|b),spdif-connection$": $ref: /schemas/types.yaml#/definitions/flag description: FSI is connected by S/PDIF - fsia,stream-mode-support: + "^fsi(a|b),stream-mode-support$": $ref: /schemas/types.yaml#/definitions/flag description: FSI supports 16bit stream mode - fsia,use-internal-clock: + "^fsi(a|b),use-internal-clock$": $ref: /schemas/types.yaml#/definitions/flag description: FSI uses internal clock when master mode - fsib,spdif-connection: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,stream-mode-support: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - - fsib,use-internal-clock: - $ref: /schemas/types.yaml#/definitions/flag - description: same as fsia - required: - compatible - reg -- 2.25.1

1 0

[PATCH] ASoC: codecs: Added MAX98373 Soundwire Driver
by Pierre-Louis Bossart 19 Jun '20

19 Jun '20

From: Ryan Lee <ryans.lee(a)maximintegrated.com> Add SoundWire support for this amplifier, tested on Tigerlake platform. Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98373-sdw.c | 950 ++++++++++++++++++++++++++++++++ sound/soc/codecs/max98373-sdw.h | 73 +++ sound/soc/codecs/max98373.c | 15 + sound/soc/codecs/max98373.h | 8 + 6 files changed, 1062 insertions(+) create mode 100644 sound/soc/codecs/max98373-sdw.c create mode 100644 sound/soc/codecs/max98373-sdw.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 986a6308818b..24073730fb8c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -116,6 +116,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_MAX98926 imply SND_SOC_MAX98927 imply SND_SOC_MAX98373 + imply SND_SOC_MAX98373_SDW imply SND_SOC_MAX98390 imply SND_SOC_MAX9850 imply SND_SOC_MAX9860 @@ -871,6 +872,19 @@ config SND_SOC_MAX98373 tristate "Maxim Integrated MAX98373 Speaker Amplifier" depends on I2C +config SND_SOC_MAX98373_SDW + tristate "Maxim Integrated MAX98373 Speaker Amplifier - SDW" + depends on SOUNDWIRE + select SND_SOC_MAX98373 + select REGMAP_SOUNDWIRE + help + Enable support for Maxim Integrated MAX98373 Soundwire + amplifier. MAX98373 supports either the MIPI SoundWire + compatible interface for audio and control data, or + the PCM interface for audio data and a standard I2C + interface for control data. Select this if MAX98373 is + connected via soundwire. + config SND_SOC_MAX98390 tristate "Maxim Integrated MAX98390 Speaker Amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 47ae3cebb61e..426da2db874b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -115,6 +115,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max98927-objs := max98927.o snd-soc-max98373-objs := max98373.o +snd-soc-max98373-sdw-objs := max98373-sdw.o snd-soc-max98390-objs := max98390.o snd-soc-max9850-objs := max9850.o snd-soc-max9860-objs := max9860.o @@ -418,6 +419,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX98927) += snd-soc-max98927.o obj-$(CONFIG_SND_SOC_MAX98373) += snd-soc-max98373.o +obj-$(CONFIG_SND_SOC_MAX98373_SDW) += snd-soc-max98373-sdw.o obj-$(CONFIG_SND_SOC_MAX98390) += snd-soc-max98390.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c new file mode 100644 index 000000000000..074ca4c500bc --- /dev/null +++ b/sound/soc/codecs/max98373-sdw.c @@ -0,0 +1,950 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright (c) 2020, Maxim Integrated + +#include <linux/acpi.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <linux/of.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include "max98373-sdw.h" + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +static struct reg_default max98373_reg[] = { + {MAX98373_R0040_SCP_INIT_STAT_1, 0x00}, + {MAX98373_R0041_SCP_INIT_MASK_1, 0x00}, + {MAX98373_R0042_SCP_INIT_STAT_2, 0x00}, + {MAX98373_R0044_SCP_CTRL, 0x00}, + {MAX98373_R0045_SCP_SYSTEM_CTRL, 0x00}, + {MAX98373_R0046_SCP_DEV_NUMBER, 0x00}, + {MAX98373_R0050_SCP_DEV_ID_0, 0x21}, + {MAX98373_R0051_SCP_DEV_ID_1, 0x01}, + {MAX98373_R0052_SCP_DEV_ID_2, 0x9F}, + {MAX98373_R0053_SCP_DEV_ID_3, 0x87}, + {MAX98373_R0054_SCP_DEV_ID_4, 0x08}, + {MAX98373_R0055_SCP_DEV_ID_5, 0x00}, + {MAX98373_R0060_SCP_FRAME_CTLR, 0x00}, + {MAX98373_R0070_SCP_FRAME_CTLR, 0x00}, + {MAX98373_R0100_DP1_INIT_STAT, 0x00}, + {MAX98373_R0101_DP1_INIT_MASK, 0x00}, + {MAX98373_R0102_DP1_PORT_CTRL, 0x00}, + {MAX98373_R0103_DP1_BLOCK_CTRL_1, 0x00}, + {MAX98373_R0104_DP1_PREPARE_STATUS, 0x00}, + {MAX98373_R0105_DP1_PREPARE_CTRL, 0x00}, + {MAX98373_R0120_DP1_CHANNEL_EN, 0x00}, + {MAX98373_R0122_DP1_SAMPLE_CTRL1, 0x00}, + {MAX98373_R0123_DP1_SAMPLE_CTRL2, 0x00}, + {MAX98373_R0124_DP1_OFFSET_CTRL1, 0x00}, + {MAX98373_R0125_DP1_OFFSET_CTRL2, 0x00}, + {MAX98373_R0126_DP1_HCTRL, 0x00}, + {MAX98373_R0127_DP1_BLOCK_CTRL3, 0x00}, + {MAX98373_R0130_DP1_CHANNEL_EN, 0x00}, + {MAX98373_R0132_DP1_SAMPLE_CTRL1, 0x00}, + {MAX98373_R0133_DP1_SAMPLE_CTRL2, 0x00}, + {MAX98373_R0134_DP1_OFFSET_CTRL1, 0x00}, + {MAX98373_R0135_DP1_OFFSET_CTRL2, 0x00}, + {MAX98373_R0136_DP1_HCTRL, 0x0136}, + {MAX98373_R0137_DP1_BLOCK_CTRL3, 0x00}, + {MAX98373_R0300_DP3_INIT_STAT, 0x00}, + {MAX98373_R0301_DP3_INIT_MASK, 0x00}, + {MAX98373_R0302_DP3_PORT_CTRL, 0x00}, + {MAX98373_R0303_DP3_BLOCK_CTRL_1, 0x00}, + {MAX98373_R0304_DP3_PREPARE_STATUS, 0x00}, + {MAX98373_R0305_DP3_PREPARE_CTRL, 0x00}, + {MAX98373_R0320_DP3_CHANNEL_EN, 0x00}, + {MAX98373_R0322_DP3_SAMPLE_CTRL1, 0x00}, + {MAX98373_R0323_DP3_SAMPLE_CTRL2, 0x00}, + {MAX98373_R0324_DP3_OFFSET_CTRL1, 0x00}, + {MAX98373_R0325_DP3_OFFSET_CTRL2, 0x00}, + {MAX98373_R0326_DP3_HCTRL, 0x00}, + {MAX98373_R0327_DP3_BLOCK_CTRL3, 0x00}, + {MAX98373_R0330_DP3_CHANNEL_EN, 0x00}, + {MAX98373_R0332_DP3_SAMPLE_CTRL1, 0x00}, + {MAX98373_R0333_DP3_SAMPLE_CTRL2, 0x00}, + {MAX98373_R0334_DP3_OFFSET_CTRL1, 0x00}, + {MAX98373_R0335_DP3_OFFSET_CTRL2, 0x00}, + {MAX98373_R0336_DP3_HCTRL, 0x00}, + {MAX98373_R0337_DP3_BLOCK_CTRL3, 0x00}, + {MAX98373_R2000_SW_RESET, 0x00}, + {MAX98373_R2001_INT_RAW1, 0x00}, + {MAX98373_R2002_INT_RAW2, 0x00}, + {MAX98373_R2003_INT_RAW3, 0x00}, + {MAX98373_R2004_INT_STATE1, 0x00}, + {MAX98373_R2005_INT_STATE2, 0x00}, + {MAX98373_R2006_INT_STATE3, 0x00}, + {MAX98373_R2007_INT_FLAG1, 0x00}, + {MAX98373_R2008_INT_FLAG2, 0x00}, + {MAX98373_R2009_INT_FLAG3, 0x00}, + {MAX98373_R200A_INT_EN1, 0x00}, + {MAX98373_R200B_INT_EN2, 0x00}, + {MAX98373_R200C_INT_EN3, 0x00}, + {MAX98373_R200D_INT_FLAG_CLR1, 0x00}, + {MAX98373_R200E_INT_FLAG_CLR2, 0x00}, + {MAX98373_R200F_INT_FLAG_CLR3, 0x00}, + {MAX98373_R2010_IRQ_CTRL, 0x00}, + {MAX98373_R2014_THERM_WARN_THRESH, 0x10}, + {MAX98373_R2015_THERM_SHDN_THRESH, 0x27}, + {MAX98373_R2016_THERM_HYSTERESIS, 0x01}, + {MAX98373_R2017_THERM_FOLDBACK_SET, 0xC0}, + {MAX98373_R2018_THERM_FOLDBACK_EN, 0x00}, + {MAX98373_R201E_PIN_DRIVE_STRENGTH, 0x55}, + {MAX98373_R2020_PCM_TX_HIZ_EN_1, 0xFE}, + {MAX98373_R2021_PCM_TX_HIZ_EN_2, 0xFF}, + {MAX98373_R2022_PCM_TX_SRC_1, 0x00}, + {MAX98373_R2023_PCM_TX_SRC_2, 0x00}, + {MAX98373_R2024_PCM_DATA_FMT_CFG, 0xC0}, + {MAX98373_R2025_AUDIO_IF_MODE, 0x00}, + {MAX98373_R2026_PCM_CLOCK_RATIO, 0x04}, + {MAX98373_R2027_PCM_SR_SETUP_1, 0x08}, + {MAX98373_R2028_PCM_SR_SETUP_2, 0x88}, + {MAX98373_R2029_PCM_TO_SPK_MONO_MIX_1, 0x00}, + {MAX98373_R202A_PCM_TO_SPK_MONO_MIX_2, 0x00}, + {MAX98373_R202B_PCM_RX_EN, 0x00}, + {MAX98373_R202C_PCM_TX_EN, 0x00}, + {MAX98373_R202E_ICC_RX_CH_EN_1, 0x00}, + {MAX98373_R202F_ICC_RX_CH_EN_2, 0x00}, + {MAX98373_R2030_ICC_TX_HIZ_EN_1, 0xFF}, + {MAX98373_R2031_ICC_TX_HIZ_EN_2, 0xFF}, + {MAX98373_R2032_ICC_LINK_EN_CFG, 0x30}, + {MAX98373_R2034_ICC_TX_CNTL, 0x00}, + {MAX98373_R2035_ICC_TX_EN, 0x00}, + {MAX98373_R2036_SOUNDWIRE_CTRL, 0x05}, + {MAX98373_R203D_AMP_DIG_VOL_CTRL, 0x00}, + {MAX98373_R203E_AMP_PATH_GAIN, 0x08}, + {MAX98373_R203F_AMP_DSP_CFG, 0x02}, + {MAX98373_R2040_TONE_GEN_CFG, 0x00}, + {MAX98373_R2041_AMP_CFG, 0x03}, + {MAX98373_R2042_AMP_EDGE_RATE_CFG, 0x00}, + {MAX98373_R2043_AMP_EN, 0x00}, + {MAX98373_R2046_IV_SENSE_ADC_DSP_CFG, 0x04}, + {MAX98373_R2047_IV_SENSE_ADC_EN, 0x00}, + {MAX98373_R2051_MEAS_ADC_SAMPLING_RATE, 0x00}, + {MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, 0x00}, + {MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, 0x00}, + {MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0x00}, + {MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0x00}, + {MAX98373_R2056_MEAS_ADC_PVDD_CH_EN, 0x00}, + {MAX98373_R2090_BDE_LVL_HOLD, 0x00}, + {MAX98373_R2091_BDE_GAIN_ATK_REL_RATE, 0x00}, + {MAX98373_R2092_BDE_CLIPPER_MODE, 0x00}, + {MAX98373_R2097_BDE_L1_THRESH, 0x00}, + {MAX98373_R2098_BDE_L2_THRESH, 0x00}, + {MAX98373_R2099_BDE_L3_THRESH, 0x00}, + {MAX98373_R209A_BDE_L4_THRESH, 0x00}, + {MAX98373_R209B_BDE_THRESH_HYST, 0x00}, + {MAX98373_R20A8_BDE_L1_CFG_1, 0x00}, + {MAX98373_R20A9_BDE_L1_CFG_2, 0x00}, + {MAX98373_R20AA_BDE_L1_CFG_3, 0x00}, + {MAX98373_R20AB_BDE_L2_CFG_1, 0x00}, + {MAX98373_R20AC_BDE_L2_CFG_2, 0x00}, + {MAX98373_R20AD_BDE_L2_CFG_3, 0x00}, + {MAX98373_R20AE_BDE_L3_CFG_1, 0x00}, + {MAX98373_R20AF_BDE_L3_CFG_2, 0x00}, + {MAX98373_R20B0_BDE_L3_CFG_3, 0x00}, + {MAX98373_R20B1_BDE_L4_CFG_1, 0x00}, + {MAX98373_R20B2_BDE_L4_CFG_2, 0x00}, + {MAX98373_R20B3_BDE_L4_CFG_3, 0x00}, + {MAX98373_R20B4_BDE_INFINITE_HOLD_RELEASE, 0x00}, + {MAX98373_R20B5_BDE_EN, 0x00}, + {MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0x00}, + {MAX98373_R20D1_DHT_CFG, 0x01}, + {MAX98373_R20D2_DHT_ATTACK_CFG, 0x02}, + {MAX98373_R20D3_DHT_RELEASE_CFG, 0x03}, + {MAX98373_R20D4_DHT_EN, 0x00}, + {MAX98373_R20E0_LIMITER_THRESH_CFG, 0x00}, + {MAX98373_R20E1_LIMITER_ATK_REL_RATES, 0x00}, + {MAX98373_R20E2_LIMITER_EN, 0x00}, + {MAX98373_R20FE_DEVICE_AUTO_RESTART_CFG, 0x00}, + {MAX98373_R20FF_GLOBAL_SHDN, 0x00}, + {MAX98373_R21FF_REV_ID, 0x42}, +}; + +static bool max98373_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98373_R21FF_REV_ID: + case MAX98373_R2010_IRQ_CTRL: + /* SoundWire Control Port Registers */ + case MAX98373_R0040_SCP_INIT_STAT_1 ... MAX98373_R0070_SCP_FRAME_CTLR: + /* Soundwire Data Port 1 Registers */ + case MAX98373_R0100_DP1_INIT_STAT ... MAX98373_R0137_DP1_BLOCK_CTRL3: + /* Soundwire Data Port 3 Registers */ + case MAX98373_R0300_DP3_INIT_STAT ... MAX98373_R0337_DP3_BLOCK_CTRL3: + case MAX98373_R2000_SW_RESET ... MAX98373_R200C_INT_EN3: + case MAX98373_R2014_THERM_WARN_THRESH + ... MAX98373_R2018_THERM_FOLDBACK_EN: + case MAX98373_R201E_PIN_DRIVE_STRENGTH + ... MAX98373_R2036_SOUNDWIRE_CTRL: + case MAX98373_R203D_AMP_DIG_VOL_CTRL ... MAX98373_R2043_AMP_EN: + case MAX98373_R2046_IV_SENSE_ADC_DSP_CFG + ... MAX98373_R2047_IV_SENSE_ADC_EN: + case MAX98373_R2051_MEAS_ADC_SAMPLING_RATE + ... MAX98373_R2056_MEAS_ADC_PVDD_CH_EN: + case MAX98373_R2090_BDE_LVL_HOLD ... MAX98373_R2092_BDE_CLIPPER_MODE: + case MAX98373_R2097_BDE_L1_THRESH + ... MAX98373_R209B_BDE_THRESH_HYST: + case MAX98373_R20A8_BDE_L1_CFG_1 ... MAX98373_R20B3_BDE_L4_CFG_3: + case MAX98373_R20B5_BDE_EN ... MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R20D1_DHT_CFG ... MAX98373_R20D4_DHT_EN: + case MAX98373_R20E0_LIMITER_THRESH_CFG ... MAX98373_R20E2_LIMITER_EN: + case MAX98373_R20FE_DEVICE_AUTO_RESTART_CFG + ... MAX98373_R20FF_GLOBAL_SHDN: + return true; + default: + return false; + } +}; + +static bool max98373_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98373_R203E_AMP_PATH_GAIN: + case MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK: + case MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK: + case MAX98373_R20B6_BDE_CUR_STATE_READBACK: + case MAX98373_R21FF_REV_ID: + /* SoundWire Control Port Registers */ + case MAX98373_R0040_SCP_INIT_STAT_1 ... MAX98373_R0070_SCP_FRAME_CTLR: + /* Soundwire Data Port 1 Registers */ + case MAX98373_R0100_DP1_INIT_STAT ... MAX98373_R0137_DP1_BLOCK_CTRL3: + /* Soundwire Data Port 3 Registers */ + case MAX98373_R0300_DP3_INIT_STAT ... MAX98373_R0337_DP3_BLOCK_CTRL3: + case MAX98373_R2000_SW_RESET ... MAX98373_R2009_INT_FLAG3: + return true; + default: + return false; + } +} + +static const struct regmap_config max98373_sdw_regmap = { + .reg_bits = 32, + .val_bits = 8, + .max_register = MAX98373_R21FF_REV_ID, + .reg_defaults = max98373_reg, + .num_reg_defaults = ARRAY_SIZE(max98373_reg), + .readable_reg = max98373_readable_register, + .volatile_reg = max98373_volatile_reg, + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static void max98373_read_config(struct sdw_slave *slave) +{ + int value; + struct device *dev = &slave->dev; + struct max98373_priv *max98373 = dev_get_drvdata(dev); + + if (!device_property_read_u32(dev, "maxim,vmon-slot-no", &value)) + max98373->v_slot = value & 0xF; + else + max98373->v_slot = 0; + + if (!device_property_read_u32(dev, "maxim,imon-slot-no", &value)) + max98373->i_slot = value & 0xF; + else + max98373->i_slot = 1; + + if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value)) + max98373->spkfb_slot = value & 0xF; + else + max98373->spkfb_slot = 2; + + /* update interleave mode info */ + if (device_property_read_bool(dev, "maxim,interleave_mode")) + max98373->interleave_mode = true; + else + max98373->interleave_mode = false; +} + +/* MAX98373 reset and initialization. */ +static int max98373_reset(struct device *dev) +{ + int ret, reg, count; + struct max98373_priv *max98373 = dev_get_drvdata(dev); + + /* Software Reset */ + ret = regmap_update_bits(max98373->regmap, + MAX98373_R2000_SW_RESET, + MAX98373_SOFT_RESET, + MAX98373_SOFT_RESET); + if (ret) { + dev_err(dev, "Reset command failed. (ret:%d)\n", ret); + return ret; + } + + for (count = 0; count < 3; count++) { + usleep_range(10000, 11000); + /* Software Reset Verification */ + ret = regmap_read(max98373->regmap, + MAX98373_R21FF_REV_ID, &reg); + if (!ret) { + dev_dbg(dev, "Reset completed (retry:%d)\n", count); + return 0; + } + } + + dev_err(dev, "Reset failed. (ret:%d)\n", ret); + return ret; +} + +/* Power management functions and structure */ +static int max98373_suspend(struct device *dev) +{ + struct max98373_priv *max98373 = dev_get_drvdata(dev); + + regcache_cache_only(max98373->regmap, true); + regcache_mark_dirty(max98373->regmap); + return 0; +} + +static int max98373_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct max98373_priv *max98373 = dev_get_drvdata(dev); + unsigned long time; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(2000)); + if (!time) { + dev_err(dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + + regcache_cache_only(max98373->regmap, false); + regcache_sync(max98373->regmap); + + return 0; +} + +static const struct dev_pm_ops max98373_pm = { + SET_SYSTEM_SLEEP_PM_OPS(max98373_suspend, max98373_resume) + SET_RUNTIME_PM_OPS(max98373_suspend, max98373_resume, NULL) +}; + +static int max98373_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval, i, num_of_ports; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + /* BITMAP: 00001000 Dataport 3 is active */ + prop->source_ports = BIT(3); + /* BITMAP: 00000010 Dataport 1 is active */ + prop->sink_ports = BIT(1); + prop->paging_support = true; + prop->clk_stop_timeout = 20; + + nval = hweight32(prop->source_ports); + num_of_ports = nval; + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), + GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + num_of_ports += nval; + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), + GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* Allocate port_ready based on num_of_ports */ + slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports, + sizeof(*slave->port_ready), + GFP_KERNEL); + if (!slave->port_ready) + return -ENOMEM; + + /* Initialize completion */ + for (i = 0; i < num_of_ports; i++) + init_completion(&slave->port_ready[i]); + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int max98373_io_init(struct sdw_slave *slave) +{ + struct device *dev = &slave->dev; + struct max98373_priv *max98373 = dev_get_drvdata(dev); + int ret; + + if (max98373->pm_init_once) { + regcache_cache_only(max98373->regmap, false); + regcache_cache_bypass(max98373->regmap, true); + } + + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + if (!max98373->pm_init_once) { + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(dev, 3000); + pm_runtime_use_autosuspend(dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(dev); + + pm_runtime_enable(dev); + } + + pm_runtime_get_noresume(dev); + + /* Software Reset */ + ret = max98373_reset(dev); + if (ret) + return ret; + + /* Set soundwire mode */ + regmap_write(max98373->regmap, MAX98373_R2025_AUDIO_IF_MODE, 3); + /* Enable ADC */ + regmap_write(max98373->regmap, MAX98373_R2047_IV_SENSE_ADC_EN, 3); + /* Set default Soundwire clock */ + regmap_write(max98373->regmap, MAX98373_R2036_SOUNDWIRE_CTRL, 5); + /* Set default sampling rate for speaker and IVDAC */ + regmap_write(max98373->regmap, MAX98373_R2028_PCM_SR_SETUP_2, 0x88); + /* IV default slot configuration */ + regmap_write(max98373->regmap, + MAX98373_R2020_PCM_TX_HIZ_EN_1, + 0xFF); + regmap_write(max98373->regmap, + MAX98373_R2021_PCM_TX_HIZ_EN_2, + 0xFF); + /* L/R mix configuration */ + regmap_write(max98373->regmap, + MAX98373_R2029_PCM_TO_SPK_MONO_MIX_1, + 0x80); + regmap_write(max98373->regmap, + MAX98373_R202A_PCM_TO_SPK_MONO_MIX_2, + 0x1); + /* Set initial volume (0dB) */ + regmap_write(max98373->regmap, + MAX98373_R203D_AMP_DIG_VOL_CTRL, + 0x00); + regmap_write(max98373->regmap, + MAX98373_R203E_AMP_PATH_GAIN, + 0x00); + /* Enable DC blocker */ + regmap_write(max98373->regmap, + MAX98373_R203F_AMP_DSP_CFG, + 0x3); + /* Enable IMON VMON DC blocker */ + regmap_write(max98373->regmap, + MAX98373_R2046_IV_SENSE_ADC_DSP_CFG, + 0x7); + /* voltage, current slot configuration */ + regmap_write(max98373->regmap, + MAX98373_R2022_PCM_TX_SRC_1, + (max98373->i_slot << MAX98373_PCM_TX_CH_SRC_A_I_SHIFT | + max98373->v_slot) & 0xFF); + if (max98373->v_slot < 8) + regmap_update_bits(max98373->regmap, + MAX98373_R2020_PCM_TX_HIZ_EN_1, + 1 << max98373->v_slot, 0); + else + regmap_update_bits(max98373->regmap, + MAX98373_R2021_PCM_TX_HIZ_EN_2, + 1 << (max98373->v_slot - 8), 0); + + if (max98373->i_slot < 8) + regmap_update_bits(max98373->regmap, + MAX98373_R2020_PCM_TX_HIZ_EN_1, + 1 << max98373->i_slot, 0); + else + regmap_update_bits(max98373->regmap, + MAX98373_R2021_PCM_TX_HIZ_EN_2, + 1 << (max98373->i_slot - 8), 0); + + /* speaker feedback slot configuration */ + regmap_write(max98373->regmap, + MAX98373_R2023_PCM_TX_SRC_2, + max98373->spkfb_slot & 0xFF); + + /* Set interleave mode */ + if (max98373->interleave_mode) + regmap_update_bits(max98373->regmap, + MAX98373_R2024_PCM_DATA_FMT_CFG, + MAX98373_PCM_TX_CH_INTERLEAVE_MASK, + MAX98373_PCM_TX_CH_INTERLEAVE_MASK); + + /* Speaker enable */ + regmap_update_bits(max98373->regmap, + MAX98373_R2043_AMP_EN, + MAX98373_SPK_EN_MASK, 1); + + regmap_write(max98373->regmap, MAX98373_R20B5_BDE_EN, 1); + regmap_write(max98373->regmap, MAX98373_R20E2_LIMITER_EN, 1); + + if (max98373->pm_init_once) { + regcache_cache_bypass(max98373->regmap, false); + regcache_mark_dirty(max98373->regmap); + } + + max98373->pm_init_once = true; + max98373->hw_init = true; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + + return 0; +} + +static int max98373_clock_calculate(struct sdw_slave *slave, + unsigned int clk_freq) +{ + int x, y; + static const int max98373_clk_family[] = { + 7680000, 8400000, 9600000, 11289600, + 12000000, 12288000, 13000000 + }; + + for (x = 0; x < 4; x++) + for (y = 0; y < ARRAY_SIZE(max98373_clk_family); y++) + if (clk_freq == (max98373_clk_family[y] >> x)) + return (x << 3) + y; + + /* Set default clock (12.288 Mhz) if the value is not in the list */ + dev_err(&slave->dev, "Requested clock not found. (clk_freq = %d)\n", + clk_freq); + return 0x5; +} + +static int max98373_clock_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + struct device *dev = &slave->dev; + struct max98373_priv *max98373 = dev_get_drvdata(dev); + unsigned int clk_freq, value; + + clk_freq = (params->curr_dr_freq >> 1); + + /* + * Select the proper value for the register based on the + * requested clock. If the value is not in the list, + * use reasonable default - 12.288 Mhz + */ + value = max98373_clock_calculate(slave, clk_freq); + + /* SWCLK */ + regmap_write(max98373->regmap, MAX98373_R2036_SOUNDWIRE_CTRL, value); + + /* The default Sampling Rate value for IV is 48KHz*/ + regmap_write(max98373->regmap, MAX98373_R2028_PCM_SR_SETUP_2, 0x88); + + return 0; +} + +#define MAX98373_RATES SNDRV_PCM_RATE_8000_96000 +#define MAX98373_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) + +static int max98373_sdw_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98373_priv *max98373 = + snd_soc_component_get_drvdata(component); + + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int ret, chan_sz, sampling_rate; + + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!max98373->slave) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port_config.num = 1; + } else { + direction = SDW_DATA_DIR_TX; + port_config.num = 3; + } + + stream_config.frame_rate = params_rate(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + if (max98373->slot && direction == SDW_DATA_DIR_RX) { + stream_config.ch_count = max98373->slot; + port_config.ch_mask = max98373->rx_mask; + } else { + /* only IV are supported by capture */ + if (direction == SDW_DATA_DIR_TX) + stream_config.ch_count = 2; + else + stream_config.ch_count = params_channels(params); + + port_config.ch_mask = GENMASK(stream_config.ch_count - 1, 0); + } + + ret = sdw_stream_add_slave(max98373->slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (ret) { + dev_err(dai->dev, "Unable to configure port\n"); + return ret; + } + + if (params_channels(params) > 16) { + dev_err(component->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + /* Channel size configuration */ + switch (snd_pcm_format_width(params_format(params))) { + case 16: + chan_sz = MAX98373_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98373_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98373_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(component->dev, "Channel size unsupported %d\n", + params_format(params)); + return -EINVAL; + } + + max98373->ch_size = snd_pcm_format_width(params_format(params)); + + regmap_update_bits(max98373->regmap, + MAX98373_R2024_PCM_DATA_FMT_CFG, + MAX98373_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + dev_dbg(component->dev, "Format supported %d", params_format(params)); + + /* Sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate = MAX98373_PCM_SR_SET1_SR_8000; + break; + case 11025: + sampling_rate = MAX98373_PCM_SR_SET1_SR_11025; + break; + case 12000: + sampling_rate = MAX98373_PCM_SR_SET1_SR_12000; + break; + case 16000: + sampling_rate = MAX98373_PCM_SR_SET1_SR_16000; + break; + case 22050: + sampling_rate = MAX98373_PCM_SR_SET1_SR_22050; + break; + case 24000: + sampling_rate = MAX98373_PCM_SR_SET1_SR_24000; + break; + case 32000: + sampling_rate = MAX98373_PCM_SR_SET1_SR_32000; + break; + case 44100: + sampling_rate = MAX98373_PCM_SR_SET1_SR_44100; + break; + case 48000: + sampling_rate = MAX98373_PCM_SR_SET1_SR_48000; + break; + case 88200: + sampling_rate = MAX98373_PCM_SR_SET1_SR_88200; + break; + case 96000: + sampling_rate = MAX98373_PCM_SR_SET1_SR_96000; + break; + default: + dev_err(component->dev, "Rate %d is not supported\n", + params_rate(params)); + return -EINVAL; + } + + /* set correct sampling frequency */ + regmap_update_bits(max98373->regmap, + MAX98373_R2028_PCM_SR_SETUP_2, + MAX98373_PCM_SR_SET2_SR_MASK, + sampling_rate << MAX98373_PCM_SR_SET2_SR_SHIFT); + + /* set sampling rate of IV */ + regmap_update_bits(max98373->regmap, + MAX98373_R2028_PCM_SR_SETUP_2, + MAX98373_PCM_SR_SET2_IVADC_SR_MASK, + sampling_rate); + + return 0; +} + +static int max98373_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98373_priv *max98373 = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!max98373->slave) + return -EINVAL; + + sdw_stream_remove_slave(max98373->slave, stream->sdw_stream); + return 0; +} + +static int max98373_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void max98373_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int max98373_sdw_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct max98373_priv *max98373 = + snd_soc_component_get_drvdata(component); + + /* tx_mask is unused since it's irrelevant for I/V feedback */ + if (tx_mask) + return -EINVAL; + + if (!rx_mask && !slots && !slot_width) + max98373->tdm_mode = false; + else + max98373->tdm_mode = true; + + max98373->rx_mask = rx_mask; + max98373->slot = slots; + + return 0; +} + +static const struct snd_soc_dai_ops max98373_dai_sdw_ops = { + .hw_params = max98373_sdw_dai_hw_params, + .hw_free = max98373_pcm_hw_free, + .set_sdw_stream = max98373_set_sdw_stream, + .shutdown = max98373_shutdown, + .set_tdm_slot = max98373_sdw_set_tdm_slot, +}; + +static struct snd_soc_dai_driver max98373_sdw_dai[] = { + { + .name = "max98373-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98373_RATES, + .formats = MAX98373_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MAX98373_RATES, + .formats = MAX98373_FORMATS, + }, + .ops = &max98373_dai_sdw_ops, + } +}; + +static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) +{ + struct max98373_priv *max98373; + int ret; + struct device *dev = &slave->dev; + + /* Allocate and assign private driver data structure */ + max98373 = devm_kzalloc(dev, sizeof(*max98373), GFP_KERNEL); + if (!max98373) + return -ENOMEM; + + dev_set_drvdata(dev, max98373); + max98373->regmap = regmap; + max98373->slave = slave; + + /* Read voltage and slot configuration */ + max98373_read_config(slave); + + max98373->hw_init = false; + max98373->pm_init_once = false; + + /* codec registration */ + ret = devm_snd_soc_register_component(dev, &soc_codec_dev_max98373_sdw, + max98373_sdw_dai, + ARRAY_SIZE(max98373_sdw_dai)); + if (ret < 0) + dev_err(dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int max98373_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct max98373_priv *max98373 = dev_get_drvdata(&slave->dev); + + if (status == SDW_SLAVE_UNATTACHED) + max98373->hw_init = false; + + /* + * Perform initialization only if slave status is SDW_SLAVE_ATTACHED + */ + if (max98373->hw_init || status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return max98373_io_init(slave); +} + +static int max98373_bus_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + int ret; + + ret = max98373_clock_config(slave, params); + if (ret < 0) + dev_err(&slave->dev, "Invalid clk config"); + + return ret; +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static struct sdw_slave_ops max98373_slave_ops = { + .read_prop = max98373_read_prop, + .update_status = max98373_update_status, + .bus_config = max98373_bus_config, +}; + +static int max98373_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + + /* Regmap Initialization */ + regmap = devm_regmap_init_sdw(slave, &max98373_sdw_regmap); + if (!regmap) + return -EINVAL; + + return max98373_init(slave, regmap); +} + +#if defined(CONFIG_OF) +static const struct of_device_id max98373_of_match[] = { + { .compatible = "maxim,max98373", }, + {}, +}; +MODULE_DEVICE_TABLE(of, max98373_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id max98373_acpi_match[] = { + { "MX98373", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, max98373_acpi_match); +#endif + +static const struct sdw_device_id max98373_id[] = { + SDW_SLAVE_ENTRY(0x019F, 0x8373, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, max98373_id); + +static struct sdw_driver max98373_sdw_driver = { + .driver = { + .name = "max98373", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(max98373_of_match), + .acpi_match_table = ACPI_PTR(max98373_acpi_match), + .pm = &max98373_pm, + }, + .probe = max98373_sdw_probe, + .remove = NULL, + .ops = &max98373_slave_ops, + .id_table = max98373_id, +}; + +module_sdw_driver(max98373_sdw_driver); + +MODULE_DESCRIPTION("ASoC MAX98373 driver SDW"); +MODULE_AUTHOR("Oleg Sherbakov <oleg.sherbakov(a)maximintegrated.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/max98373-sdw.h b/sound/soc/codecs/max98373-sdw.h new file mode 100644 index 000000000000..9c40e671240c --- /dev/null +++ b/sound/soc/codecs/max98373-sdw.h @@ -0,0 +1,73 @@ +/* SPDX-License-Identifier: GPL-2.0-only + * Copyright (c) 2020 Maxim Integrated + */ + +#ifndef _MAX98373_SDW_H +#define _MAX98373_SDW_H + +#include "max98373.h" + +/* SoundWire Slave Control Port (SCP) */ +#define MAX98373_R0040_SCP_INIT_STAT_1 0x0040 +#define MAX98373_R0041_SCP_INIT_MASK_1 0x0041 +#define MAX98373_R0042_SCP_INIT_STAT_2 0x0042 +#define MAX98373_R0044_SCP_CTRL 0x0044 +#define MAX98373_R0045_SCP_SYSTEM_CTRL 0x0045 +#define MAX98373_R0046_SCP_DEV_NUMBER 0x0046 +#define MAX98373_R0050_SCP_DEV_ID_0 0x0050 +#define MAX98373_R0051_SCP_DEV_ID_1 0x0051 +#define MAX98373_R0052_SCP_DEV_ID_2 0x0052 +#define MAX98373_R0053_SCP_DEV_ID_3 0x0053 +#define MAX98373_R0054_SCP_DEV_ID_4 0x0054 +#define MAX98373_R0055_SCP_DEV_ID_5 0x0055 +#define MAX98373_R0060_SCP_FRAME_CTLR 0x0060 +#define MAX98373_R0070_SCP_FRAME_CTLR 0x0070 + +/* SoundWire Device Data Port (DP) */ +/* Data Port 1 Registers */ +#define MAX98373_R0100_DP1_INIT_STAT 0x0100 +#define MAX98373_R0101_DP1_INIT_MASK 0x0101 +#define MAX98373_R0102_DP1_PORT_CTRL 0x0102 +#define MAX98373_R0103_DP1_BLOCK_CTRL_1 0x0103 +#define MAX98373_R0104_DP1_PREPARE_STATUS 0x0104 +#define MAX98373_R0105_DP1_PREPARE_CTRL 0x0105 +/* Data Port 1 Bank 0 Registers */ +#define MAX98373_R0120_DP1_CHANNEL_EN 0x0120 +#define MAX98373_R0122_DP1_SAMPLE_CTRL1 0x0122 +#define MAX98373_R0123_DP1_SAMPLE_CTRL2 0x0123 +#define MAX98373_R0124_DP1_OFFSET_CTRL1 0x0124 +#define MAX98373_R0125_DP1_OFFSET_CTRL2 0x0125 +#define MAX98373_R0126_DP1_HCTRL 0x0126 +#define MAX98373_R0127_DP1_BLOCK_CTRL3 0x0127 +/* Data Port 1 Bank 1 Registers */ +#define MAX98373_R0130_DP1_CHANNEL_EN 0x0130 +#define MAX98373_R0132_DP1_SAMPLE_CTRL1 0x0132 +#define MAX98373_R0133_DP1_SAMPLE_CTRL2 0x0133 +#define MAX98373_R0134_DP1_OFFSET_CTRL1 0x0134 +#define MAX98373_R0135_DP1_OFFSET_CTRL2 0x0135 +#define MAX98373_R0136_DP1_HCTRL 0x0136 +#define MAX98373_R0137_DP1_BLOCK_CTRL3 0x0137 +/* Data Port 3 Registers */ +#define MAX98373_R0300_DP3_INIT_STAT 0x0300 +#define MAX98373_R0301_DP3_INIT_MASK 0x0301 +#define MAX98373_R0302_DP3_PORT_CTRL 0x0302 +#define MAX98373_R0303_DP3_BLOCK_CTRL_1 0x0303 +#define MAX98373_R0304_DP3_PREPARE_STATUS 0x0304 +#define MAX98373_R0305_DP3_PREPARE_CTRL 0x0305 +/* Data Port 3 Bank 0 Registers */ +#define MAX98373_R0320_DP3_CHANNEL_EN 0x0320 +#define MAX98373_R0322_DP3_SAMPLE_CTRL1 0x0322 +#define MAX98373_R0323_DP3_SAMPLE_CTRL2 0x0323 +#define MAX98373_R0324_DP3_OFFSET_CTRL1 0x0324 +#define MAX98373_R0325_DP3_OFFSET_CTRL2 0x0325 +#define MAX98373_R0326_DP3_HCTRL 0x0326 +#define MAX98373_R0327_DP3_BLOCK_CTRL3 0x0327 +/* Data Port 3 Bank 1 Registers */ +#define MAX98373_R0330_DP3_CHANNEL_EN 0x0330 +#define MAX98373_R0332_DP3_SAMPLE_CTRL1 0x0332 +#define MAX98373_R0333_DP3_SAMPLE_CTRL2 0x0333 +#define MAX98373_R0334_DP3_OFFSET_CTRL1 0x0334 +#define MAX98373_R0335_DP3_OFFSET_CTRL2 0x0335 +#define MAX98373_R0336_DP3_HCTRL 0x0336 +#define MAX98373_R0337_DP3_BLOCK_CTRL3 0x0337 +#endif diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 96718e3a1ad0..f03835e1f3e4 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -875,6 +875,21 @@ static const struct snd_soc_component_driver soc_codec_dev_max98373 = { .non_legacy_dai_naming = 1, }; +const struct snd_soc_component_driver soc_codec_dev_max98373_sdw = { + .probe = NULL, + .controls = max98373_snd_controls, + .num_controls = ARRAY_SIZE(max98373_snd_controls), + .dapm_widgets = max98373_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98373_dapm_widgets), + .dapm_routes = max98373_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98373_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_max98373_sdw); + static const struct regmap_config max98373_regmap = { .reg_bits = 16, .val_bits = 8, diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h index 63dae8be7105..507b3e967951 100644 --- a/sound/soc/codecs/max98373.h +++ b/sound/soc/codecs/max98373.h @@ -212,5 +212,13 @@ struct max98373_priv { bool interleave_mode; unsigned int ch_size; bool tdm_mode; + /* variables to support soundwire */ + struct sdw_slave *slave; + bool hw_init; + bool pm_init_once; + int slot; + unsigned int rx_mask; }; + +extern const struct snd_soc_component_driver soc_codec_dev_max98373_sdw; #endif base-commit: f61b9273c347980f570d1f9cecb867a7835c613d -- 2.20.1

3 4

presonus StudioLive 24.4.2
by m.eik michalke 19 Jun '20

19 Jun '20

hi, i'm trying to get a presonus StudioLive 24.4.2 up and running: https://www.presonus.com/products/studiolive-24.4.2 so far i'm partly successful -- looks like the board is automatically detected and supported by the firewire dice module. however, i'm only offered 16 of the 24 supported audio channels. in the list of soundcards it is merely shown as "studiolive" without further info on the model. i wasn't able to find further info on this board's status with regards to alsa, so if there's documentation on this somewhere, could someone please point me to it? in case that as of now the studiolive isn't officially supported or the above behavior is due to the fact that so far only the 16.4.2 is known to the module, i'm willing to help as far as i can ;) i hope that already seeing 16 channels is a good point to start and that it's not too hard to make the remaining 8 show up as well... best regards :: m.eik

2 8

[PATCH] ASoC: rockchip: Fix a reference count leak.
by wu000273＠umn.edu 18 Jun '20

18 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> Calling pm_runtime_get_sync increments the counter even in case of failure, causing incorrect ref count if pm_runtime_put is not called in error handling paths. Call pm_runtime_put if pm_runtime_get_sync fails. Fixes: fc05a5b22253 ("ASoC: rockchip: add support for pdm controller") Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> --- sound/soc/rockchip/rockchip_pdm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 7cd42fcfcf38..1707414cfa92 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -590,8 +590,10 @@ static int rockchip_pdm_resume(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put(dev); return ret; + } ret = regcache_sync(pdm->regmap); -- 2.17.1

3 2

[PATCH 13/29] dt: fix broken links due to txt->yaml renames
by Mauro Carvalho Chehab 18 Jun '20

18 Jun '20

There are some new broken doc links due to yaml renames at DT. Developers should really run: ./scripts/documentation-file-ref-check in order to solve those issues while submitting patches. This tool can even fix most of the issues with: ./scripts/documentation-file-ref-check --fix Signed-off-by: Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> --- Documentation/devicetree/bindings/display/bridge/sii902x.txt | 2 +- .../devicetree/bindings/display/rockchip/rockchip-drm.yaml | 2 +- Documentation/devicetree/bindings/net/mediatek-bluetooth.txt | 2 +- Documentation/devicetree/bindings/sound/audio-graph-card.txt | 2 +- Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt | 2 +- Documentation/mips/ingenic-tcu.rst | 2 +- MAINTAINERS | 4 ++-- 7 files changed, 8 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/display/bridge/sii902x.txt b/Documentation/devicetree/bindings/display/bridge/sii902x.txt index 6e14e087c0d0..0d1db3f9da84 100644 --- a/Documentation/devicetree/bindings/display/bridge/sii902x.txt +++ b/Documentation/devicetree/bindings/display/bridge/sii902x.txt @@ -37,7 +37,7 @@ Optional properties: simple-card or audio-graph-card binding. See their binding documents on how to describe the way the sii902x device is connected to the rest of the audio system: - Documentation/devicetree/bindings/sound/simple-card.txt + Documentation/devicetree/bindings/sound/simple-card.yaml Documentation/devicetree/bindings/sound/audio-graph-card.txt Note: In case of the audio-graph-card binding the used port index should be 3. diff --git a/Documentation/devicetree/bindings/display/rockchip/rockchip-drm.yaml b/Documentation/devicetree/bindings/display/rockchip/rockchip-drm.yaml index ec8ae742d4da..7204da5eb4c5 100644 --- a/Documentation/devicetree/bindings/display/rockchip/rockchip-drm.yaml +++ b/Documentation/devicetree/bindings/display/rockchip/rockchip-drm.yaml @@ -24,7 +24,7 @@ properties: description: | Should contain a list of phandles pointing to display interface port of vop devices. vop definitions as defined in - Documentation/devicetree/bindings/display/rockchip/rockchip-vop.txt + Documentation/devicetree/bindings/display/rockchip/rockchip-vop.yaml required: - compatible diff --git a/Documentation/devicetree/bindings/net/mediatek-bluetooth.txt b/Documentation/devicetree/bindings/net/mediatek-bluetooth.txt index 219bcbd0d344..9ef5bacda8c1 100644 --- a/Documentation/devicetree/bindings/net/mediatek-bluetooth.txt +++ b/Documentation/devicetree/bindings/net/mediatek-bluetooth.txt @@ -3,7 +3,7 @@ MediaTek SoC built-in Bluetooth Devices This device is a serial attached device to BTIF device and thus it must be a child node of the serial node with BTIF. The dt-bindings details for BTIF -device can be known via Documentation/devicetree/bindings/serial/8250.txt. +device can be known via Documentation/devicetree/bindings/serial/8250.yaml. Required properties: diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt index 269682619a70..d5f6919a2d69 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-card.txt @@ -5,7 +5,7 @@ It is based on common bindings for device graphs. see ${LINUX}/Documentation/devicetree/bindings/graph.txt Basically, Audio Graph Card property is same as Simple Card. -see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.txt +see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.yaml Below are same as Simple-Card. diff --git a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt index 4d51f3f5ea98..a6ffcdec6f6a 100644 --- a/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt @@ -5,7 +5,7 @@ codec or external codecs. sti sound drivers allows to expose sti SoC audio interface through the generic ASoC simple card. For details about sound card declaration please refer to -Documentation/devicetree/bindings/sound/simple-card.txt. +Documentation/devicetree/bindings/sound/simple-card.yaml. 1) sti-uniperiph-dai: audio dai device. --------------------------------------- diff --git a/Documentation/mips/ingenic-tcu.rst b/Documentation/mips/ingenic-tcu.rst index c5a646b14450..2b75760619b4 100644 --- a/Documentation/mips/ingenic-tcu.rst +++ b/Documentation/mips/ingenic-tcu.rst @@ -68,4 +68,4 @@ and frameworks can be controlled from the same registers, all of these drivers access their registers through the same regmap. For more information regarding the devicetree bindings of the TCU drivers, -have a look at Documentation/devicetree/bindings/timer/ingenic,tcu.txt. +have a look at Documentation/devicetree/bindings/timer/ingenic,tcu.yaml. diff --git a/MAINTAINERS b/MAINTAINERS index 68f21d46614c..0617dd671c2d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3946,7 +3946,7 @@ L: linux-crypto(a)vger.kernel.org S: Supported F: drivers/char/hw_random/cctrng.c F: drivers/char/hw_random/cctrng.h -F: Documentation/devicetree/bindings/rng/arm-cctrng.txt +F: Documentation/devicetree/bindings/rng/arm-cctrng.yaml W: https://developer.arm.com/products/system-ip/trustzone-cryptocell/cryptocel… CEC FRAMEWORK @@ -5490,7 +5490,7 @@ F: include/uapi/drm/r128_drm.h DRM DRIVER FOR RAYDIUM RM67191 PANELS M: Robert Chiras <robert.chiras(a)nxp.com> S: Maintained -F: Documentation/devicetree/bindings/display/panel/raydium,rm67191.txt +F: Documentation/devicetree/bindings/display/panel/raydium,rm67191.yaml F: drivers/gpu/drm/panel/panel-raydium-rm67191.c DRM DRIVER FOR ROCKTECH JH057N00900 PANELS -- 2.26.2

4 5

[PATCH AUTOSEL 5.7 055/388] ASoC: SOF: Do nothing when DSP PM callbacks are not set
by Sasha Levin 18 Jun '20

18 Jun '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> [ Upstream commit c26fde3b15ed41f5f452f1da727795f787833287 ] This provides a better separation between runtime and PM sleep callbacks. Only do nothing if given runtime flag is set and calback is not set. With the current implementation, if PM sleep callback is set but runtime callback is not set then at runtime resume we reload the firmware even if we do not support runtime resume callback. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20200515135958.17511-2-kai.vehmanen@linux.intel.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/pm.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index c410822d9920..01d83ddc16ba 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -90,7 +90,10 @@ static int sof_resume(struct device *dev, bool runtime_resume) int ret; /* do nothing if dsp resume callbacks are not set */ - if (!sof_ops(sdev)->resume || !sof_ops(sdev)->runtime_resume) + if (!runtime_resume && !sof_ops(sdev)->resume) + return 0; + + if (runtime_resume && !sof_ops(sdev)->runtime_resume) return 0; /* DSP was never successfully started, nothing to resume */ @@ -175,7 +178,10 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) int ret; /* do nothing if dsp suspend callback is not set */ - if (!sof_ops(sdev)->suspend) + if (!runtime_suspend && !sof_ops(sdev)->suspend) + return 0; + + if (runtime_suspend && !sof_ops(sdev)->runtime_suspend) return 0; if (sdev->fw_state != SOF_FW_BOOT_COMPLETE) -- 2.25.1

4 4

[PATCH RESEND 0/3] ASoC: Intel: simplify driver/card names for SOF/UCM integration
by Pierre-Louis Bossart 18 Jun '20

18 Jun '20

As suggested by Jaroslav, this patchset simplifies legacy cards compiled with SOF: they now expose an 'SOF' driver name and an 'sof-bytcht <codec>' card name. UCM uses this driver name and additionally checks for the card name to load a configuration shared with the SST driver. This patchset is just a rename with no functionality change. There is no modification when SOF is not used, and Kconfig for SOF are disabled when SST is enabled so no risk of interference. This helps make 5.9 the first kernel version where SOF 'just works'(tm) on legacy platforms with no additional configuration needed (as is already the case with more recent HDaudio+DMIC platforms). Thanks to Jaroslav for all the hard work on the alsa-lib/UCM side (the majority of the configs are already merged in alsa-ucm-conf, only minor updates are in-flight for max98090 and broadwell). Patches sent initially on May 29, probably missed with the merge window. No changes except for the addition of Jaroslav's Reviewed-by tag. Pierre-Louis Bossart (3): ASoC: Intel: byt*: simplify card names for SOF uses ASoC: Intel: cht*: simplify card names for SOF uses ASoC: Intel: broadwell: simplify card names for SOF uses sound/soc/intel/boards/bdw-rt5650.c | 12 +++++++++++- sound/soc/intel/boards/bdw-rt5677.c | 12 +++++++++++- sound/soc/intel/boards/broadwell.c | 12 +++++++++++- sound/soc/intel/boards/bytcht_cx2072x.c | 12 +++++++++++- sound/soc/intel/boards/bytcht_da7213.c | 12 +++++++++++- sound/soc/intel/boards/bytcht_es8316.c | 12 +++++++++++- sound/soc/intel/boards/bytcr_rt5640.c | 12 +++++++++++- sound/soc/intel/boards/bytcr_rt5651.c | 12 +++++++++++- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 12 +++++++++++- sound/soc/intel/boards/cht_bsw_nau8824.c | 12 +++++++++++- sound/soc/intel/boards/cht_bsw_rt5645.c | 17 +++++++++++++++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 12 +++++++++++- 12 files changed, 136 insertions(+), 13 deletions(-) -- 2.20.1

2 4

[PATCH] ANDROID: sound: usb: Add vendor's hooking interface
by JaeHun Jung 18 Jun '20

18 Jun '20

In mobile, a co-processor is used when using USB audio to improve power consumption. hooking is required for sync-up when operating the co-processor. So register call-back function. The main operation of the call-back function is as follows: - Initialize the co-processor by transmitting data when initializing. - Change the co-processor setting value through the interface function. - Configure sampling rate - pcm open/close Bug: 156315379 Change-Id: I32e1dd408e64aaef68ee06c480c4b4d4c95546dc Signed-off-by: JaeHun Jung <jh0801.jung(a)samsung.com> --- sound/usb/card.c | 16 ++++++++++++++++ sound/usb/card.h | 1 + sound/usb/clock.c | 5 +++++ sound/usb/pcm.c | 33 +++++++++++++++++++++++++++++++++ sound/usb/usbaudio.h | 30 ++++++++++++++++++++++++++++++ 5 files changed, 85 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index fd6fd17..2f3fa14 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -111,6 +111,7 @@ MODULE_PARM_DESC(skip_validation, "Skip unit descriptor validation (default: no) static DEFINE_MUTEX(register_mutex); static struct snd_usb_audio *usb_chip[SNDRV_CARDS]; static struct usb_driver usb_audio_driver; +struct snd_usb_audio_vendor_ops *usb_audio_ops; /* * disconnect streams @@ -210,6 +211,12 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int return 0; } +void snd_set_vender_interface(struct snd_usb_audio_vendor_ops *vendor_ops) +{ + usb_audio_ops = vendor_ops; +} +EXPORT_SYMBOL_GPL(snd_set_vender_interface); + /* * parse audio control descriptor and create pcm/midi streams */ @@ -598,6 +605,9 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) return err; + if (usb_audio_ops && usb_audio_ops->vendor_conn) + usb_audio_ops->vendor_conn(intf, dev); + /* * found a config. now register to ALSA */ @@ -653,6 +663,9 @@ static int usb_audio_probe(struct usb_interface *intf, } dev_set_drvdata(&dev->dev, chip); + if (usb_audio_ops && usb_audio_ops->vendor_usb_add_ctls) + usb_audio_ops->vendor_usb_add_ctls(chip, 0); + /* * For devices with more than one control interface, we assume the * first contains the audio controls. We might need a more specific @@ -737,6 +750,9 @@ static void usb_audio_disconnect(struct usb_interface *intf) card = chip->card; + if (usb_audio_ops && usb_audio_ops->vendor_disc) + usb_audio_ops->vendor_disc(); + mutex_lock(&register_mutex); if (atomic_inc_return(&chip->shutdown) == 1) { struct snd_usb_stream *as; diff --git a/sound/usb/card.h b/sound/usb/card.h index 395403a..a55bb4c 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -175,5 +175,6 @@ struct snd_usb_stream { struct snd_usb_substream substream[2]; struct list_head list; }; +void snd_set_vender_interface(struct snd_usb_audio_vendor_ops *vendor_ops); #endif /* __USBAUDIO_CARD_H */ diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b118cf9..0ceeccb 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -642,8 +642,13 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, * interface is active. */ if (rate != prev_rate) { usb_set_interface(dev, iface, 0); + if (usb_audio_ops && usb_audio_ops->vendor_set_intf) + usb_audio_ops->vendor_set_intf(dev, alts, iface, 0); snd_usb_set_interface_quirk(dev); usb_set_interface(dev, iface, fmt->altsetting); + if (usb_audio_ops && usb_audio_ops->vendor_set_intf) + usb_audio_ops->vendor_set_intf(dev, alts, iface, + fmt->altsetting); snd_usb_set_interface_quirk(dev); } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a4e4064..6cdacac 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -134,6 +134,9 @@ static struct audioformat *find_format(struct snd_usb_substream *subs) found = fp; cur_attr = attr; } + + if (usb_audio_ops && usb_audio_ops->vendor_pcm_binterval) + usb_audio_ops->vendor_pcm_binterval(fp, found, &cur_attr, &attr); } return found; } @@ -568,6 +571,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) } dev_dbg(&dev->dev, "setting usb interface %d:%d\n", fmt->iface, fmt->altsetting); + if (usb_audio_ops && usb_audio_ops->vendor_set_pcm_intf) + usb_audio_ops->vendor_set_pcm_intf(dev, fmt->iface, + fmt->altsetting, subs->direction); snd_usb_set_interface_quirk(dev); } @@ -891,6 +897,15 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) struct usb_interface *iface; int ret; + if (usb_audio_ops && usb_audio_ops->vendor_set_pcmbuf) { + ret = usb_audio_ops->vendor_set_pcmbuf(subs->dev); + + if (ret < 0) { + dev_err(&subs->dev->dev, "pcm buf transfer failed\n"); + return ret; + } + } + if (! subs->cur_audiofmt) { dev_err(&subs->dev->dev, "no format is specified!\n"); return -ENXIO; @@ -924,6 +939,15 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) if (ret < 0) goto unlock; + if (usb_audio_ops && usb_audio_ops->vendor_set_rate) { + subs->need_setup_ep = false; + usb_audio_ops->vendor_set_rate( + subs->cur_audiofmt->iface, + subs->cur_rate, + subs->cur_audiofmt->altsetting); + goto unlock; + } + ret = configure_endpoint(subs); if (ret < 0) goto unlock; @@ -1333,6 +1357,9 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream) struct snd_usb_substream *subs = &as->substream[direction]; int ret; + if (usb_audio_ops && usb_audio_ops->vendor_pcm_con) + usb_audio_ops->vendor_pcm_con(true, direction); + subs->interface = -1; subs->altset_idx = 0; runtime->hw = snd_usb_hardware; @@ -1361,12 +1388,18 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream) struct snd_usb_substream *subs = &as->substream[direction]; int ret; + if (usb_audio_ops && usb_audio_ops->vendor_pcm_con) + usb_audio_ops->vendor_pcm_con(false, direction); + snd_media_stop_pipeline(subs); if (!as->chip->keep_iface && subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { usb_set_interface(subs->dev, subs->interface, 0); + if (usb_audio_ops && usb_audio_ops->vendor_set_pcm_intf) + usb_audio_ops->vendor_set_pcm_intf(subs->dev, + subs->interface, 0, direction); subs->interface = -1; ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D1); snd_usb_unlock_shutdown(subs->stream->chip); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 1c892c7..a2fd8a4 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -124,4 +124,34 @@ void snd_usb_unlock_shutdown(struct snd_usb_audio *chip); extern bool snd_usb_use_vmalloc; extern bool snd_usb_skip_validation; +/* for vender function mapping */ +extern struct snd_usb_audio_vendor_ops *usb_audio_ops; + +/* USB audio interface function for audio core */ +struct snd_usb_audio_vendor_ops { + /* Set descriptors and memory map */ + void (*vendor_conn)(struct usb_interface *intf, + struct usb_device *udev); + /* Set disconnection */ + void (*vendor_disc)(void); + /* Set interface info and setting value */ + int (*vendor_set_intf)(struct usb_device *udev, + struct usb_host_interface *alts, int iface, int alt); + /* Set sample rate */ + int (*vendor_set_rate)(int iface, int rate, int alt); + /* Alloc pcm buffer */ + int (*vendor_set_pcmbuf)(struct usb_device *udev); + /* Set pcm interface */ + int (*vendor_set_pcm_intf)(struct usb_device *udev, + int iface, int alt, int direction); + /* informed whether pcm open/close to vendor */ + void (*vendor_pcm_con)(int onoff, int direction); + /* set datainterval */ + void (*vendor_pcm_binterval)(void *fp, void *found, + int *cur_attr, int *attr); + /* control USB F/W */ + int (*vendor_usb_add_ctls)(struct snd_usb_audio *chip, + unsigned long private_value); +}; + #endif /* __USBAUDIO_H */ -- 2.7.4

4 3

[PATCH 2/2] ASoC: amd: closing specific instance.
by Ravulapati Vishnu vardhan rao 18 Jun '20

18 Jun '20

The steps to reproduce: Record from the internal mic : (arecord -D hw:1,2 -f dat /dev/null -V stereos) Record from the headphone mic: (arecord -D hw:1,0 -f dat /dev/null -V stereos) Kill the recording from internal mic. We can see the recording from the headphone mic is broken. This patch rectifies the issue reported. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 30 ++++++++++++++++++++++------- 1 file changed, 23 insertions(+), 7 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index d8f554f369a8..e6386de20ac7 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -342,11 +342,34 @@ static int acp3x_dma_close(struct snd_soc_component *component, { struct snd_soc_pcm_runtime *prtd; struct i2s_dev_data *adata; + struct i2s_stream_instance *ins; prtd = substream->private_data; component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); adata = dev_get_drvdata(component->dev); + ins = substream->runtime->private_data; + if (!ins) + return -EINVAL; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (ins->i2s_instance) { + case I2S_BT_INSTANCE: + adata->play_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->i2ssp_play_stream = NULL; + } + } else { + switch (ins->i2s_instance) { + case I2S_BT_INSTANCE: + adata->capture_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->i2ssp_capture_stream = NULL; + } + } /* Disable ACP irq, when the current stream is being closed and * another stream is also not active. @@ -354,13 +377,6 @@ static int acp3x_dma_close(struct snd_soc_component *component, if (!adata->play_stream && !adata->capture_stream && !adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) rv_writel(0, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adata->play_stream = NULL; - adata->i2ssp_play_stream = NULL; - } else { - adata->capture_stream = NULL; - adata->i2ssp_capture_stream = NULL; - } return 0; } -- 2.17.1

2 1

[PATCH 00/13] ALSA: hda: Use LED classdev for mute controls consistently
by Takashi Iwai 18 Jun '20

18 Jun '20

Hi, this is a patch set inspired by the recent patch Kai-Heng posted about the HD-audio mic-mute LED control. Currently HD-audio driver deals with the mute and mic-mute LED in several different ways: primarily with the direct callback of vmaster hook and capture sync hook, while another with the LED class device binding. The latter has been used for binding with the platform device LEDs like Thinkpad, Dell, Huawei. And, yet, recently we added our own LED classdev for the mic-mute LED on some HP systems although they are controlled directly with the callback; it's exposed, however, for the DMIC that is governed by a different ASoC driver. This patch set is an attempt to sort out and make them consistent: namely, * All LEDs are now controlled via LED class device * The generic driver provides helper functions to easily build up the LED class dev and the relevant mixer controls * Conversion of the existing framework and clean ups The patches are lightly tested in my side with a couple of machines and also through hda-emu tests. Some devices receive new kcontrols for the mute LED behavior (that have been missing so far), but anything else look good though my tests. thanks, Takashi === Takashi Iwai (13): ALSA: hda: generic: Always call led-trigger for mic mute LED ALSA: hda: generic: Add a helper for mic-mute LED with LED classdev ALSA: hda/realtek: Convert to cdev-variant of mic-mute LED controls ALSA: hda/conexant: Convert to cdev-variant of mic-mute LED controls ALSA: hda/sigmatel: Convert to cdev-variant of mic-mute LED controls ALSA: hda: generic: Drop unused snd_hda_gen_fixup_micmute_led() ALSA: hda: generic: Drop the old mic-mute LED hook ALSA: hda: generic: Add vmaster mute LED helper ALSA: hda/realtek: Use the new vmaster mute LED helper ALSA: hda/conexant: Use the new vmaster mute LED helper ALSA: hda/sigmatel: Use the new vmaster mute LED helper ALSA: hda/realtek: Unify LED helper code ALSA: hda: Let LED cdev handling suspend/resume sound/pci/hda/Kconfig | 9 ++ sound/pci/hda/hda_generic.c | 134 ++++++++++++++++-------- sound/pci/hda/hda_generic.h | 13 ++- sound/pci/hda/patch_conexant.c | 47 ++++----- sound/pci/hda/patch_realtek.c | 220 +++++++++++++++++++--------------------- sound/pci/hda/patch_sigmatel.c | 22 ++-- sound/pci/hda/thinkpad_helper.c | 19 +--- 7 files changed, 248 insertions(+), 216 deletions(-) -- 2.16.4

1 13

[PATCH v3 1/2] ALSA: hda/realtek: Add COEF controlled micmute LED support
by Kai-Heng Feng 18 Jun '20

18 Jun '20

Currently, HDA codec LED class can only be used by GPIO controlled LED. However, there are some new systems that control LED via COEF instead of GPIO. In order to support those systems, create a new helper that can be facilitated by both COEF controlled and GPIO controlled LED. In addition to that, add LED_CORE_SUSPENDRESUME flag since some systems don't restore the LED properly after suspend. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v3: - Wording. - Properly prefix exported symbol. v2: - Prevent platforms like Dell, Lenovoe and Huawei create double LED class devices. sound/pci/hda/hda_generic.c | 7 ++++--- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/patch_realtek.c | 29 ++++++++++++++++------------- 3 files changed, 21 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f4e9d9445e18..e4f534e9a88c 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3897,7 +3897,7 @@ enum { MICMUTE_LED_FOLLOW_MUTE, }; -static void call_micmute_led_update(struct hda_codec *codec) +void snd_hda_gen_call_micmute_led_update(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; unsigned int val; @@ -3924,6 +3924,7 @@ static void call_micmute_led_update(struct hda_codec *codec) if (spec->micmute_led.update) spec->micmute_led.update(codec); } +EXPORT_SYMBOL_GPL(snd_hda_gen_call_micmute_led_update); static void update_micmute_led(struct hda_codec *codec, struct snd_kcontrol *kcontrol, @@ -3945,7 +3946,7 @@ static void update_micmute_led(struct hda_codec *codec, spec->micmute_led.capture |= mask; else spec->micmute_led.capture &= ~mask; - call_micmute_led_update(codec); + snd_hda_gen_call_micmute_led_update(codec); } } @@ -3982,7 +3983,7 @@ static int micmute_led_mode_put(struct snd_kcontrol *kcontrol, if (mode == spec->micmute_led.led_mode) return 0; spec->micmute_led.led_mode = mode; - call_micmute_led_update(codec); + snd_hda_gen_call_micmute_led_update(codec); return 1; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index fb9f1a90238b..062be242339a 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -353,6 +353,7 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on); int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin); +void snd_hda_gen_call_micmute_led_update(struct hda_codec *codec); int snd_hda_gen_add_micmute_led(struct hda_codec *codec, void (*hook)(struct hda_codec *)); void snd_hda_gen_fixup_micmute_led(struct hda_codec *codec, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d73f8beadb6..0587d1e96b19 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4114,10 +4114,10 @@ static int micmute_led_set(struct led_classdev *led_cdev, enum led_brightness brightness) { struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent); - struct alc_spec *spec = codec->spec; + struct hda_gen_spec *spec = codec->spec; - alc_update_gpio_led(codec, spec->gpio_mic_led_mask, - spec->micmute_led_polarity, !!brightness); + spec->micmute_led.led_mode = !brightness; + snd_hda_gen_call_micmute_led_update(codec); return 0; } @@ -4126,7 +4126,17 @@ static struct led_classdev micmute_led_cdev = { .max_brightness = 1, .brightness_set_blocking = micmute_led_set, .default_trigger = "audio-micmute", + .flags = LED_CORE_SUSPENDRESUME, }; + +static void alc_register_micmute_led(struct hda_codec *codec) +{ + micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + if (devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev)) + codec_warn(codec, "failed to register micmute LED\n"); +} +#else +static inline void alc_register_micmute_led(struct hda_codec *codec) {} #endif /* setup mute and mic-mute GPIO bits, add hooks appropriately */ @@ -4136,9 +4146,6 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, unsigned int micmute_mask) { struct alc_spec *spec = codec->spec; -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - int err; -#endif alc_fixup_gpio(codec, action, mute_mask | micmute_mask); @@ -4151,13 +4158,7 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, if (micmute_mask) { spec->gpio_mic_led_mask = micmute_mask; snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); - -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); - err = devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev); - if (err) - codec_warn(codec, "failed to register micmute LED\n"); -#endif + alc_register_micmute_led(codec); } } @@ -4305,6 +4306,7 @@ static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec, spec->mic_led_coefbit_on = 1<<13; spec->mic_led_coefbit_off = 0; snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + alc_register_micmute_led(codec); } } @@ -4319,6 +4321,7 @@ static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec, spec->mic_led_coefbit_on = 2<<2; spec->mic_led_coefbit_off = 1<<2; snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + alc_register_micmute_led(codec); } } -- 2.17.1

2 11

[RFC PATCH 0/2] TAS2563 DSP Firmware Loader
by Dan Murphy 18 Jun '20

18 Jun '20

Hello The TAS2563 amplifier has a DSP that can run programs and configurations to produce alternate audio experiences. The DSP firmware is not a typical firmware as the binary may contain various programs and configurations that are selectable during run time. These programs and configurations are selectable via files under the I2C dev node. There may be a better way to select this through ALSA controls but I was unable to find a good example of this. This is why this is an RFC patchset. Dan Dan Murphy (2): dt-bindings: tas2562: Add firmware support for tas2563 ASoc: tas2563: DSP Firmware loading support .../devicetree/bindings/sound/tas2562.yaml | 4 + sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/tas2562.c | 48 ++- sound/soc/codecs/tas2562.h | 25 ++ sound/soc/codecs/tas25xx_dsp_loader.c | 377 ++++++++++++++++++ sound/soc/codecs/tas25xx_dsp_loader.h | 93 +++++ 6 files changed, 530 insertions(+), 19 deletions(-) create mode 100644 sound/soc/codecs/tas25xx_dsp_loader.c create mode 100644 sound/soc/codecs/tas25xx_dsp_loader.h -- 2.26.2

3 19

File Plugin seems to drop data when writing to FIFO
by GitHub issues - opened 18 Jun '20

18 Jun '20

alsa-project/alsa-lib issue #63 was opened from riggiding: Hi, we're using the file plugin to pipe data for use in a vu-meter application. This worked well but since updating to 1.1.4.1 the data received is incomplete (tested using sine test tones and Audacity to playback / inspect the data that went through the fifo). **When writing to a normal file all is fine.** Also tried using tee, same result. Any hints or ideas would be great! **The fifo is created by:** #! /bin/sh if [ ! -p "$ALSA_FIFO" ]; then # create a fifo for alsa -> vumeter mkfifo "$ALSA_FIFO" chmod 666 "$ALSA_FIFO" fi cat <>"$ALSA_FIFO" **asound.conf:** pcm.hiface { type hw card 0 device 0 } pcm.vumeter { type rate slave { pcm "alsaFifo" rate 44100 format "S16_LE" } } pcm.alsaFifo { type file slave.pcm "null" file "/tmp/alsa-fifo" } pcm.alsaFifoTee { type empty slave.pcm "tee:null,'/tmp/alsa-fifo',raw" } pcm.direct { type route slave.pcm "split_direct" ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 } pcm.split_direct { type multi slaves.a.pcm "hiface" slaves.a.channels 2 slaves.b.pcm "vumeter" slaves.b.channels 2 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave b bindings.2.channel 0 bindings.3.slave b bindings.3.channel 1 } Issue URL : https://github.com/alsa-project/alsa-lib/issues/63 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda: Intel: add missing PCI IDs for ICL-H, TGL-H and EKL
by Pierre-Louis Bossart 18 Jun '20

18 Jun '20

Mirror PCI ids used for SOF. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d20aedd103c6..3565e2ab0965 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2470,6 +2470,9 @@ static const struct pci_device_id azx_ids[] = { /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Icelake-H */ + { PCI_DEVICE(0x8086, 0x3dc8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Jasperlake */ { PCI_DEVICE(0x8086, 0x38c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, @@ -2478,9 +2481,14 @@ static const struct pci_device_id azx_ids[] = { /* Tigerlake */ { PCI_DEVICE(0x8086, 0xa0c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Tigerlake-H */ + { PCI_DEVICE(0x8086, 0x43c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4b58), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, -- 2.20.1

2 1

[PATCH 1/2] ASoC: amd: Removing unnecessary instance initialization
by Ravulapati Vishnu vardhan rao 18 Jun '20

18 Jun '20

In trigger we already get the selected instance details from runtime->private_data.So, removing the local initialization which may corrupt the instance selected details and this leads to corrupt data. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-i2s.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-i2s.c b/sound/soc/amd/raven/acp3x-i2s.c index a532e01a2622..14607563abd2 100644 --- a/sound/soc/amd/raven/acp3x-i2s.c +++ b/sound/soc/amd/raven/acp3x-i2s.c @@ -151,20 +151,12 @@ static int acp3x_i2s_trigger(struct snd_pcm_substream *substream, struct i2s_stream_instance *rtd; struct snd_soc_pcm_runtime *prtd; struct snd_soc_card *card; - struct acp3x_platform_info *pinfo; u32 ret, val, period_bytes, reg_val, ier_val, water_val; u32 buf_size, buf_reg; prtd = substream->private_data; rtd = substream->runtime->private_data; card = prtd->card; - pinfo = snd_soc_card_get_drvdata(card); - if (pinfo) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rtd->i2s_instance = pinfo->play_i2s_instance; - else - rtd->i2s_instance = pinfo->cap_i2s_instance; - } period_bytes = frames_to_bytes(substream->runtime, substream->runtime->period_size); buf_size = frames_to_bytes(substream->runtime, -- 2.17.1

1 0

[PATCH AUTOSEL 4.4 52/60] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ] fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index ffc000bc1f15..56a873ba08e4 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -243,6 +243,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } -- 2.25.1

1 0

[PATCH AUTOSEL 4.4 24/60] ALSA: usb-audio: Improve frames size computation
by Sasha Levin 18 Jun '20

18 Jun '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ] For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 4 ++++ sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++----- sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 2 ++ 4 files changed, 45 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 71778ca4b26a..844c68863810 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -80,6 +80,10 @@ struct snd_usb_endpoint { dma_addr_t sync_dma; /* DMA address of syncbuf */ unsigned int pipe; /* the data i/o pipe */ + unsigned int framesize[2]; /* small/large frame sizes in samples */ + unsigned int sample_rem; /* remainder from division fs/fps */ + unsigned int sample_accum; /* sample accumulator */ + unsigned int fps; /* frames per second */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ int freqshift; /* how much to shift the feedback value to get Q16.16 */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 66648b4bdd28..666731317b33 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -137,12 +137,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) /* * For streaming based on information derived from sync endpoints, - * prepare_outbound_urb_sizes() will call next_packet_size() to + * prepare_outbound_urb_sizes() will call slave_next_packet_size() to * determine the number of samples to be sent in the next packet. * - * For implicit feedback, next_packet_size() is unused. + * For implicit feedback, slave_next_packet_size() is unused. */ -int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -159,6 +159,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) return ret; } +/* + * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes() + * will call next_packet_size() to determine the number of samples to be + * sent in the next packet. + */ +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +{ + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + ep->sample_accum += ep->sample_rem; + if (ep->sample_accum >= ep->fps) { + ep->sample_accum -= ep->fps; + ret = ep->framesize[1]; + } else { + ret = ep->framesize[0]; + } + + return ret; +} + static void retire_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *urb_ctx) { @@ -203,6 +226,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep, if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); @@ -879,10 +904,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->maxpacksize = fmt->maxpacksize; ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); - if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) { ep->freqn = get_usb_full_speed_rate(rate); - else + ep->fps = 1000; + } else { ep->freqn = get_usb_high_speed_rate(rate); + ep->fps = 8000; + } + + ep->sample_rem = rate % ep->fps; + ep->framesize[0] = rate / ep->fps; + ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps; /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -941,6 +973,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; + ep->sample_accum = 0; snd_usb_endpoint_start_quirk(ep); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 584f295d7c77..4aad49cbeb5f 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -27,6 +27,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f84c55ecd0fb..53d91cae86f9 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1473,6 +1473,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); -- 2.25.1

1 0

[PATCH AUTOSEL 4.4 05/60] ALSA: isa/wavefront: prevent out of bounds write in ioctl
by Sasha Levin 18 Jun '20

18 Jun '20

From: Dan Carpenter <dan.carpenter(a)oracle.com> [ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ] The "header->number" comes from the ioctl and it needs to be clamped to prevent out of bounds writes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/wavefront/wavefront_synth.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 718d5e3b7806..6c06d0645779 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1174,7 +1174,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) "alias for %d\n", header->number, header->hdr.a.OriginalSample); - + + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + munge_int32 (header->number, &alias_hdr[0], 2); munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2); munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset), @@ -1205,6 +1208,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) int num_samples; unsigned char *msample_hdr; + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL); if (! msample_hdr) return -ENOMEM; -- 2.25.1

1 0

[PATCH AUTOSEL 4.9 69/80] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ] fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index dc30d780f874..3fcf174b99d3 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -243,6 +243,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } -- 2.25.1

1 0

[PATCH AUTOSEL 4.9 31/80] ALSA: usb-audio: Improve frames size computation
by Sasha Levin 18 Jun '20

18 Jun '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ] For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 4 ++++ sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++----- sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 2 ++ 4 files changed, 45 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 111b0f009afa..c4599cf0ddc9 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -80,6 +80,10 @@ struct snd_usb_endpoint { dma_addr_t sync_dma; /* DMA address of syncbuf */ unsigned int pipe; /* the data i/o pipe */ + unsigned int framesize[2]; /* small/large frame sizes in samples */ + unsigned int sample_rem; /* remainder from division fs/fps */ + unsigned int sample_accum; /* sample accumulator */ + unsigned int fps; /* frames per second */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ int freqshift; /* how much to shift the feedback value to get Q16.16 */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 30aa5f2df6da..b5207e71ed72 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -137,12 +137,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) /* * For streaming based on information derived from sync endpoints, - * prepare_outbound_urb_sizes() will call next_packet_size() to + * prepare_outbound_urb_sizes() will call slave_next_packet_size() to * determine the number of samples to be sent in the next packet. * - * For implicit feedback, next_packet_size() is unused. + * For implicit feedback, slave_next_packet_size() is unused. */ -int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -159,6 +159,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) return ret; } +/* + * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes() + * will call next_packet_size() to determine the number of samples to be + * sent in the next packet. + */ +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +{ + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + ep->sample_accum += ep->sample_rem; + if (ep->sample_accum >= ep->fps) { + ep->sample_accum -= ep->fps; + ret = ep->framesize[1]; + } else { + ret = ep->framesize[0]; + } + + return ret; +} + static void retire_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *urb_ctx) { @@ -203,6 +226,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep, if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); @@ -875,10 +900,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->maxpacksize = fmt->maxpacksize; ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); - if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) { ep->freqn = get_usb_full_speed_rate(rate); - else + ep->fps = 1000; + } else { ep->freqn = get_usb_high_speed_rate(rate); + ep->fps = 8000; + } + + ep->sample_rem = rate % ep->fps; + ep->framesize[0] = rate / ep->fps; + ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps; /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -937,6 +969,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; + ep->sample_accum = 0; snd_usb_endpoint_start_quirk(ep); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 584f295d7c77..4aad49cbeb5f 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -27,6 +27,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 9bc995f9b4e1..615213aeda33 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1483,6 +1483,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); -- 2.25.1

1 0

[PATCH AUTOSEL 4.9 09/80] ALSA: isa/wavefront: prevent out of bounds write in ioctl
by Sasha Levin 18 Jun '20

18 Jun '20

From: Dan Carpenter <dan.carpenter(a)oracle.com> [ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ] The "header->number" comes from the ioctl and it needs to be clamped to prevent out of bounds writes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/wavefront/wavefront_synth.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 718d5e3b7806..6c06d0645779 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1174,7 +1174,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) "alias for %d\n", header->number, header->hdr.a.OriginalSample); - + + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + munge_int32 (header->number, &alias_hdr[0], 2); munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2); munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset), @@ -1205,6 +1208,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) int num_samples; unsigned char *msample_hdr; + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL); if (! msample_hdr) return -ENOMEM; -- 2.25.1

1 0

[PATCH AUTOSEL 4.14 086/108] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ] fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index e1b97e59275a..15d7e6da0555 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -243,6 +243,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } -- 2.25.1

1 0

[PATCH AUTOSEL 4.14 041/108] ALSA: usb-audio: Improve frames size computation
by Sasha Levin 18 Jun '20

18 Jun '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ] For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 4 ++++ sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++----- sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 2 ++ 4 files changed, 45 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index ed87cc83eb47..9dbcbb27c28e 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -81,6 +81,10 @@ struct snd_usb_endpoint { dma_addr_t sync_dma; /* DMA address of syncbuf */ unsigned int pipe; /* the data i/o pipe */ + unsigned int framesize[2]; /* small/large frame sizes in samples */ + unsigned int sample_rem; /* remainder from division fs/fps */ + unsigned int sample_accum; /* sample accumulator */ + unsigned int fps; /* frames per second */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ int freqshift; /* how much to shift the feedback value to get Q16.16 */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 8caf0b57f9c6..841219560872 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -137,12 +137,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) /* * For streaming based on information derived from sync endpoints, - * prepare_outbound_urb_sizes() will call next_packet_size() to + * prepare_outbound_urb_sizes() will call slave_next_packet_size() to * determine the number of samples to be sent in the next packet. * - * For implicit feedback, next_packet_size() is unused. + * For implicit feedback, slave_next_packet_size() is unused. */ -int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -159,6 +159,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) return ret; } +/* + * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes() + * will call next_packet_size() to determine the number of samples to be + * sent in the next packet. + */ +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +{ + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + ep->sample_accum += ep->sample_rem; + if (ep->sample_accum >= ep->fps) { + ep->sample_accum -= ep->fps; + ret = ep->framesize[1]; + } else { + ret = ep->framesize[0]; + } + + return ret; +} + static void retire_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *urb_ctx) { @@ -203,6 +226,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep, if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); @@ -889,10 +914,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->maxpacksize = fmt->maxpacksize; ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); - if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) { ep->freqn = get_usb_full_speed_rate(rate); - else + ep->fps = 1000; + } else { ep->freqn = get_usb_high_speed_rate(rate); + ep->fps = 8000; + } + + ep->sample_rem = rate % ep->fps; + ep->framesize[0] = rate / ep->fps; + ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps; /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -951,6 +983,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; + ep->sample_accum = 0; snd_usb_endpoint_start_quirk(ep); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 63a39d4fa8d8..d23fa0a8c11b 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -28,6 +28,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index ff38fca1781b..fd73186d6003 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1484,6 +1484,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); -- 2.25.1

1 0

[PATCH AUTOSEL 4.14 012/108] ALSA: isa/wavefront: prevent out of bounds write in ioctl
by Sasha Levin 18 Jun '20

18 Jun '20

From: Dan Carpenter <dan.carpenter(a)oracle.com> [ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ] The "header->number" comes from the ioctl and it needs to be clamped to prevent out of bounds writes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/wavefront/wavefront_synth.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 0b1e4b34b299..13c8e6542a2f 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1175,7 +1175,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) "alias for %d\n", header->number, header->hdr.a.OriginalSample); - + + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + munge_int32 (header->number, &alias_hdr[0], 2); munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2); munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset), @@ -1206,6 +1209,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) int num_samples; unsigned char *msample_hdr; + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL); if (! msample_hdr) return -ENOMEM; -- 2.25.1

1 0

[PATCH AUTOSEL 4.14 009/108] ASoC: davinci-mcasp: Fix dma_chan refcnt leak when getting dma type
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit a697ae6ea56e23397341b027098c1b11d9ab13da ] davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a reference of the specified dma_chan object to "chan" with increased refcnt. When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in one exception handling path of davinci_mcasp_get_dma_type(). When chan device is NULL, the function forgets to decrease the refcnt increased by dma_request_chan(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when chan device is NULL. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/davinci/davinci-mcasp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e10e03800cce..6991718d7c8a 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1747,8 +1747,10 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) PTR_ERR(chan)); return PTR_ERR(chan); } - if (WARN_ON(!chan->device || !chan->device->dev)) + if (WARN_ON(!chan->device || !chan->device->dev)) { + dma_release_channel(chan); return -EINVAL; + } if (chan->device->dev->of_node) ret = of_property_read_string(chan->device->dev->of_node, -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 166/172] ASoC: rt5645: Add platform-data for Asus T101HA
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 79d4f823a06796656289f97b922493da5690e46c ] The Asus T101HA uses the default jack-detect mode 3, but instead of using an analog microphone it is using a DMIC on dmic-data-pin 1, like the Asus T100HA. Note unlike the T100HA its jack-detect is not inverted. Add a DMI quirk with the correct settings for this model. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608204634.93407-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7e3b47eeea04..9185bd7c5a6d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3656,6 +3656,12 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = { .inv_jd1_1 = true, }; +static const struct rt5645_platform_data asus_t101ha_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_IN2N, + .dmic2_data_pin = RT5645_DMIC2_DISABLE, + .jd_mode = 3, +}; + static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = { .jd_mode = 3, .in2_diff = true, @@ -3733,6 +3739,14 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&asus_t100ha_platform_data, }, + { + .ident = "ASUS T101HA", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"), + }, + .driver_data = (void *)&asus_t101ha_platform_data, + }, { .ident = "MINIX Z83-4", .matches = { -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 165/172] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT10-A tablet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 199a5e8fda54ab3c8c6f6bf980c004e97ebf5ccb ] The Toshiba Encore WT10-A tablet almost fully works with the default settings for Bay Trail CR devices. The only issue is that it uses a digital mic. connected the the DMIC1 input instead of an analog mic. Add a quirk for this model using the default settings with the input-map replaced with BYT_RT5640_DMIC1_MAP. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608204634.93407-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 5dfc17987b2b..0dcd249877c5 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -724,6 +724,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_JD_NOT_INV | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT10-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 164/172] ASoC: core: only convert non DPCM link to DPCM link
by Sasha Levin 18 Jun '20

18 Jun '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 607fa205a7e4dfad28b8a67ab1c985756ddbccb0 ] Additional checks for valid DAIs expose a corner case, where existing BE dailinks get modified, e.g. HDMI links are tagged with dpcm_capture=1 even if the DAIs are for playback. This patch makes those changes conditional and flags configuration issues when a BE dailink is has no_pcm=0 but dpcm_playback or dpcm_capture=1 (which makes no sense). As discussed on the alsa-devel mailing list, there are redundant flags for dpcm_playback, dpcm_capture, playback_only, capture_only. This will have to be cleaned-up in a future update. For now only correct and flag problematic configurations. Fixes: 218fe9b7ec7f3 ("ASoC: soc-core: Set dpcm_playback / dpcm_capture") Suggested-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608194415.4663-3-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-core.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e45dfcb62b6a..595fe20bbc6d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1920,9 +1920,25 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dai_link->platform_name = component->name; /* convert non BE into BE */ - dai_link->no_pcm = 1; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; + if (!dai_link->no_pcm) { + dai_link->no_pcm = 1; + + if (dai_link->dpcm_playback) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n", + dai_link->name); + if (dai_link->dpcm_capture) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n", + dai_link->name); + + /* convert normal link into DPCM one */ + if (!(dai_link->dpcm_playback || + dai_link->dpcm_capture)) { + dai_link->dpcm_playback = !dai_link->capture_only; + dai_link->dpcm_capture = !dai_link->playback_only; + } + } /* override any BE fixups */ dai_link->be_hw_params_fixup = -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 133/172] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ] fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 1033ac6631b0..b9ac448989ed 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -241,6 +241,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 119/172] ASoC: fix incomplete error-handling in img_i2s_in_probe.
by Sasha Levin 18 Jun '20

18 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> [ Upstream commit 25bf943e4e7b47282bd86ae7d39e039217ebb007 ] Function "pm_runtime_get_sync()" is not handled by "pm_runtime_put()" if "PTR_ERR(rst) == -EPROBE_DEFER". Fix this issue by adding "pm_runtime_put()" into this error path. Fixes: f65bb92ca12e ("ASoC: img-i2s-in: Add runtime PM") Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> Link: https://lore.kernel.org/r/20200525055011.31925-1-wu000273@umn.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/img/img-i2s-in.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index 388cefd7340a..c22880aea82a 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -485,6 +485,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) if (IS_ERR(rst)) { if (PTR_ERR(rst) == -EPROBE_DEFER) { ret = -EPROBE_DEFER; + pm_runtime_put(&pdev->dev); goto err_suspend; } -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 098/172] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT8-A tablet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 0e0e10fde0e9808d1991268f5dca69fb36c025f7 ] The Toshiba Encore WT8-A tablet almost fully works with the default settings for non-CR Bay Trail devices. The only problem is that its jack-detect switch is not inverted (it is active high instead of the normal active low). Add a quirk for this model using the default settings + BYT_RT5640_JD_NOT_INV. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200518072416.5348-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index f29014a7d672..5dfc17987b2b 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -712,6 +712,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT8-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT8-A"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 071/172] soundwire: slave: don't init debugfs on device registration error
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 8893ab5e8ee5d7c12e0fc1dca4a309475064473d ] The error handling flow seems incorrect, there is no reason to try and add debugfs support if the device registration did not succeed. Return on error. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Link: https://lore.kernel.org/r/20200419185117.4233-2-yung-chuan.liao@linux.intel… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/slave.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index ac103bd0c176..b6330b6672d5 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -55,6 +55,8 @@ static int sdw_slave_add(struct sdw_bus *bus, list_del(&slave->node); mutex_unlock(&bus->bus_lock); put_device(&slave->dev); + + return ret; } return ret; -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 069/172] slimbus: ngd: get drvdata from correct device
by Sasha Levin 18 Jun '20

18 Jun '20

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit b58c663059b484f7ff547d076a34cf6d7a302e56 ] Get drvdata directly from parent instead of ngd dev, as ngd dev can probe defer and previously set drvdata will become null. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20200417093618.7929-1-srinivas.kandagatla@linaro.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 9221ba7b7863..f40ac8dcb081 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1350,7 +1350,6 @@ static int of_qcom_slim_ngd_register(struct device *parent, ngd->pdev->driver_override = QCOM_SLIM_NGD_DRV_NAME; ngd->pdev->dev.of_node = node; ctrl->ngd = ngd; - platform_set_drvdata(ngd->pdev, ctrl); platform_device_add(ngd->pdev); ngd->base = ctrl->base + ngd->id * data->offset + @@ -1365,12 +1364,13 @@ static int of_qcom_slim_ngd_register(struct device *parent, static int qcom_slim_ngd_probe(struct platform_device *pdev) { - struct qcom_slim_ngd_ctrl *ctrl = platform_get_drvdata(pdev); struct device *dev = &pdev->dev; + struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev->parent); int ret; ctrl->ctrl.dev = dev; + platform_set_drvdata(pdev, ctrl); pm_runtime_use_autosuspend(dev); pm_runtime_set_autosuspend_delay(dev, QCOM_SLIM_NGD_AUTOSUSPEND); pm_runtime_set_suspended(dev); -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 065/172] ALSA: usb-audio: Fix racy list management in output queue
by Sasha Levin 18 Jun '20

18 Jun '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5b6cc38f3f3f37109ce72b60bda215a5f6892c0b ] The linked list entry from FIFO is peeked at queue_pending_output_urbs() but the actual element pop-out is performed outside the spinlock, and it's potentially racy. Do delete the link at the right place inside the spinlock. Fixes: 8fdff6a319e7 ("ALSA: snd-usb: implement new endpoint streaming model") Link: https://lore.kernel.org/r/20200424074016.14301-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/endpoint.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 36e255d88937..48611849b79b 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -359,17 +359,17 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) ep->next_packet_read_pos %= MAX_URBS; /* take URB out of FIFO */ - if (!list_empty(&ep->ready_playback_urbs)) + if (!list_empty(&ep->ready_playback_urbs)) { ctx = list_first_entry(&ep->ready_playback_urbs, struct snd_urb_ctx, ready_list); + list_del_init(&ctx->ready_list); + } } spin_unlock_irqrestore(&ep->lock, flags); if (ctx == NULL) return; - list_del_init(&ctx->ready_list); - /* copy over the length information */ for (i = 0; i < packet->packets; i++) ctx->packet_size[i] = packet->packet_size[i]; -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 064/172] ALSA: usb-audio: Improve frames size computation
by Sasha Levin 18 Jun '20

18 Jun '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ] For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 4 ++++ sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++----- sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 2 ++ 4 files changed, 45 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 7f11655bde50..36198ddb4754 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -83,6 +83,10 @@ struct snd_usb_endpoint { dma_addr_t sync_dma; /* DMA address of syncbuf */ unsigned int pipe; /* the data i/o pipe */ + unsigned int framesize[2]; /* small/large frame sizes in samples */ + unsigned int sample_rem; /* remainder from division fs/fps */ + unsigned int sample_accum; /* sample accumulator */ + unsigned int fps; /* frames per second */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ int freqshift; /* how much to shift the feedback value to get Q16.16 */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index aeb74cc6ceff..36e255d88937 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -137,12 +137,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) /* * For streaming based on information derived from sync endpoints, - * prepare_outbound_urb_sizes() will call next_packet_size() to + * prepare_outbound_urb_sizes() will call slave_next_packet_size() to * determine the number of samples to be sent in the next packet. * - * For implicit feedback, next_packet_size() is unused. + * For implicit feedback, slave_next_packet_size() is unused. */ -int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -159,6 +159,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) return ret; } +/* + * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes() + * will call next_packet_size() to determine the number of samples to be + * sent in the next packet. + */ +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +{ + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + ep->sample_accum += ep->sample_rem; + if (ep->sample_accum >= ep->fps) { + ep->sample_accum -= ep->fps; + ret = ep->framesize[1]; + } else { + ret = ep->framesize[0]; + } + + return ret; +} + static void retire_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *urb_ctx) { @@ -203,6 +226,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep, if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); @@ -887,10 +912,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->maxpacksize = fmt->maxpacksize; ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); - if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) { ep->freqn = get_usb_full_speed_rate(rate); - else + ep->fps = 1000; + } else { ep->freqn = get_usb_high_speed_rate(rate); + ep->fps = 8000; + } + + ep->sample_rem = rate % ep->fps; + ep->framesize[0] = rate / ep->fps; + ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps; /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -949,6 +981,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; + ep->sample_accum = 0; snd_usb_endpoint_start_quirk(ep); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 63a39d4fa8d8..d23fa0a8c11b 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -28,6 +28,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 8b91be394407..42ae774f6db7 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1558,6 +1558,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 015/172] ALSA: isa/wavefront: prevent out of bounds write in ioctl
by Sasha Levin 18 Jun '20

18 Jun '20

From: Dan Carpenter <dan.carpenter(a)oracle.com> [ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ] The "header->number" comes from the ioctl and it needs to be clamped to prevent out of bounds writes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/wavefront/wavefront_synth.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 0b1e4b34b299..13c8e6542a2f 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1175,7 +1175,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) "alias for %d\n", header->number, header->hdr.a.OriginalSample); - + + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + munge_int32 (header->number, &alias_hdr[0], 2); munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2); munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset), @@ -1206,6 +1209,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) int num_samples; unsigned char *msample_hdr; + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL); if (! msample_hdr) return -ENOMEM; -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 014/172] ALSA: hda/realtek - Introduce polarity for micmute LED GPIO
by Sasha Levin 18 Jun '20

18 Jun '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit dbd13179780555ecd3c992dea1222ca31920e892 ] Currently mute LED and micmute LED share the same GPIO polarity. So split the polarity for mute and micmute, in case they have different polarities. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200430083255.5093-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b06f7d52faad..9ab82ead1825 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -94,6 +94,7 @@ struct alc_spec { /* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */ int mute_led_polarity; + int micmute_led_polarity; hda_nid_t mute_led_nid; hda_nid_t cap_mute_led_nid; @@ -3862,11 +3863,9 @@ static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec, /* update LED status via GPIO */ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, - bool enabled) + int polarity, bool enabled) { - struct alc_spec *spec = codec->spec; - - if (spec->mute_led_polarity) + if (polarity) enabled = !enabled; alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */ } @@ -3877,7 +3876,8 @@ static void alc_fixup_gpio_mute_hook(void *private_data, int enabled) struct hda_codec *codec = private_data; struct alc_spec *spec = codec->spec; - alc_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); + alc_update_gpio_led(codec, spec->gpio_mute_led_mask, + spec->mute_led_polarity, enabled); } /* turn on/off mic-mute LED via GPIO per capture hook */ @@ -3886,6 +3886,7 @@ static void alc_gpio_micmute_update(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_update_gpio_led(codec, spec->gpio_mic_led_mask, + spec->micmute_led_polarity, spec->gen.micmute_led.led_value); } @@ -5476,7 +5477,8 @@ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, snd_hda_gen_hp_automute(codec, jack); /* mute_led_polarity is set to 0, so we pass inverted value here */ - alc_update_gpio_led(codec, 0x10, !spec->gen.hp_jack_present); + alc_update_gpio_led(codec, 0x10, spec->mute_led_polarity, + !spec->gen.hp_jack_present); } /* Manage GPIOs for HP EliteBook Folio 9480m. -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 011/172] ASoC: davinci-mcasp: Fix dma_chan refcnt leak when getting dma type
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit a697ae6ea56e23397341b027098c1b11d9ab13da ] davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a reference of the specified dma_chan object to "chan" with increased refcnt. When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in one exception handling path of davinci_mcasp_get_dma_type(). When chan device is NULL, the function forgets to decrease the refcnt increased by dma_request_chan(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when chan device is NULL. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/davinci/davinci-mcasp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 14ab16e1369f..1203de8aab99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1758,8 +1758,10 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) PTR_ERR(chan)); return PTR_ERR(chan); } - if (WARN_ON(!chan->device || !chan->device->dev)) + if (WARN_ON(!chan->device || !chan->device->dev)) { + dma_release_channel(chan); return -EINVAL; + } if (chan->device->dev->of_node) ret = of_property_read_string(chan->device->dev->of_node, -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 003/172] ASoC: tegra: tegra_wm8903: Support nvidia, headset property
by Sasha Levin 18 Jun '20

18 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 69bc9461974b..301850df368d 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -173,6 +173,7 @@ 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; @@ -185,12 +186,15 @@ 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

1 0

[PATCH AUTOSEL 5.4 253/266] ASoC: rt5645: Add platform-data for Asus T101HA
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 79d4f823a06796656289f97b922493da5690e46c ] The Asus T101HA uses the default jack-detect mode 3, but instead of using an analog microphone it is using a DMIC on dmic-data-pin 1, like the Asus T100HA. Note unlike the T100HA its jack-detect is not inverted. Add a DMI quirk with the correct settings for this model. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608204634.93407-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 19662ee330d6..c83f7f5da96b 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3625,6 +3625,12 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = { .inv_jd1_1 = true, }; +static const struct rt5645_platform_data asus_t101ha_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_IN2N, + .dmic2_data_pin = RT5645_DMIC2_DISABLE, + .jd_mode = 3, +}; + static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = { .jd_mode = 3, .in2_diff = true, @@ -3702,6 +3708,14 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&asus_t100ha_platform_data, }, + { + .ident = "ASUS T101HA", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"), + }, + .driver_data = (void *)&asus_t101ha_platform_data, + }, { .ident = "MINIX Z83-4", .matches = { -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 252/266] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT10-A tablet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 199a5e8fda54ab3c8c6f6bf980c004e97ebf5ccb ] The Toshiba Encore WT10-A tablet almost fully works with the default settings for Bay Trail CR devices. The only issue is that it uses a digital mic. connected the the DMIC1 input instead of an analog mic. Add a quirk for this model using the default settings with the input-map replaced with BYT_RT5640_DMIC1_MAP. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608204634.93407-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index f222e1091bae..f7964d1ec486 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -754,6 +754,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_JD_NOT_INV | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT10-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 251/266] ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit ba4e5abc6c4e173af7c941c03c067263b686665d ] With additional checks on dailinks, we see errors such as [ 3.000418] sof-nocodec sof-nocodec: CPU DAI DMIC01 Pin for rtd NoCodec-6 does not support playback It's not clear why we set the dpcm_playback and dpcm_capture flags unconditionally, add a check on number of channels for each direction to avoid invalid configurations. Fixes: 8017b8fd37bf5e ('ASoC: SOF: Add Nocodec machine driver support') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608194415.4663-5-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/nocodec.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c index 3d128e5a132c..ea0fe9a09f3f 100644 --- a/sound/soc/sof/nocodec.c +++ b/sound/soc/sof/nocodec.c @@ -52,8 +52,10 @@ static int sof_nocodec_bes_setup(struct device *dev, links[i].platforms->name = dev_name(dev); links[i].codecs->dai_name = "snd-soc-dummy-dai"; links[i].codecs->name = "snd-soc-dummy"; - links[i].dpcm_playback = 1; - links[i].dpcm_capture = 1; + if (ops->drv[i].playback.channels_min) + links[i].dpcm_playback = 1; + if (ops->drv[i].capture.channels_min) + links[i].dpcm_capture = 1; } card->dai_link = links; -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 250/266] ASoC: core: only convert non DPCM link to DPCM link
by Sasha Levin 18 Jun '20

18 Jun '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 607fa205a7e4dfad28b8a67ab1c985756ddbccb0 ] Additional checks for valid DAIs expose a corner case, where existing BE dailinks get modified, e.g. HDMI links are tagged with dpcm_capture=1 even if the DAIs are for playback. This patch makes those changes conditional and flags configuration issues when a BE dailink is has no_pcm=0 but dpcm_playback or dpcm_capture=1 (which makes no sense). As discussed on the alsa-devel mailing list, there are redundant flags for dpcm_playback, dpcm_capture, playback_only, capture_only. This will have to be cleaned-up in a future update. For now only correct and flag problematic configurations. Fixes: 218fe9b7ec7f3 ("ASoC: soc-core: Set dpcm_playback / dpcm_capture") Suggested-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608194415.4663-3-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-core.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0215e2c94bf0..9df20768a8f2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1895,9 +1895,25 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dai_link->platforms->name = component->name; /* convert non BE into BE */ - dai_link->no_pcm = 1; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; + if (!dai_link->no_pcm) { + dai_link->no_pcm = 1; + + if (dai_link->dpcm_playback) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n", + dai_link->name); + if (dai_link->dpcm_capture) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n", + dai_link->name); + + /* convert normal link into DPCM one */ + if (!(dai_link->dpcm_playback || + dai_link->dpcm_capture)) { + dai_link->dpcm_playback = !dai_link->capture_only; + dai_link->dpcm_capture = !dai_link->playback_only; + } + } /* override any BE fixups */ dai_link->be_hw_params_fixup = -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 209/266] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ] fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 01052a0808b0..5aee6b8366d2 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -241,6 +241,7 @@ static int fsl_asrc_dma_hw_params(struct snd_pcm_substream *substream, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 190/266] ASoC: fix incomplete error-handling in img_i2s_in_probe.
by Sasha Levin 18 Jun '20

18 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> [ Upstream commit 25bf943e4e7b47282bd86ae7d39e039217ebb007 ] Function "pm_runtime_get_sync()" is not handled by "pm_runtime_put()" if "PTR_ERR(rst) == -EPROBE_DEFER". Fix this issue by adding "pm_runtime_put()" into this error path. Fixes: f65bb92ca12e ("ASoC: img-i2s-in: Add runtime PM") Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> Link: https://lore.kernel.org/r/20200525055011.31925-1-wu000273@umn.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/img/img-i2s-in.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index fdd2c73fd2fa..869fe0068cbd 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -482,6 +482,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) if (IS_ERR(rst)) { if (PTR_ERR(rst) == -EPROBE_DEFER) { ret = -EPROBE_DEFER; + pm_runtime_put(&pdev->dev); goto err_suspend; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 157/266] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT8-A tablet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 0e0e10fde0e9808d1991268f5dca69fb36c025f7 ] The Toshiba Encore WT8-A tablet almost fully works with the default settings for non-CR Bay Trail devices. The only problem is that its jack-detect switch is not inverted (it is active high instead of the normal active low). Add a quirk for this model using the default settings + BYT_RT5640_JD_NOT_INV. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200518072416.5348-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index e62e1d7815aa..f222e1091bae 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -742,6 +742,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT8-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT8-A"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 145/266] ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback
by Sasha Levin 18 Jun '20

18 Jun '20

From: Erwin Burema <e.burema(a)gmail.com> [ Upstream commit 10ce77e4817fef99e1166be7e6685a80c63bf77f ] For USB sound devices using implicit feedback the endpoint used for this feedback should be able to be opened twice, once for required feedback and second time for audio data. This way these devices can be put in duplex audio mode. Since this only works if the settings of the endpoint don't change a check is included for this. This fixes bug 207023 ("MOTU M2 regression on duplex audio") and should also fix bug 103751 ("M-Audio Fast Track Ultra usb audio device will not operate full-duplex") Fixes: c249177944b6 ("ALSA: usb-audio: add implicit fb quirk for MOTU M Series") Signed-off-by: Erwin Burema <e.burema(a)gmail.com> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207023 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=103751 Link: https://lore.kernel.org/r/2410739.SCZni40SNb@alpha-wolf Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 1 + sound/usb/endpoint.c | 195 ++++++++++++++++++++++++++++++++++++++++++- sound/usb/pcm.c | 5 ++ 3 files changed, 197 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 820e564656ed..d6219fba9699 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -108,6 +108,7 @@ struct snd_usb_endpoint { int iface, altsetting; int skip_packets; /* quirks for devices to ignore the first n packets in a stream */ + bool is_implicit_feedback; /* This endpoint is used as implicit feedback */ spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 50104f658ed4..9bea7d3f99f8 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -522,6 +522,8 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, list_add_tail(&ep->list, &chip->ep_list); + ep->is_implicit_feedback = 0; + __exit_unlock: mutex_unlock(&chip->mutex); @@ -621,6 +623,178 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force) ep->nurbs = 0; } +/* + * Check data endpoint for format differences + */ +static bool check_ep_params(struct snd_usb_endpoint *ep, + snd_pcm_format_t pcm_format, + unsigned int channels, + unsigned int period_bytes, + unsigned int frames_per_period, + unsigned int periods_per_buffer, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep) +{ + unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb; + unsigned int max_packs_per_period, urbs_per_period, urb_packs; + unsigned int max_urbs; + int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels; + int tx_length_quirk = (ep->chip->tx_length_quirk && + usb_pipeout(ep->pipe)); + bool ret = 1; + + if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) { + /* + * When operating in DSD DOP mode, the size of a sample frame + * in hardware differs from the actual physical format width + * because we need to make room for the DOP markers. + */ + frame_bits += channels << 3; + } + + ret = ret && (ep->datainterval == fmt->datainterval); + ret = ret && (ep->stride == frame_bits >> 3); + + switch (pcm_format) { + case SNDRV_PCM_FORMAT_U8: + ret = ret && (ep->silence_value == 0x80); + break; + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + ret = ret && (ep->silence_value == 0x69); + break; + default: + ret = ret && (ep->silence_value == 0); + } + + /* assume max. frequency is 50% higher than nominal */ + ret = ret && (ep->freqmax == ep->freqn + (ep->freqn >> 1)); + /* Round up freqmax to nearest integer in order to calculate maximum + * packet size, which must represent a whole number of frames. + * This is accomplished by adding 0x0.ffff before converting the + * Q16.16 format into integer. + * In order to accurately calculate the maximum packet size when + * the data interval is more than 1 (i.e. ep->datainterval > 0), + * multiply by the data interval prior to rounding. For instance, + * a freqmax of 41 kHz will result in a max packet size of 6 (5.125) + * frames with a data interval of 1, but 11 (10.25) frames with a + * data interval of 2. + * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the + * maximum datainterval value of 3, at USB full speed, higher for + * USB high speed, noting that ep->freqmax is in units of + * frames per packet in Q16.16 format.) + */ + maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * + (frame_bits >> 3); + if (tx_length_quirk) + maxsize += sizeof(__le32); /* Space for length descriptor */ + /* but wMaxPacketSize might reduce this */ + if (ep->maxpacksize && ep->maxpacksize < maxsize) { + /* whatever fits into a max. size packet */ + unsigned int data_maxsize = maxsize = ep->maxpacksize; + + if (tx_length_quirk) + /* Need to remove the length descriptor to calc freq */ + data_maxsize -= sizeof(__le32); + ret = ret && (ep->freqmax == (data_maxsize / (frame_bits >> 3)) + << (16 - ep->datainterval)); + } + + if (ep->fill_max) + ret = ret && (ep->curpacksize == ep->maxpacksize); + else + ret = ret && (ep->curpacksize == maxsize); + + if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) { + packs_per_ms = 8 >> ep->datainterval; + max_packs_per_urb = MAX_PACKS_HS; + } else { + packs_per_ms = 1; + max_packs_per_urb = MAX_PACKS; + } + if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep)) + max_packs_per_urb = min(max_packs_per_urb, + 1U << sync_ep->syncinterval); + max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval); + + /* + * Capture endpoints need to use small URBs because there's no way + * to tell in advance where the next period will end, and we don't + * want the next URB to complete much after the period ends. + * + * Playback endpoints with implicit sync much use the same parameters + * as their corresponding capture endpoint. + */ + if (usb_pipein(ep->pipe) || + snd_usb_endpoint_implicit_feedback_sink(ep)) { + + urb_packs = packs_per_ms; + /* + * Wireless devices can poll at a max rate of once per 4ms. + * For dataintervals less than 5, increase the packet count to + * allow the host controller to use bursting to fill in the + * gaps. + */ + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) { + int interval = ep->datainterval; + + while (interval < 5) { + urb_packs <<= 1; + ++interval; + } + } + /* make capture URBs <= 1 ms and smaller than a period */ + urb_packs = min(max_packs_per_urb, urb_packs); + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + ret = ret && (ep->nurbs == MAX_URBS); + + /* + * Playback endpoints without implicit sync are adjusted so that + * a period fits as evenly as possible in the smallest number of + * URBs. The total number of URBs is adjusted to the size of the + * ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits. + */ + } else { + /* determine how small a packet can be */ + minsize = (ep->freqn >> (16 - ep->datainterval)) * + (frame_bits >> 3); + /* with sync from device, assume it can be 12% lower */ + if (sync_ep) + minsize -= minsize >> 3; + minsize = max(minsize, 1u); + + /* how many packets will contain an entire ALSA period? */ + max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize); + + /* how many URBs will contain a period? */ + urbs_per_period = DIV_ROUND_UP(max_packs_per_period, + max_packs_per_urb); + /* how many packets are needed in each URB? */ + urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period); + + /* limit the number of frames in a single URB */ + ret = ret && (ep->max_urb_frames == + DIV_ROUND_UP(frames_per_period, urbs_per_period)); + + /* try to use enough URBs to contain an entire ALSA buffer */ + max_urbs = min((unsigned) MAX_URBS, + MAX_QUEUE * packs_per_ms / urb_packs); + ret = ret && (ep->nurbs == min(max_urbs, + urbs_per_period * periods_per_buffer)); + } + + ret = ret && (ep->datainterval == fmt->datainterval); + ret = ret && (ep->maxpacksize == fmt->maxpacksize); + ret = ret && + (ep->fill_max == !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX)); + + return ret; +} + /* * configure a data endpoint */ @@ -886,10 +1060,23 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, int err; if (ep->use_count != 0) { - usb_audio_warn(ep->chip, - "Unable to change format on ep #%x: already in use\n", - ep->ep_num); - return -EBUSY; + bool check = ep->is_implicit_feedback && + check_ep_params(ep, pcm_format, + channels, period_bytes, + period_frames, buffer_periods, + fmt, sync_ep); + + if (!check) { + usb_audio_warn(ep->chip, + "Unable to change format on ep #%x: already in use\n", + ep->ep_num); + return -EBUSY; + } + + usb_audio_dbg(ep->chip, + "Ep #%x already in use as implicit feedback but format not changed\n", + ep->ep_num); + return 0; } /* release old buffers, if any */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e52d129085c0..6c391e5fad2a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -386,6 +386,8 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, if (!subs->sync_endpoint) return -EINVAL; + subs->sync_endpoint->is_implicit_feedback = 1; + subs->data_endpoint->sync_master = subs->sync_endpoint; return 1; @@ -484,12 +486,15 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, implicit_fb ? SND_USB_ENDPOINT_TYPE_DATA : SND_USB_ENDPOINT_TYPE_SYNC); + if (!subs->sync_endpoint) { if (is_playback && attr == USB_ENDPOINT_SYNC_NONE) return 0; return -EINVAL; } + subs->sync_endpoint->is_implicit_feedback = implicit_fb; + subs->data_endpoint->sync_master = subs->sync_endpoint; return 0; -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 136/266] ASoC: ti: omap-mcbsp: Fix an error handling path in 'asoc_mcbsp_probe()'
by Sasha Levin 18 Jun '20

18 Jun '20

From: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> [ Upstream commit 03990fd58d2b7c8f7d53e514ba9b8749fac260f9 ] If an error occurs after the call to 'omap_mcbsp_init()', the reference to 'mcbsp->fclk' must be decremented, as already done in the remove function. This can be achieved easily by using the devm_ variant of 'clk_get()' when the reference is taken in 'omap_mcbsp_init()' This fixes the leak in the probe and has the side effect to simplify both the error handling path of 'omap_mcbsp_init()' and the remove function. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> Acked-by: Peter Ujfalusi <peter.ujflausi(a)ti.com> Link: https://lore.kernel.org/r/20200512134325.252073-1-christophe.jaillet@wanado… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-mcbsp.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 26b503bbdb5f..3273b317fa3b 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -686,7 +686,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, SNDRV_PCM_STREAM_CAPTURE); - mcbsp->fclk = clk_get(&pdev->dev, "fck"); + mcbsp->fclk = devm_clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { ret = PTR_ERR(mcbsp->fclk); dev_err(mcbsp->dev, "unable to get fck: %d\n", ret); @@ -711,7 +711,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); - goto err_thres; + return ret; } } @@ -724,8 +724,6 @@ static int omap_mcbsp_init(struct platform_device *pdev) err_st: if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); -err_thres: - clk_put(mcbsp->fclk); return ret; } @@ -1442,8 +1440,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) omap_mcbsp_st_cleanup(pdev); - clk_put(mcbsp->fclk); - return 0; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 135/266] ASoC: ux500: mop500: Fix some refcounted resources issues
by Sasha Levin 18 Jun '20

18 Jun '20

From: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> [ Upstream commit 4e8748fcaeec073e3ba794871ce86c545e4f961f ] There are 2 issues here: - if one of the 'of_parse_phandle' fails, calling 'mop500_of_node_put()' is a no-op because the 'mop500_dai_links' structure has not been initialized yet, so the referenced are not decremented - The reference stored in 'mop500_dai_links[i].codecs' is refcounted only once in the probe and must be decremented only once. Fixes: 39013bd60e79 ("ASoC: Ux500: Dispose of device nodes correctly") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> Link: https://lore.kernel.org/r/20200512100705.246349-1-christophe.jaillet@wanado… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ux500/mop500.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 2873e8e6f02b..cdae1190b930 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -63,10 +63,11 @@ static void mop500_of_node_put(void) { int i; - for (i = 0; i < 2; i++) { + for (i = 0; i < 2; i++) of_node_put(mop500_dai_links[i].cpus->of_node); - of_node_put(mop500_dai_links[i].codecs->of_node); - } + + /* Both links use the same codec, which is refcounted only once */ + of_node_put(mop500_dai_links[0].codecs->of_node); } static int mop500_of_probe(struct platform_device *pdev, @@ -81,7 +82,9 @@ static int mop500_of_probe(struct platform_device *pdev, if (!(msp_np[0] && msp_np[1] && codec_np)) { dev_err(&pdev->dev, "Phandle missing or invalid\n"); - mop500_of_node_put(); + for (i = 0; i < 2; i++) + of_node_put(msp_np[i]); + of_node_put(codec_np); return -EINVAL; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 134/266] SoC: rsnd: add interrupt support for SSI BUSIF buffer
by Sasha Levin 18 Jun '20

18 Jun '20

From: Yongbo Zhang <giraffesnn123(a)gmail.com> [ Upstream commit 66c705d07d784fb6b4622c6e47b6acae357472db ] SSI BUSIF buffer is possible to overflow or underflow, especially in a hypervisor environment. If there is no interrupt support, it will eventually lead to errors in pcm data. This patch adds overflow and underflow interrupt support for SSI BUSIF buffer. Reported-by: Chen Li <licheng0822(a)thundersoft.com> Signed-off-by: Yongbo Zhang <giraffesnn123(a)gmail.com> Tested-by: Chen Li <licheng0822(a)thundersoft.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/20200512093003.28332-1-giraffesnn123@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/gen.c | 8 +++ sound/soc/sh/rcar/rsnd.h | 9 +++ sound/soc/sh/rcar/ssi.c | 145 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 162 insertions(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index af19010b9d88..8bd49c8a9517 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -224,6 +224,14 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SSI_SYS_STATUS5, 0x884), RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888), RSND_GEN_S_REG(SSI_SYS_STATUS7, 0x88c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c), RSND_GEN_S_REG(HDMI0_SEL, 0x9e0), RSND_GEN_S_REG(HDMI1_SEL, 0x9e4), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ea6cbaa9743e..d47608ff5fac 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -189,6 +189,14 @@ enum rsnd_reg { SSI_SYS_STATUS5, SSI_SYS_STATUS6, SSI_SYS_STATUS7, + SSI_SYS_INT_ENABLE0, + SSI_SYS_INT_ENABLE1, + SSI_SYS_INT_ENABLE2, + SSI_SYS_INT_ENABLE3, + SSI_SYS_INT_ENABLE4, + SSI_SYS_INT_ENABLE5, + SSI_SYS_INT_ENABLE6, + SSI_SYS_INT_ENABLE7, HDMI0_SEL, HDMI1_SEL, SSI9_BUSIF0_MODE, @@ -237,6 +245,7 @@ enum rsnd_reg { #define SSI9_BUSIF_ADINR(i) (SSI9_BUSIF0_ADINR + (i)) #define SSI9_BUSIF_DALIGN(i) (SSI9_BUSIF0_DALIGN + (i)) #define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) +#define SSI_SYS_INT_ENABLE(i) (SSI_SYS_INT_ENABLE0 + (i)) struct rsnd_priv; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4a7d3413917f..47d5ddb526f2 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -372,6 +372,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, u32 wsr = ssi->wsr; int width; int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + int i; + u32 sys_int_enable = 0; is_tdm = rsnd_runtime_is_tdm(io); is_tdm_split = rsnd_runtime_is_tdm_split(io); @@ -447,6 +450,38 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } + /* enable busif buffer over/under run interrupt. */ + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE(i * 2)); + sys_int_enable |= 0xf << (id * 4); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE(i * 2), + sys_int_enable); + } + + break; + case 9: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1)); + sys_int_enable |= 0xf << 4; + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1), + sys_int_enable); + } + + break; + } + } + init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; @@ -496,6 +531,13 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct device *dev = rsnd_priv_to_dev(priv); + int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + int i; + u32 sys_int_enable = 0; + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); if (!rsnd_ssi_is_run_mods(mod, io)) return 0; @@ -517,6 +559,38 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, ssi->wsr = 0; } + /* disable busif buffer over/under run interrupt. */ + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE(i * 2)); + sys_int_enable &= ~(0xf << (id * 4)); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE(i * 2), + sys_int_enable); + } + + break; + case 9: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1)); + sys_int_enable &= ~(0xf << 4); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1), + sys_int_enable); + } + + break; + } + } + return 0; } @@ -622,6 +696,11 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, int enable) { u32 val = 0; + int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); if (rsnd_is_gen1(priv)) return 0; @@ -635,6 +714,19 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, if (enable) val = rsnd_ssi_is_dma_mode(mod) ? 0x0e000000 : 0x0f000000; + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + case 9: + val |= 0x0000ff00; + break; + } + } + rsnd_mod_write(mod, SSI_INT_ENABLE, val); return 0; @@ -651,6 +743,12 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, u32 status; bool elapsed = false; bool stop = false; + int id = rsnd_mod_id(mod); + int i; + int is_tdm, is_tdm_split; + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); spin_lock(&priv->lock); @@ -672,6 +770,53 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, stop = true; } + status = 0; + + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + status = rsnd_mod_read(mod, + SSI_SYS_STATUS(i * 2)); + status &= 0xf << (id * 4); + + if (status) { + rsnd_dbg_irq_status(dev, + "%s err status : 0x%08x\n", + rsnd_mod_name(mod), status); + rsnd_mod_write(mod, + SSI_SYS_STATUS(i * 2), + 0xf << (id * 4)); + stop = true; + break; + } + } + break; + case 9: + for (i = 0; i < 4; i++) { + status = rsnd_mod_read(mod, + SSI_SYS_STATUS((i * 2) + 1)); + status &= 0xf << 4; + + if (status) { + rsnd_dbg_irq_status(dev, + "%s err status : 0x%08x\n", + rsnd_mod_name(mod), status); + rsnd_mod_write(mod, + SSI_SYS_STATUS((i * 2) + 1), + 0xf << 4); + stop = true; + break; + } + } + break; + } + } + rsnd_ssi_status_clear(mod); rsnd_ssi_interrupt_out: spin_unlock(&priv->lock); -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 130/266] ASoC: SOF: core: fix error return code in sof_probe_continue()
by Sasha Levin 18 Jun '20

18 Jun '20

From: Wei Yongjun <weiyongjun1(a)huawei.com> [ Upstream commit 7d8785bc7adbb4dc5ba8ee06994107637848ded8 ] Fix to return negative error code -ENOMEM from the IPC init error handling case instead of 0, as done elsewhere in this function. Fixes: c16211d6226d ("ASoC: SOF: Add Sound Open Firmware driver core") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> Link: https://lore.kernel.org/r/20200509093337.78897-1-weiyongjun1@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 12aec140819a..2a6b84d2781e 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -372,6 +372,7 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) /* init the IPC */ sdev->ipc = snd_sof_ipc_init(sdev); if (!sdev->ipc) { + ret = -ENOMEM; dev_err(sdev->dev, "error: failed to init DSP IPC %d\n", ret); goto ipc_err; } -- 2.25.1

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[PATCH AUTOSEL 5.4 127/266] ALSA: firewire-lib: fix invalid assignment to union data for directional parameter
by Sasha Levin 18 Jun '20

18 Jun '20

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 8304cf77c92038cd1c50c27b69d30be695cc8003 ] Although the value of FDF is used just for outgoing stream, the assignment to union member is done for both directions of stream. At present this causes no issue because the value of same position is reassigned later for opposite stream. However, it's better to add if statement. Fixes: d3d10a4a1b19 ("ALSA: firewire-lib: use union for directional parameters") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20200508043635.349339-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/amdtp-am824.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c index 67d735e9a6a4..fea92e148790 100644 --- a/sound/firewire/amdtp-am824.c +++ b/sound/firewire/amdtp-am824.c @@ -82,7 +82,8 @@ int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate, if (err < 0) return err; - s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc; + if (s->direction == AMDTP_OUT_STREAM) + s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc; p->pcm_channels = pcm_channels; p->midi_ports = midi_ports; -- 2.25.1

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[PATCH AUTOSEL 5.4 114/266] soundwire: slave: don't init debugfs on device registration error
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 8893ab5e8ee5d7c12e0fc1dca4a309475064473d ] The error handling flow seems incorrect, there is no reason to try and add debugfs support if the device registration did not succeed. Return on error. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Link: https://lore.kernel.org/r/20200419185117.4233-2-yung-chuan.liao@linux.intel… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/slave.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 6473fa602f82..611f4f5bc36a 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -57,6 +57,8 @@ static int sdw_slave_add(struct sdw_bus *bus, list_del(&slave->node); mutex_unlock(&bus->bus_lock); put_device(&slave->dev); + + return ret; } sdw_slave_debugfs_init(slave); -- 2.25.1

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[PATCH AUTOSEL 5.4 113/266] ASoC: max98373: reorder max98373_reset() in resume
by Sasha Levin 18 Jun '20

18 Jun '20

From: Yong Zhi <yong.zhi(a)intel.com> [ Upstream commit 1a446873d7dd3a450f685928ce7f1907bde4583d ] During S3 test, the following error was observed: [ 726.174237] i2c_designware i2c_designware.0: platform_pm_resume+0x0/0x3d returned 0 after 0 usecs [ 726.184187] max98373 i2c-MX98373:00: calling max98373_resume+0x0/0x30 [snd_soc_max98373] @ 12698, parent: i2c-11 [ 726.195589] max98373 i2c-MX98373:00: Reset command failed. (ret:-16) When calling regmap_update_bits(), since map->reg_update_bits is NULL, _regmap_read() is entered with the following logic: if (!map->cache_bypass) { ret = regcache_read(map, reg, val); if (ret == 0) return 0; } if (map->cache_only) return -EBUSY; regcache_read() hits -EINVAL because MAX98373_R2000_SW_RESET is volatile, as map->cache_only is set by codec suspend, thus -EBUSY is returned. Fix by moving max98373_reset() after cache_only set to false in max98373_resume(). Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Link: https://lore.kernel.org/r/1588376661-29799-1-git-send-email-yong.zhi@intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98373.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index cae1def8902d..96718e3a1ad0 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -850,8 +850,8 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); + max98373_reset(max98373, dev); regcache_sync(max98373->regmap); return 0; } -- 2.25.1

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[PATCH AUTOSEL 5.4 106/266] slimbus: ngd: get drvdata from correct device
by Sasha Levin 18 Jun '20

18 Jun '20

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit b58c663059b484f7ff547d076a34cf6d7a302e56 ] Get drvdata directly from parent instead of ngd dev, as ngd dev can probe defer and previously set drvdata will become null. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20200417093618.7929-1-srinivas.kandagatla@linaro.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 29fbab55c3b3..01a17d84b606 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1354,7 +1354,6 @@ static int of_qcom_slim_ngd_register(struct device *parent, ngd->pdev->driver_override = QCOM_SLIM_NGD_DRV_NAME; ngd->pdev->dev.of_node = node; ctrl->ngd = ngd; - platform_set_drvdata(ngd->pdev, ctrl); platform_device_add(ngd->pdev); ngd->base = ctrl->base + ngd->id * data->offset + @@ -1369,12 +1368,13 @@ static int of_qcom_slim_ngd_register(struct device *parent, static int qcom_slim_ngd_probe(struct platform_device *pdev) { - struct qcom_slim_ngd_ctrl *ctrl = platform_get_drvdata(pdev); struct device *dev = &pdev->dev; + struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev->parent); int ret; ctrl->ctrl.dev = dev; + platform_set_drvdata(pdev, ctrl); pm_runtime_use_autosuspend(dev); pm_runtime_set_autosuspend_delay(dev, QCOM_SLIM_NGD_AUTOSUSPEND); pm_runtime_set_suspended(dev); -- 2.25.1

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[PATCH AUTOSEL 5.4 102/266] ALSA: usb-audio: Fix racy list management in output queue
by Sasha Levin 18 Jun '20

18 Jun '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5b6cc38f3f3f37109ce72b60bda215a5f6892c0b ] The linked list entry from FIFO is peeked at queue_pending_output_urbs() but the actual element pop-out is performed outside the spinlock, and it's potentially racy. Do delete the link at the right place inside the spinlock. Fixes: 8fdff6a319e7 ("ALSA: snd-usb: implement new endpoint streaming model") Link: https://lore.kernel.org/r/20200424074016.14301-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/endpoint.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index d8dc7cb56d43..50104f658ed4 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -346,17 +346,17 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) ep->next_packet_read_pos %= MAX_URBS; /* take URB out of FIFO */ - if (!list_empty(&ep->ready_playback_urbs)) + if (!list_empty(&ep->ready_playback_urbs)) { ctx = list_first_entry(&ep->ready_playback_urbs, struct snd_urb_ctx, ready_list); + list_del_init(&ctx->ready_list); + } } spin_unlock_irqrestore(&ep->lock, flags); if (ctx == NULL) return; - list_del_init(&ctx->ready_list); - /* copy over the length information */ for (i = 0; i < packet->packets; i++) ctx->packet_size[i] = packet->packet_size[i]; -- 2.25.1

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[PATCH AUTOSEL 5.4 101/266] ALSA: usb-audio: Improve frames size computation
by Sasha Levin 18 Jun '20

18 Jun '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ] For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 4 ++++ sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++----- sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 2 ++ 4 files changed, 45 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 395403a2d33f..820e564656ed 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -84,6 +84,10 @@ struct snd_usb_endpoint { dma_addr_t sync_dma; /* DMA address of syncbuf */ unsigned int pipe; /* the data i/o pipe */ + unsigned int framesize[2]; /* small/large frame sizes in samples */ + unsigned int sample_rem; /* remainder from division fs/fps */ + unsigned int sample_accum; /* sample accumulator */ + unsigned int fps; /* frames per second */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ int freqshift; /* how much to shift the feedback value to get Q16.16 */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 4a9a2f6ef5a4..d8dc7cb56d43 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -124,12 +124,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) /* * For streaming based on information derived from sync endpoints, - * prepare_outbound_urb_sizes() will call next_packet_size() to + * prepare_outbound_urb_sizes() will call slave_next_packet_size() to * determine the number of samples to be sent in the next packet. * - * For implicit feedback, next_packet_size() is unused. + * For implicit feedback, slave_next_packet_size() is unused. */ -int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -146,6 +146,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) return ret; } +/* + * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes() + * will call next_packet_size() to determine the number of samples to be + * sent in the next packet. + */ +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +{ + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + ep->sample_accum += ep->sample_rem; + if (ep->sample_accum >= ep->fps) { + ep->sample_accum -= ep->fps; + ret = ep->framesize[1]; + } else { + ret = ep->framesize[0]; + } + + return ret; +} + static void retire_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *urb_ctx) { @@ -190,6 +213,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep, if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); @@ -874,10 +899,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->maxpacksize = fmt->maxpacksize; ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); - if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) { ep->freqn = get_usb_full_speed_rate(rate); - else + ep->fps = 1000; + } else { ep->freqn = get_usb_high_speed_rate(rate); + ep->fps = 8000; + } + + ep->sample_rem = rate % ep->fps; + ep->framesize[0] = rate / ep->fps; + ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps; /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -936,6 +968,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; + ep->sample_accum = 0; snd_usb_endpoint_start_quirk(ep); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 63a39d4fa8d8..d23fa0a8c11b 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -28,6 +28,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index ad8f38380aa3..e52d129085c0 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1575,6 +1575,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); -- 2.25.1

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[PATCH AUTOSEL 5.4 079/266] ASoC: meson: add missing free_irq() in error path
by Sasha Levin 18 Jun '20

18 Jun '20

From: "Pavel Machek (CIP)" <pavel(a)denx.de> [ Upstream commit 3b8a299a58b2afce464ae11324b59dcf0f1d10a7 ] free_irq() is missing in case of error, fix that. Signed-off-by: Pavel Machek (CIP) <pavel(a)denx.de> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200606153103.GA17905@amd Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/axg-fifo.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index d286dff3171d..898ef1d5608f 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -244,7 +244,7 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) /* Enable pclk to access registers and clock the fifo ip */ ret = clk_prepare_enable(fifo->pclk); if (ret) - return ret; + goto free_irq; /* Setup status2 so it reports the memory pointer */ regmap_update_bits(fifo->map, FIFO_CTRL1, @@ -264,8 +264,14 @@ static int axg_fifo_pcm_open(struct snd_pcm_substream *ss) /* Take memory arbitror out of reset */ ret = reset_control_deassert(fifo->arb); if (ret) - clk_disable_unprepare(fifo->pclk); + goto free_clk; + + return 0; +free_clk: + clk_disable_unprepare(fifo->pclk); +free_irq: + free_irq(fifo->irq, ss); return ret; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 067/266] ASoC: qcom: q6asm-dai: kCFI fix
by Sasha Levin 18 Jun '20

18 Jun '20

From: John Stultz <john.stultz(a)linaro.org> [ Upstream commit a6b675a89e51a1cdad0481b809b7840d3f86e4b5 ] Fixes the following kCFI crash seen on db845c, caused by the function prototypes not matching the callback function prototype. [ 82.585661] Unable to handle kernel NULL pointer dereference at virtual address 0000000000000001 [ 82.595387] Mem abort info: [ 82.599463] ESR = 0x96000005 [ 82.602658] EC = 0x25: DABT (current EL), IL = 32 bits [ 82.608177] SET = 0, FnV = 0 [ 82.611829] EA = 0, S1PTW = 0 [ 82.615369] Data abort info: [ 82.618751] ISV = 0, ISS = 0x00000005 [ 82.622641] CM = 0, WnR = 0 [ 82.625774] user pgtable: 4k pages, 39-bit VAs, pgdp=0000000174259000 [ 82.632292] [0000000000000001] pgd=0000000000000000, pud=0000000000000000 [ 82.639167] Internal error: Oops: 96000005 [#1] PREEMPT SMP [ 82.644795] Modules linked in: hci_uart btqca xhci_plat_hcd xhci_pci_renesas xhci_pci xhci_hcd wcn36xx wcnss_ctrl wcd934x vctrl_regulator ufs_qcom syscon_reboot_e [ 82.644927] qcom_apcs_ipc_mailbox q6asm_dai q6routing q6asm q6afe_dai q6adm q6afe q6core q6dsp_common pm8941_pwrkey pm8916_wdt platform_mhu pinctrl_spmi_mpp pine [ 82.812982] CPU: 3 PID: 240 Comm: kworker/u16:4 Tainted: G W 5.6.0-rc7-mainline-00960-g0c34353d11b9-dirty #1 [ 82.824201] Hardware name: Thundercomm Dragonboard 845c (DT) [ 82.829937] Workqueue: qcom_apr_rx apr_rxwq [apr] [ 82.834698] pstate: 80c00005 (Nzcv daif +PAN +UAO) [ 82.839553] pc : __cfi_check_fail+0x4/0x1c [q6asm_dai] [ 82.844754] lr : __cfi_check+0x3a8/0x3b0 [q6asm_dai] [ 82.849767] sp : ffffffc0105f3c20 [ 82.853123] x29: ffffffc0105f3c30 x28: 0000000000000020 [ 82.858489] x27: ffffff80f4588400 x26: ffffff80f458ec94 [ 82.863854] x25: ffffff80f458ece8 x24: ffffffe3670c7000 [ 82.869220] x23: ffffff8094bb7b34 x22: ffffffe367137000 [ 82.874585] x21: bd07909b332eada6 x20: 0000000000000001 [ 82.879950] x19: ffffffe36713863c x18: ffffff80f8df4430 [ 82.885316] x17: 0000000000000001 x16: ffffffe39d15e660 [ 82.890681] x15: 0000000000000001 x14: 0000000000000027 [ 82.896047] x13: 0000000000000000 x12: ffffffe39e6465a0 [ 82.901413] x11: 0000000000000051 x10: 000000000000ffff [ 82.906779] x9 : 000ffffffe366c19 x8 : c3c5f18762d1ceef [ 82.912145] x7 : 0000000000000000 x6 : ffffffc010877698 [ 82.917511] x5 : ffffffc0105f3c00 x4 : 0000000000000000 [ 82.922877] x3 : 0000000000000000 x2 : 0000000000000001 [ 82.928243] x1 : ffffffe36713863c x0 : 0000000000000001 [ 82.933610] Call trace: [ 82.936099] __cfi_check_fail+0x4/0x1c [q6asm_dai] [ 82.940955] q6asm_srvc_callback+0x22c/0x618 [q6asm] [ 82.945973] apr_rxwq+0x1a8/0x27c [apr] [ 82.949861] process_one_work+0x2e8/0x54c [ 82.953919] worker_thread+0x27c/0x4d4 [ 82.957715] kthread+0x144/0x154 [ 82.960985] ret_from_fork+0x10/0x18 [ 82.964603] Code: a8c37bfd f85f8e5e d65f03c0 b40000a0 (39400008) [ 82.970762] ---[ end trace 410accb839617143 ]--- [ 82.975429] Kernel panic - not syncing: Fatal exception Signed-off-by: John Stultz <john.stultz(a)linaro.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Stephan Gerhold <stephan(a)gerhold.net> Cc: Sami Tolvanen <samitolvanen(a)google.com> Cc: Todd Kjos <tkjos(a)google.com> Cc: Alistair Delva <adelva(a)google.com> Cc: Amit Pundir <amit.pundir(a)linaro.org> Cc: Sumit Semwal <sumit.semwal(a)linaro.org> Cc: alsa-devel(a)alsa-project.org Link: https://lore.kernel.org/r/20200529213823.98812-1-john.stultz@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 548eb4fa2da6..9f0ffdcef637 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -171,7 +171,7 @@ static const struct snd_compr_codec_caps q6asm_compr_caps = { }; static void event_handler(uint32_t opcode, uint32_t token, - uint32_t *payload, void *priv) + void *payload, void *priv) { struct q6asm_dai_rtd *prtd = priv; struct snd_pcm_substream *substream = prtd->substream; @@ -494,7 +494,7 @@ static struct snd_pcm_ops q6asm_dai_ops = { }; static void compress_event_handler(uint32_t opcode, uint32_t token, - uint32_t *payload, void *priv) + void *payload, void *priv) { struct q6asm_dai_rtd *prtd = priv; struct snd_compr_stream *substream = prtd->cstream; -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 041/266] ASoC: SOF: Do nothing when DSP PM callbacks are not set
by Sasha Levin 18 Jun '20

18 Jun '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> [ Upstream commit c26fde3b15ed41f5f452f1da727795f787833287 ] This provides a better separation between runtime and PM sleep callbacks. Only do nothing if given runtime flag is set and calback is not set. With the current implementation, if PM sleep callback is set but runtime callback is not set then at runtime resume we reload the firmware even if we do not support runtime resume callback. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20200515135958.17511-2-kai.vehmanen@linux.intel.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/pm.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 195af259e78e..128680b09c20 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -266,7 +266,10 @@ static int sof_resume(struct device *dev, bool runtime_resume) int ret; /* do nothing if dsp resume callbacks are not set */ - if (!sof_ops(sdev)->resume || !sof_ops(sdev)->runtime_resume) + if (!runtime_resume && !sof_ops(sdev)->resume) + return 0; + + if (runtime_resume && !sof_ops(sdev)->runtime_resume) return 0; /* DSP was never successfully started, nothing to resume */ @@ -346,7 +349,10 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) int ret; /* do nothing if dsp suspend callback is not set */ - if (!sof_ops(sdev)->suspend) + if (!runtime_suspend && !sof_ops(sdev)->suspend) + return 0; + + if (runtime_suspend && !sof_ops(sdev)->runtime_suspend) return 0; if (sdev->fw_state != SOF_FW_BOOT_COMPLETE) -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 020/266] ALSA: isa/wavefront: prevent out of bounds write in ioctl
by Sasha Levin 18 Jun '20

18 Jun '20

From: Dan Carpenter <dan.carpenter(a)oracle.com> [ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ] The "header->number" comes from the ioctl and it needs to be clamped to prevent out of bounds writes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/wavefront/wavefront_synth.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index c5b1d5900eed..d6420d224d09 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1171,7 +1171,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) "alias for %d\n", header->number, header->hdr.a.OriginalSample); - + + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + munge_int32 (header->number, &alias_hdr[0], 2); munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2); munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset), @@ -1202,6 +1205,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) int num_samples; unsigned char *msample_hdr; + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL); if (! msample_hdr) return -ENOMEM; -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 019/266] ALSA: hda/realtek - Introduce polarity for micmute LED GPIO
by Sasha Levin 18 Jun '20

18 Jun '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit dbd13179780555ecd3c992dea1222ca31920e892 ] Currently mute LED and micmute LED share the same GPIO polarity. So split the polarity for mute and micmute, in case they have different polarities. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200430083255.5093-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index df5afac0b600..459a7d61326e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -81,6 +81,7 @@ struct alc_spec { /* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */ int mute_led_polarity; + int micmute_led_polarity; hda_nid_t mute_led_nid; hda_nid_t cap_mute_led_nid; @@ -4080,11 +4081,9 @@ static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec, /* update LED status via GPIO */ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, - bool enabled) + int polarity, bool enabled) { - struct alc_spec *spec = codec->spec; - - if (spec->mute_led_polarity) + if (polarity) enabled = !enabled; alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */ } @@ -4095,7 +4094,8 @@ static void alc_fixup_gpio_mute_hook(void *private_data, int enabled) struct hda_codec *codec = private_data; struct alc_spec *spec = codec->spec; - alc_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); + alc_update_gpio_led(codec, spec->gpio_mute_led_mask, + spec->mute_led_polarity, enabled); } /* turn on/off mic-mute LED via GPIO per capture hook */ @@ -4104,6 +4104,7 @@ static void alc_gpio_micmute_update(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_update_gpio_led(codec, spec->gpio_mic_led_mask, + spec->micmute_led_polarity, spec->gen.micmute_led.led_value); } @@ -5808,7 +5809,8 @@ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, snd_hda_gen_hp_automute(codec, jack); /* mute_led_polarity is set to 0, so we pass inverted value here */ - alc_update_gpio_led(codec, 0x10, !spec->gen.hp_jack_present); + alc_update_gpio_led(codec, 0x10, spec->mute_led_polarity, + !spec->gen.hp_jack_present); } /* Manage GPIOs for HP EliteBook Folio 9480m. -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 013/266] ASoC: davinci-mcasp: Fix dma_chan refcnt leak when getting dma type
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit a697ae6ea56e23397341b027098c1b11d9ab13da ] davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a reference of the specified dma_chan object to "chan" with increased refcnt. When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in one exception handling path of davinci_mcasp_get_dma_type(). When chan device is NULL, the function forgets to decrease the refcnt increased by dma_request_chan(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when chan device is NULL. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 7aa3c32e4a49..0541071f454b 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1875,8 +1875,10 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) PTR_ERR(chan)); return PTR_ERR(chan); } - if (WARN_ON(!chan->device || !chan->device->dev)) + if (WARN_ON(!chan->device || !chan->device->dev)) { + dma_release_channel(chan); return -EINVAL; + } if (chan->device->dev->of_node) ret = of_property_read_string(chan->device->dev->of_node, -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 010/266] ASoC: fsl_esai: Disable exception interrupt before scheduling tasklet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 1fecbb71fe0e46b886f84e3b6decca6643c3af6d ] Disable exception interrupt before scheduling tasklet, otherwise if the tasklet isn't handled immediately, there will be endless xrun interrupt. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/a8f2ad955aac9e52587beedc1133b3efbe746895.15879688… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7a49d03463a..84290be778f0 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -87,6 +87,10 @@ static irqreturn_t esai_isr(int irq, void *devid) if ((saisr & (ESAI_SAISR_TUE | ESAI_SAISR_ROE)) && esai_priv->reset_at_xrun) { dev_dbg(&pdev->dev, "reset module for xrun\n"); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xEIE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xEIE_MASK, 0); tasklet_schedule(&esai_priv->task); } -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 005/266] ASoC: SOF: imx8: Fix randbuild error
by Sasha Levin 18 Jun '20

18 Jun '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit fe17e6cdc0fefca96ba9659be4b2b07487cbf0c5 ] when do randconfig like this: CONFIG_SND_SOC_SOF_IMX8_SUPPORT=y CONFIG_SND_SOC_SOF_IMX8=y CONFIG_SND_SOC_SOF_OF=y CONFIG_IMX_DSP=m CONFIG_IMX_SCU=y there is a link error: sound/soc/sof/imx/imx8.o: In function 'imx8_send_msg': imx8.c:(.text+0x380): undefined reference to 'imx_dsp_ring_doorbell' Select IMX_DSP in SND_SOC_SOF_IMX8_SUPPORT to fix this Fixes: f9ad75468453 ("ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Link: https://lore.kernel.org/r/20200409071832.2039-2-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 71f318bc2c74..b4f0426685c4 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -14,7 +14,7 @@ if SND_SOC_SOF_IMX_TOPLEVEL config SND_SOC_SOF_IMX8_SUPPORT bool "SOF support for i.MX8" depends on IMX_SCU - depends on IMX_DSP + select IMX_DSP help This adds support for Sound Open Firmware for NXP i.MX8 platforms Say Y if you have such a device. -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 003/266] ASoC: tegra: tegra_wm8903: Support nvidia, headset property
by Sasha Levin 18 Jun '20

18 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 6211dfda2195..0fa01cacfec9 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -159,6 +159,7 @@ 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; @@ -171,12 +172,15 @@ 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

1 0

[PATCH AUTOSEL 5.7 366/388] ASoC: rt5645: Add platform-data for Asus T101HA
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 79d4f823a06796656289f97b922493da5690e46c ] The Asus T101HA uses the default jack-detect mode 3, but instead of using an analog microphone it is using a DMIC on dmic-data-pin 1, like the Asus T100HA. Note unlike the T100HA its jack-detect is not inverted. Add a DMI quirk with the correct settings for this model. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608204634.93407-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 6ba1849a77b0..e2e1d5b03b38 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3625,6 +3625,12 @@ static const struct rt5645_platform_data asus_t100ha_platform_data = { .inv_jd1_1 = true, }; +static const struct rt5645_platform_data asus_t101ha_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_IN2N, + .dmic2_data_pin = RT5645_DMIC2_DISABLE, + .jd_mode = 3, +}; + static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = { .jd_mode = 3, .in2_diff = true, @@ -3708,6 +3714,14 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&asus_t100ha_platform_data, }, + { + .ident = "ASUS T101HA", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "T101HA"), + }, + .driver_data = (void *)&asus_t101ha_platform_data, + }, { .ident = "MINIX Z83-4", .matches = { -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 365/388] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT10-A tablet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 199a5e8fda54ab3c8c6f6bf980c004e97ebf5ccb ] The Toshiba Encore WT10-A tablet almost fully works with the default settings for Bay Trail CR devices. The only issue is that it uses a digital mic. connected the the DMIC1 input instead of an analog mic. Add a quirk for this model using the default settings with the input-map replaced with BYT_RT5640_DMIC1_MAP. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608204634.93407-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index fbfd53874b47..5c1a5e2aff6f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -754,6 +754,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_JD_NOT_INV | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT10-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT10-A-103"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF2 | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 364/388] ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit ba4e5abc6c4e173af7c941c03c067263b686665d ] With additional checks on dailinks, we see errors such as [ 3.000418] sof-nocodec sof-nocodec: CPU DAI DMIC01 Pin for rtd NoCodec-6 does not support playback It's not clear why we set the dpcm_playback and dpcm_capture flags unconditionally, add a check on number of channels for each direction to avoid invalid configurations. Fixes: 8017b8fd37bf5e ('ASoC: SOF: Add Nocodec machine driver support') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608194415.4663-5-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/nocodec.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c index 2233146386cc..71cf5f9db79d 100644 --- a/sound/soc/sof/nocodec.c +++ b/sound/soc/sof/nocodec.c @@ -52,8 +52,10 @@ static int sof_nocodec_bes_setup(struct device *dev, links[i].platforms->name = dev_name(dev); links[i].codecs->dai_name = "snd-soc-dummy-dai"; links[i].codecs->name = "snd-soc-dummy"; - links[i].dpcm_playback = 1; - links[i].dpcm_capture = 1; + if (ops->drv[i].playback.channels_min) + links[i].dpcm_playback = 1; + if (ops->drv[i].capture.channels_min) + links[i].dpcm_capture = 1; } card->dai_link = links; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 363/388] ASoC: core: only convert non DPCM link to DPCM link
by Sasha Levin 18 Jun '20

18 Jun '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 607fa205a7e4dfad28b8a67ab1c985756ddbccb0 ] Additional checks for valid DAIs expose a corner case, where existing BE dailinks get modified, e.g. HDMI links are tagged with dpcm_capture=1 even if the DAIs are for playback. This patch makes those changes conditional and flags configuration issues when a BE dailink is has no_pcm=0 but dpcm_playback or dpcm_capture=1 (which makes no sense). As discussed on the alsa-devel mailing list, there are redundant flags for dpcm_playback, dpcm_capture, playback_only, capture_only. This will have to be cleaned-up in a future update. For now only correct and flag problematic configurations. Fixes: 218fe9b7ec7f3 ("ASoC: soc-core: Set dpcm_playback / dpcm_capture") Suggested-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200608194415.4663-3-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-core.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 843b8b1c89d4..e5433e8fcf19 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1720,9 +1720,25 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) dai_link->platforms->name = component->name; /* convert non BE into BE */ - dai_link->no_pcm = 1; - dai_link->dpcm_playback = 1; - dai_link->dpcm_capture = 1; + if (!dai_link->no_pcm) { + dai_link->no_pcm = 1; + + if (dai_link->dpcm_playback) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_playback=1\n", + dai_link->name); + if (dai_link->dpcm_capture) + dev_warn(card->dev, + "invalid configuration, dailink %s has flags no_pcm=0 and dpcm_capture=1\n", + dai_link->name); + + /* convert normal link into DPCM one */ + if (!(dai_link->dpcm_playback || + dai_link->dpcm_capture)) { + dai_link->dpcm_playback = !dai_link->capture_only; + dai_link->dpcm_capture = !dai_link->playback_only; + } + } /* override any BE fixups */ dai_link->be_hw_params_fixup = -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 362/388] ASoC: soc-pcm: dpcm: fix playback/capture checks
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit b73287f0b0745961b14e5ebcce92cc8ed24d4d52 ] Recent changes in the ASoC core prevent multi-cpu BE dailinks from being used. DPCM does support multi-cpu DAIs for BE Dailinks, but not for FE. Handle the FE checks first, and make sure all DAIs support the same capabilities within the same dailink. Fixes: 9b5db059366ae2 ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> BugLink: https://github.com/thesofproject/linux/issues/2031 Link: https://lore.kernel.org/r/20200608194415.4663-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-pcm.c | 44 ++++++++++++++++++++++++++++++++++---------- 1 file changed, 34 insertions(+), 10 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1f302de44052..39ce61c5b874 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2908,20 +2908,44 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + int stream; int i; + if (rtd->dai_link->dynamic && rtd->num_cpus > 1) { + dev_err(rtd->dev, + "DPCM doesn't support Multi CPU for Front-Ends yet\n"); + return -EINVAL; + } + if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { - cpu_dai = asoc_rtd_to_cpu(rtd, 0); - if (rtd->num_cpus > 1) { - dev_err(rtd->dev, - "DPCM doesn't support Multi CPU yet\n"); - return -EINVAL; + if (rtd->dai_link->dpcm_playback) { + stream = SNDRV_PCM_STREAM_PLAYBACK; + + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + if (!snd_soc_dai_stream_valid(cpu_dai, + stream)) { + dev_err(rtd->card->dev, + "CPU DAI %s for rtd %s does not support playback\n", + cpu_dai->name, + rtd->dai_link->stream_name); + return -EINVAL; + } + playback = 1; + } + if (rtd->dai_link->dpcm_capture) { + stream = SNDRV_PCM_STREAM_CAPTURE; + + for_each_rtd_cpu_dais(rtd, i, cpu_dai) + if (!snd_soc_dai_stream_valid(cpu_dai, + stream)) { + dev_err(rtd->card->dev, + "CPU DAI %s for rtd %s does not support capture\n", + cpu_dai->name, + rtd->dai_link->stream_name); + return -EINVAL; + } + capture = 1; } - - playback = rtd->dai_link->dpcm_playback && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK); - capture = rtd->dai_link->dpcm_capture && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE); } else { /* Adapt stream for codec2codec links */ int cpu_capture = rtd->dai_link->params ? -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 307/388] ASoC: SOF: Intel: hda: fix generic hda codec support
by Sasha Levin 18 Jun '20

18 Jun '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 89d73ccab20a684d8446cea4d8ac6a2608c8d390 ] Add support for using generic codec driver with SOF. Generic driver is used if: - snd_sof_intel_hda_common.hda_model="generic" is set, or - fallback if no other codec driver is found The implementation is aligned with snd-hda-intel driver, and fixes audio support for systems like Acer Swift 3 SF314-57G, on which this issue was originally reported. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Hui Wang <hui.wang(a)canonical.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> BugLink: https://github.com/thesofproject/linux/issues/1807 BugLink: https://bugs.launchpad.net/bugs/1877757 Link: https://lore.kernel.org/r/20200529160358.12134-1-kai.vehmanen@linux.intel.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-codec.c | 51 +++++++++++++++++++++++++++++---- 1 file changed, 45 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 3041fbbb010a..ea021db697b8 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -24,19 +24,44 @@ #define IDISP_VID_INTEL 0x80860000 /* load the legacy HDA codec driver */ -static int hda_codec_load_module(struct hda_codec *codec) +static int request_codec_module(struct hda_codec *codec) { #ifdef MODULE char alias[MODULE_NAME_LEN]; - const char *module = alias; + const char *mod = NULL; - snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias)); - dev_dbg(&codec->core.dev, "loading codec module: %s\n", module); - request_module(module); + switch (codec->probe_id) { + case HDA_CODEC_ID_GENERIC: +#if IS_MODULE(CONFIG_SND_HDA_GENERIC) + mod = "snd-hda-codec-generic"; #endif + break; + default: + snd_hdac_codec_modalias(&codec->core, alias, sizeof(alias)); + mod = alias; + break; + } + + if (mod) { + dev_dbg(&codec->core.dev, "loading codec module: %s\n", mod); + request_module(mod); + } +#endif /* MODULE */ return device_attach(hda_codec_dev(codec)); } +static int hda_codec_load_module(struct hda_codec *codec) +{ + int ret = request_codec_module(codec); + + if (ret <= 0) { + codec->probe_id = HDA_CODEC_ID_GENERIC; + ret = request_codec_module(codec); + } + + return ret; +} + /* enable controller wake up event for all codecs with jack connectors */ void hda_codec_jack_wake_enable(struct snd_sof_dev *sdev) { @@ -78,6 +103,13 @@ void hda_codec_jack_check(struct snd_sof_dev *sdev) {} EXPORT_SYMBOL_NS(hda_codec_jack_wake_enable, SND_SOC_SOF_HDA_AUDIO_CODEC); EXPORT_SYMBOL_NS(hda_codec_jack_check, SND_SOC_SOF_HDA_AUDIO_CODEC); +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) +#define is_generic_config(bus) \ + ((bus)->modelname && !strcmp((bus)->modelname, "generic")) +#else +#define is_generic_config(x) 0 +#endif + /* probe individual codec */ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, bool hda_codec_use_common_hdmi) @@ -87,6 +119,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, #endif struct hda_bus *hbus = sof_to_hbus(sdev); struct hdac_device *hdev; + struct hda_codec *codec; u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; u32 resp = -1; @@ -108,6 +141,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, hda_priv->codec.bus = hbus; hdev = &hda_priv->codec.core; + codec = &hda_priv->codec; ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev); if (ret < 0) @@ -122,6 +156,11 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, hda_priv->need_display_power = true; } + if (is_generic_config(hbus)) + codec->probe_id = HDA_CODEC_ID_GENERIC; + else + codec->probe_id = 0; + /* * if common HDMI codec driver is not used, codec load * is skipped here and hdac_hdmi is used instead @@ -129,7 +168,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (hda_codec_use_common_hdmi || (resp & 0xFFFF0000) != IDISP_VID_INTEL) { hdev->type = HDA_DEV_LEGACY; - ret = hda_codec_load_module(&hda_priv->codec); + ret = hda_codec_load_module(codec); /* * handle ret==0 (no driver bound) as an error, but pass * other return codes without modification -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 306/388] ASoC: fsl_asrc_dma: Fix dma_chan leak when config DMA channel failed
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ] fsl_asrc_dma_hw_params() invokes dma_request_channel() or fsl_asrc_get_dma_channel(), which returns a reference of the specified dma_chan object to "pair->dma_chan[dir]" with increased refcnt. The reference counting issue happens in one exception handling path of fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End, the function forgets to decrease the refcnt increased by dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when config DMA channel failed. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_asrc_dma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index e7178817d7a7..1ee10eafe3e6 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -252,6 +252,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, ret = dmaengine_slave_config(pair->dma_chan[dir], &config_be); if (ret) { dev_err(dev, "failed to config DMA channel for Back-End\n"); + dma_release_channel(pair->dma_chan[dir]); return ret; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 294/388] ASoC: dapm: Move dai_link widgets to runtime to fix use after free
by Sasha Levin 18 Jun '20

18 Jun '20

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit f4aa5e214eeaf7f1c7f157526a5aa29784cb6a1f ] The newly added CODEC to CODEC DAI link widget pointers in snd_soc_dai_link are better placed in snd_soc_pcm_runtime. snd_soc_dai_link is really intended for static configuration of the DAI, and the runtime for dynamic data. The snd_soc_dai_link structures are not destroyed if the card is unbound. The widgets are cleared up on unbind, however if the card is rebound as the snd_soc_dai_link structures are reused these pointers will be left at their old values, causing access to freed memory. Fixes: 595571cca4de ("ASoC: dapm: Fix regression introducing multiple copies of DAI widgets") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200526161930.30759-1-ckeepax@opensource.cirrus.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc.h | 6 +++--- sound/soc/soc-dapm.c | 12 ++++++------ 2 files changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e0371e70242d..8e480efeda2a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -790,9 +790,6 @@ struct snd_soc_dai_link { const struct snd_soc_pcm_stream *params; unsigned int num_params; - struct snd_soc_dapm_widget *playback_widget; - struct snd_soc_dapm_widget *capture_widget; - unsigned int dai_fmt; /* format to set on init */ enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ @@ -1156,6 +1153,9 @@ struct snd_soc_pcm_runtime { struct snd_soc_dai **cpu_dais; unsigned int num_cpus; + struct snd_soc_dapm_widget *playback_widget; + struct snd_soc_dapm_widget *capture_widget; + struct delayed_work delayed_work; void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd); #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e2632841b321..c0aa64ff8e32 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4340,16 +4340,16 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card, codec = codec_dai->playback_widget; if (playback_cpu && codec) { - if (dai_link->params && !dai_link->playback_widget) { + if (dai_link->params && !rtd->playback_widget) { substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; dai = snd_soc_dapm_new_dai(card, substream, "playback"); if (IS_ERR(dai)) goto capture; - dai_link->playback_widget = dai; + rtd->playback_widget = dai; } dapm_connect_dai_routes(&card->dapm, cpu_dai, playback_cpu, - dai_link->playback_widget, + rtd->playback_widget, codec_dai, codec); } @@ -4358,16 +4358,16 @@ static void dapm_connect_dai_pair(struct snd_soc_card *card, codec = codec_dai->capture_widget; if (codec && capture_cpu) { - if (dai_link->params && !dai_link->capture_widget) { + if (dai_link->params && !rtd->capture_widget) { substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; dai = snd_soc_dapm_new_dai(card, substream, "capture"); if (IS_ERR(dai)) return; - dai_link->capture_widget = dai; + rtd->capture_widget = dai; } dapm_connect_dai_routes(&card->dapm, codec_dai, codec, - dai_link->capture_widget, + rtd->capture_widget, cpu_dai, capture_cpu); } } -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 279/388] ASoC: fix incomplete error-handling in img_i2s_in_probe.
by Sasha Levin 18 Jun '20

18 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> [ Upstream commit 25bf943e4e7b47282bd86ae7d39e039217ebb007 ] Function "pm_runtime_get_sync()" is not handled by "pm_runtime_put()" if "PTR_ERR(rst) == -EPROBE_DEFER". Fix this issue by adding "pm_runtime_put()" into this error path. Fixes: f65bb92ca12e ("ASoC: img-i2s-in: Add runtime PM") Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> Link: https://lore.kernel.org/r/20200525055011.31925-1-wu000273@umn.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/img/img-i2s-in.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index a495d1050d49..e30b66b94bf6 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -482,6 +482,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) if (IS_ERR(rst)) { if (PTR_ERR(rst) == -EPROBE_DEFER) { ret = -EPROBE_DEFER; + pm_runtime_put(&pdev->dev); goto err_suspend; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 232/388] ASoC: Intel: bytcr_rt5640: Add quirk for Toshiba Encore WT8-A tablet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 0e0e10fde0e9808d1991268f5dca69fb36c025f7 ] The Toshiba Encore WT8-A tablet almost fully works with the default settings for non-CR Bay Trail devices. The only problem is that its jack-detect switch is not inverted (it is active high instead of the normal active low). Add a quirk for this model using the default settings + BYT_RT5640_JD_NOT_INV. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200518072416.5348-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 08f4ae964b02..fbfd53874b47 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -742,6 +742,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* Toshiba Encore WT8-A */ + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "TOSHIBA"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "TOSHIBA WT8-A"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_MCLK_EN), + }, { /* Catch-all for generic Insyde tablets, must be last */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Insyde"), -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 217/388] ASoC: codecs: rt*-sdw: fix memory leak in set_sdw_stream()
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 07b542fe831cbefce163ad1b3aa7292c8a6332b8 ] Now that the sdw_stream is allocated in machine driver, set_sdw_stream() is also called with a NULL argument during the dailink shutdown. In this case, the drivers should not allocate any memory, and just return. Detected with KASAN/kmemleak. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> Cc: Shuming Fan <shumingf(a)realtek.com> Cc: Jack Yu <jack.yu(a)realtek.com> Link: https://lore.kernel.org/r/20200515211531.11416-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt1308-sdw.c | 3 +++ sound/soc/codecs/rt5682.c | 3 +++ sound/soc/codecs/rt700.c | 3 +++ sound/soc/codecs/rt711.c | 3 +++ sound/soc/codecs/rt715.c | 3 +++ 5 files changed, 15 insertions(+) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index a5a7e46de246..a7f45191364d 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -482,6 +482,9 @@ static int rt1308_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, { struct sdw_stream_data *stream; + if (!sdw_stream) + return 0; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index d36f560ad7a8..c4892af14850 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2958,6 +2958,9 @@ static int rt5682_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, { struct sdw_stream_data *stream; + if (!sdw_stream) + return 0; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index ff68f0e4f629..687ac2153666 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -860,6 +860,9 @@ static int rt700_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, { struct sdw_stream_data *stream; + if (!sdw_stream) + return 0; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 2daed7692a3b..65b59dbfb43c 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -906,6 +906,9 @@ static int rt711_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, { struct sdw_stream_data *stream; + if (!sdw_stream) + return 0; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 2cbc57b16b13..099c8bd20006 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -530,6 +530,9 @@ static int rt715_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, struct sdw_stream_data *stream; + if (!sdw_stream) + return 0; + stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 210/388] ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback
by Sasha Levin 18 Jun '20

18 Jun '20

From: Erwin Burema <e.burema(a)gmail.com> [ Upstream commit 10ce77e4817fef99e1166be7e6685a80c63bf77f ] For USB sound devices using implicit feedback the endpoint used for this feedback should be able to be opened twice, once for required feedback and second time for audio data. This way these devices can be put in duplex audio mode. Since this only works if the settings of the endpoint don't change a check is included for this. This fixes bug 207023 ("MOTU M2 regression on duplex audio") and should also fix bug 103751 ("M-Audio Fast Track Ultra usb audio device will not operate full-duplex") Fixes: c249177944b6 ("ALSA: usb-audio: add implicit fb quirk for MOTU M Series") Signed-off-by: Erwin Burema <e.burema(a)gmail.com> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207023 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=103751 Link: https://lore.kernel.org/r/2410739.SCZni40SNb@alpha-wolf Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 1 + sound/usb/endpoint.c | 195 ++++++++++++++++++++++++++++++++++++++++++- sound/usb/pcm.c | 5 ++ 3 files changed, 197 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 820e564656ed..d6219fba9699 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -108,6 +108,7 @@ struct snd_usb_endpoint { int iface, altsetting; int skip_packets; /* quirks for devices to ignore the first n packets in a stream */ + bool is_implicit_feedback; /* This endpoint is used as implicit feedback */ spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 50104f658ed4..9bea7d3f99f8 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -522,6 +522,8 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, list_add_tail(&ep->list, &chip->ep_list); + ep->is_implicit_feedback = 0; + __exit_unlock: mutex_unlock(&chip->mutex); @@ -621,6 +623,178 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force) ep->nurbs = 0; } +/* + * Check data endpoint for format differences + */ +static bool check_ep_params(struct snd_usb_endpoint *ep, + snd_pcm_format_t pcm_format, + unsigned int channels, + unsigned int period_bytes, + unsigned int frames_per_period, + unsigned int periods_per_buffer, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep) +{ + unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb; + unsigned int max_packs_per_period, urbs_per_period, urb_packs; + unsigned int max_urbs; + int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels; + int tx_length_quirk = (ep->chip->tx_length_quirk && + usb_pipeout(ep->pipe)); + bool ret = 1; + + if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) { + /* + * When operating in DSD DOP mode, the size of a sample frame + * in hardware differs from the actual physical format width + * because we need to make room for the DOP markers. + */ + frame_bits += channels << 3; + } + + ret = ret && (ep->datainterval == fmt->datainterval); + ret = ret && (ep->stride == frame_bits >> 3); + + switch (pcm_format) { + case SNDRV_PCM_FORMAT_U8: + ret = ret && (ep->silence_value == 0x80); + break; + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + ret = ret && (ep->silence_value == 0x69); + break; + default: + ret = ret && (ep->silence_value == 0); + } + + /* assume max. frequency is 50% higher than nominal */ + ret = ret && (ep->freqmax == ep->freqn + (ep->freqn >> 1)); + /* Round up freqmax to nearest integer in order to calculate maximum + * packet size, which must represent a whole number of frames. + * This is accomplished by adding 0x0.ffff before converting the + * Q16.16 format into integer. + * In order to accurately calculate the maximum packet size when + * the data interval is more than 1 (i.e. ep->datainterval > 0), + * multiply by the data interval prior to rounding. For instance, + * a freqmax of 41 kHz will result in a max packet size of 6 (5.125) + * frames with a data interval of 1, but 11 (10.25) frames with a + * data interval of 2. + * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the + * maximum datainterval value of 3, at USB full speed, higher for + * USB high speed, noting that ep->freqmax is in units of + * frames per packet in Q16.16 format.) + */ + maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * + (frame_bits >> 3); + if (tx_length_quirk) + maxsize += sizeof(__le32); /* Space for length descriptor */ + /* but wMaxPacketSize might reduce this */ + if (ep->maxpacksize && ep->maxpacksize < maxsize) { + /* whatever fits into a max. size packet */ + unsigned int data_maxsize = maxsize = ep->maxpacksize; + + if (tx_length_quirk) + /* Need to remove the length descriptor to calc freq */ + data_maxsize -= sizeof(__le32); + ret = ret && (ep->freqmax == (data_maxsize / (frame_bits >> 3)) + << (16 - ep->datainterval)); + } + + if (ep->fill_max) + ret = ret && (ep->curpacksize == ep->maxpacksize); + else + ret = ret && (ep->curpacksize == maxsize); + + if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) { + packs_per_ms = 8 >> ep->datainterval; + max_packs_per_urb = MAX_PACKS_HS; + } else { + packs_per_ms = 1; + max_packs_per_urb = MAX_PACKS; + } + if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep)) + max_packs_per_urb = min(max_packs_per_urb, + 1U << sync_ep->syncinterval); + max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval); + + /* + * Capture endpoints need to use small URBs because there's no way + * to tell in advance where the next period will end, and we don't + * want the next URB to complete much after the period ends. + * + * Playback endpoints with implicit sync much use the same parameters + * as their corresponding capture endpoint. + */ + if (usb_pipein(ep->pipe) || + snd_usb_endpoint_implicit_feedback_sink(ep)) { + + urb_packs = packs_per_ms; + /* + * Wireless devices can poll at a max rate of once per 4ms. + * For dataintervals less than 5, increase the packet count to + * allow the host controller to use bursting to fill in the + * gaps. + */ + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) { + int interval = ep->datainterval; + + while (interval < 5) { + urb_packs <<= 1; + ++interval; + } + } + /* make capture URBs <= 1 ms and smaller than a period */ + urb_packs = min(max_packs_per_urb, urb_packs); + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + ret = ret && (ep->nurbs == MAX_URBS); + + /* + * Playback endpoints without implicit sync are adjusted so that + * a period fits as evenly as possible in the smallest number of + * URBs. The total number of URBs is adjusted to the size of the + * ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits. + */ + } else { + /* determine how small a packet can be */ + minsize = (ep->freqn >> (16 - ep->datainterval)) * + (frame_bits >> 3); + /* with sync from device, assume it can be 12% lower */ + if (sync_ep) + minsize -= minsize >> 3; + minsize = max(minsize, 1u); + + /* how many packets will contain an entire ALSA period? */ + max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize); + + /* how many URBs will contain a period? */ + urbs_per_period = DIV_ROUND_UP(max_packs_per_period, + max_packs_per_urb); + /* how many packets are needed in each URB? */ + urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period); + + /* limit the number of frames in a single URB */ + ret = ret && (ep->max_urb_frames == + DIV_ROUND_UP(frames_per_period, urbs_per_period)); + + /* try to use enough URBs to contain an entire ALSA buffer */ + max_urbs = min((unsigned) MAX_URBS, + MAX_QUEUE * packs_per_ms / urb_packs); + ret = ret && (ep->nurbs == min(max_urbs, + urbs_per_period * periods_per_buffer)); + } + + ret = ret && (ep->datainterval == fmt->datainterval); + ret = ret && (ep->maxpacksize == fmt->maxpacksize); + ret = ret && + (ep->fill_max == !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX)); + + return ret; +} + /* * configure a data endpoint */ @@ -886,10 +1060,23 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, int err; if (ep->use_count != 0) { - usb_audio_warn(ep->chip, - "Unable to change format on ep #%x: already in use\n", - ep->ep_num); - return -EBUSY; + bool check = ep->is_implicit_feedback && + check_ep_params(ep, pcm_format, + channels, period_bytes, + period_frames, buffer_periods, + fmt, sync_ep); + + if (!check) { + usb_audio_warn(ep->chip, + "Unable to change format on ep #%x: already in use\n", + ep->ep_num); + return -EBUSY; + } + + usb_audio_dbg(ep->chip, + "Ep #%x already in use as implicit feedback but format not changed\n", + ep->ep_num); + return 0; } /* release old buffers, if any */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b50965ab3b3a..d61c2f1095b5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -404,6 +404,8 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, if (!subs->sync_endpoint) return -EINVAL; + subs->sync_endpoint->is_implicit_feedback = 1; + subs->data_endpoint->sync_master = subs->sync_endpoint; return 1; @@ -502,12 +504,15 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, implicit_fb ? SND_USB_ENDPOINT_TYPE_DATA : SND_USB_ENDPOINT_TYPE_SYNC); + if (!subs->sync_endpoint) { if (is_playback && attr == USB_ENDPOINT_SYNC_NONE) return 0; return -EINVAL; } + subs->sync_endpoint->is_implicit_feedback = implicit_fb; + subs->data_endpoint->sync_master = subs->sync_endpoint; return 0; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 209/388] ALSA: usb-audio: fixing upper volume limit for RME Babyface Pro routing crosspoints
by Sasha Levin 18 Jun '20

18 Jun '20

From: Thomas Ebeling <penguins(a)bollie.de> [ Upstream commit 47b4f5f5b65680fbef7a7a9a4796b35f38a6e43e ] In my initial patch, these were set too low. Fixes: 3e8f3bd04716 ("ALSA: usb-audio: RME Babyface Pro mixer patch") Signed-off-by: Thomas Ebeling <penguins(a)bollie.de> Link: https://lore.kernel.org/r/20200515114556.vtspnonzvp4xp44m@bollie.ca9.eu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_quirks.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index bdff8674942e..aad2683ff793 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2191,7 +2191,7 @@ static int snd_rme_controls_create(struct usb_mixer_interface *mixer) * These devices exposes a couple of DSP functions via request to EP0. * Switches are available via control registers, while routing is controlled * by controlling the volume on each possible crossing point. - * Volume control is linear, from -inf (dec. 0) to +6dB (dec. 46341) with + * Volume control is linear, from -inf (dec. 0) to +6dB (dec. 65536) with * 0dB being at dec. 32768. */ enum { @@ -2220,7 +2220,7 @@ enum { #define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff #define SND_BBFPRO_MIXER_VAL_SHIFT 9 #define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf -#define SND_BBFPRO_MIXER_VAL_MAX 46341 // +6dB +#define SND_BBFPRO_MIXER_VAL_MAX 65536 // +6dB #define SND_BBFPRO_USBREQ_CTL_REG1 0x10 #define SND_BBFPRO_USBREQ_CTL_REG2 0x17 -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 195/388] ASoC: ti: omap-mcbsp: Fix an error handling path in 'asoc_mcbsp_probe()'
by Sasha Levin 18 Jun '20

18 Jun '20

From: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> [ Upstream commit 03990fd58d2b7c8f7d53e514ba9b8749fac260f9 ] If an error occurs after the call to 'omap_mcbsp_init()', the reference to 'mcbsp->fclk' must be decremented, as already done in the remove function. This can be achieved easily by using the devm_ variant of 'clk_get()' when the reference is taken in 'omap_mcbsp_init()' This fixes the leak in the probe and has the side effect to simplify both the error handling path of 'omap_mcbsp_init()' and the remove function. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> Acked-by: Peter Ujfalusi <peter.ujflausi(a)ti.com> Link: https://lore.kernel.org/r/20200512134325.252073-1-christophe.jaillet@wanado… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-mcbsp.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 3d41ca2238d4..4f33ddb7b441 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -686,7 +686,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, SNDRV_PCM_STREAM_CAPTURE); - mcbsp->fclk = clk_get(&pdev->dev, "fck"); + mcbsp->fclk = devm_clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { ret = PTR_ERR(mcbsp->fclk); dev_err(mcbsp->dev, "unable to get fck: %d\n", ret); @@ -711,7 +711,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); - goto err_thres; + return ret; } } @@ -724,8 +724,6 @@ static int omap_mcbsp_init(struct platform_device *pdev) err_st: if (mcbsp->pdata->buffer_size) sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); -err_thres: - clk_put(mcbsp->fclk); return ret; } @@ -1442,8 +1440,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) omap_mcbsp_st_cleanup(pdev); - clk_put(mcbsp->fclk); - return 0; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 194/388] ASoC: ux500: mop500: Fix some refcounted resources issues
by Sasha Levin 18 Jun '20

18 Jun '20

From: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> [ Upstream commit 4e8748fcaeec073e3ba794871ce86c545e4f961f ] There are 2 issues here: - if one of the 'of_parse_phandle' fails, calling 'mop500_of_node_put()' is a no-op because the 'mop500_dai_links' structure has not been initialized yet, so the referenced are not decremented - The reference stored in 'mop500_dai_links[i].codecs' is refcounted only once in the probe and must be decremented only once. Fixes: 39013bd60e79 ("ASoC: Ux500: Dispose of device nodes correctly") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> Link: https://lore.kernel.org/r/20200512100705.246349-1-christophe.jaillet@wanado… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ux500/mop500.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 2873e8e6f02b..cdae1190b930 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -63,10 +63,11 @@ static void mop500_of_node_put(void) { int i; - for (i = 0; i < 2; i++) { + for (i = 0; i < 2; i++) of_node_put(mop500_dai_links[i].cpus->of_node); - of_node_put(mop500_dai_links[i].codecs->of_node); - } + + /* Both links use the same codec, which is refcounted only once */ + of_node_put(mop500_dai_links[0].codecs->of_node); } static int mop500_of_probe(struct platform_device *pdev, @@ -81,7 +82,9 @@ static int mop500_of_probe(struct platform_device *pdev, if (!(msp_np[0] && msp_np[1] && codec_np)) { dev_err(&pdev->dev, "Phandle missing or invalid\n"); - mop500_of_node_put(); + for (i = 0; i < 2; i++) + of_node_put(msp_np[i]); + of_node_put(codec_np); return -EINVAL; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 193/388] SoC: rsnd: add interrupt support for SSI BUSIF buffer
by Sasha Levin 18 Jun '20

18 Jun '20

From: Yongbo Zhang <giraffesnn123(a)gmail.com> [ Upstream commit 66c705d07d784fb6b4622c6e47b6acae357472db ] SSI BUSIF buffer is possible to overflow or underflow, especially in a hypervisor environment. If there is no interrupt support, it will eventually lead to errors in pcm data. This patch adds overflow and underflow interrupt support for SSI BUSIF buffer. Reported-by: Chen Li <licheng0822(a)thundersoft.com> Signed-off-by: Yongbo Zhang <giraffesnn123(a)gmail.com> Tested-by: Chen Li <licheng0822(a)thundersoft.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/20200512093003.28332-1-giraffesnn123@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/gen.c | 8 +++ sound/soc/sh/rcar/rsnd.h | 9 +++ sound/soc/sh/rcar/ssi.c | 145 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 162 insertions(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index af19010b9d88..8bd49c8a9517 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -224,6 +224,14 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SSI_SYS_STATUS5, 0x884), RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888), RSND_GEN_S_REG(SSI_SYS_STATUS7, 0x88c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c), RSND_GEN_S_REG(HDMI0_SEL, 0x9e0), RSND_GEN_S_REG(HDMI1_SEL, 0x9e4), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ea6cbaa9743e..d47608ff5fac 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -189,6 +189,14 @@ enum rsnd_reg { SSI_SYS_STATUS5, SSI_SYS_STATUS6, SSI_SYS_STATUS7, + SSI_SYS_INT_ENABLE0, + SSI_SYS_INT_ENABLE1, + SSI_SYS_INT_ENABLE2, + SSI_SYS_INT_ENABLE3, + SSI_SYS_INT_ENABLE4, + SSI_SYS_INT_ENABLE5, + SSI_SYS_INT_ENABLE6, + SSI_SYS_INT_ENABLE7, HDMI0_SEL, HDMI1_SEL, SSI9_BUSIF0_MODE, @@ -237,6 +245,7 @@ enum rsnd_reg { #define SSI9_BUSIF_ADINR(i) (SSI9_BUSIF0_ADINR + (i)) #define SSI9_BUSIF_DALIGN(i) (SSI9_BUSIF0_DALIGN + (i)) #define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) +#define SSI_SYS_INT_ENABLE(i) (SSI_SYS_INT_ENABLE0 + (i)) struct rsnd_priv; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4a7d3413917f..47d5ddb526f2 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -372,6 +372,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, u32 wsr = ssi->wsr; int width; int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + int i; + u32 sys_int_enable = 0; is_tdm = rsnd_runtime_is_tdm(io); is_tdm_split = rsnd_runtime_is_tdm_split(io); @@ -447,6 +450,38 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } + /* enable busif buffer over/under run interrupt. */ + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE(i * 2)); + sys_int_enable |= 0xf << (id * 4); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE(i * 2), + sys_int_enable); + } + + break; + case 9: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1)); + sys_int_enable |= 0xf << 4; + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1), + sys_int_enable); + } + + break; + } + } + init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; @@ -496,6 +531,13 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct device *dev = rsnd_priv_to_dev(priv); + int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + int i; + u32 sys_int_enable = 0; + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); if (!rsnd_ssi_is_run_mods(mod, io)) return 0; @@ -517,6 +559,38 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, ssi->wsr = 0; } + /* disable busif buffer over/under run interrupt. */ + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE(i * 2)); + sys_int_enable &= ~(0xf << (id * 4)); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE(i * 2), + sys_int_enable); + } + + break; + case 9: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1)); + sys_int_enable &= ~(0xf << 4); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1), + sys_int_enable); + } + + break; + } + } + return 0; } @@ -622,6 +696,11 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, int enable) { u32 val = 0; + int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); if (rsnd_is_gen1(priv)) return 0; @@ -635,6 +714,19 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, if (enable) val = rsnd_ssi_is_dma_mode(mod) ? 0x0e000000 : 0x0f000000; + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + case 9: + val |= 0x0000ff00; + break; + } + } + rsnd_mod_write(mod, SSI_INT_ENABLE, val); return 0; @@ -651,6 +743,12 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, u32 status; bool elapsed = false; bool stop = false; + int id = rsnd_mod_id(mod); + int i; + int is_tdm, is_tdm_split; + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); spin_lock(&priv->lock); @@ -672,6 +770,53 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, stop = true; } + status = 0; + + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + status = rsnd_mod_read(mod, + SSI_SYS_STATUS(i * 2)); + status &= 0xf << (id * 4); + + if (status) { + rsnd_dbg_irq_status(dev, + "%s err status : 0x%08x\n", + rsnd_mod_name(mod), status); + rsnd_mod_write(mod, + SSI_SYS_STATUS(i * 2), + 0xf << (id * 4)); + stop = true; + break; + } + } + break; + case 9: + for (i = 0; i < 4; i++) { + status = rsnd_mod_read(mod, + SSI_SYS_STATUS((i * 2) + 1)); + status &= 0xf << 4; + + if (status) { + rsnd_dbg_irq_status(dev, + "%s err status : 0x%08x\n", + rsnd_mod_name(mod), status); + rsnd_mod_write(mod, + SSI_SYS_STATUS((i * 2) + 1), + 0xf << 4); + stop = true; + break; + } + } + break; + } + } + rsnd_ssi_status_clear(mod); rsnd_ssi_interrupt_out: spin_unlock(&priv->lock); -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 183/388] ASoC: SOF: core: fix error return code in sof_probe_continue()
by Sasha Levin 18 Jun '20

18 Jun '20

From: Wei Yongjun <weiyongjun1(a)huawei.com> [ Upstream commit 7d8785bc7adbb4dc5ba8ee06994107637848ded8 ] Fix to return negative error code -ENOMEM from the IPC init error handling case instead of 0, as done elsewhere in this function. Fixes: c16211d6226d ("ASoC: SOF: Add Sound Open Firmware driver core") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> Link: https://lore.kernel.org/r/20200509093337.78897-1-weiyongjun1@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 91acfae7935c..74b438216216 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -176,6 +176,7 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) /* init the IPC */ sdev->ipc = snd_sof_ipc_init(sdev); if (!sdev->ipc) { + ret = -ENOMEM; dev_err(sdev->dev, "error: failed to init DSP IPC %d\n", ret); goto ipc_err; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 179/388] ALSA: firewire-lib: fix invalid assignment to union data for directional parameter
by Sasha Levin 18 Jun '20

18 Jun '20

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 8304cf77c92038cd1c50c27b69d30be695cc8003 ] Although the value of FDF is used just for outgoing stream, the assignment to union member is done for both directions of stream. At present this causes no issue because the value of same position is reassigned later for opposite stream. However, it's better to add if statement. Fixes: d3d10a4a1b19 ("ALSA: firewire-lib: use union for directional parameters") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20200508043635.349339-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/amdtp-am824.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c index 67d735e9a6a4..fea92e148790 100644 --- a/sound/firewire/amdtp-am824.c +++ b/sound/firewire/amdtp-am824.c @@ -82,7 +82,8 @@ int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate, if (err < 0) return err; - s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc; + if (s->direction == AMDTP_OUT_STREAM) + s->ctx_data.rx.fdf = AMDTP_FDF_AM824 | s->sfc; p->pcm_channels = pcm_channels; p->midi_ports = midi_ports; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 163/388] soundwire: slave: don't init debugfs on device registration error
by Sasha Levin 18 Jun '20

18 Jun '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 8893ab5e8ee5d7c12e0fc1dca4a309475064473d ] The error handling flow seems incorrect, there is no reason to try and add debugfs support if the device registration did not succeed. Return on error. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Link: https://lore.kernel.org/r/20200419185117.4233-2-yung-chuan.liao@linux.intel… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/slave.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index aace57fae7f8..4bacdb187eab 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -68,6 +68,8 @@ static int sdw_slave_add(struct sdw_bus *bus, list_del(&slave->node); mutex_unlock(&bus->bus_lock); put_device(&slave->dev); + + return ret; } sdw_slave_debugfs_init(slave); -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 162/388] ASoC: max98373: reorder max98373_reset() in resume
by Sasha Levin 18 Jun '20

18 Jun '20

From: Yong Zhi <yong.zhi(a)intel.com> [ Upstream commit 1a446873d7dd3a450f685928ce7f1907bde4583d ] During S3 test, the following error was observed: [ 726.174237] i2c_designware i2c_designware.0: platform_pm_resume+0x0/0x3d returned 0 after 0 usecs [ 726.184187] max98373 i2c-MX98373:00: calling max98373_resume+0x0/0x30 [snd_soc_max98373] @ 12698, parent: i2c-11 [ 726.195589] max98373 i2c-MX98373:00: Reset command failed. (ret:-16) When calling regmap_update_bits(), since map->reg_update_bits is NULL, _regmap_read() is entered with the following logic: if (!map->cache_bypass) { ret = regcache_read(map, reg, val); if (ret == 0) return 0; } if (map->cache_only) return -EBUSY; regcache_read() hits -EINVAL because MAX98373_R2000_SW_RESET is volatile, as map->cache_only is set by codec suspend, thus -EBUSY is returned. Fix by moving max98373_reset() after cache_only set to false in max98373_resume(). Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Link: https://lore.kernel.org/r/1588376661-29799-1-git-send-email-yong.zhi@intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98373.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index cae1def8902d..96718e3a1ad0 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -850,8 +850,8 @@ static int max98373_resume(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); - max98373_reset(max98373, dev); regcache_cache_only(max98373->regmap, false); + max98373_reset(max98373, dev); regcache_sync(max98373->regmap); return 0; } -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 160/388] ASoC: SOF: Update correct LED status at the first time usage of update_mute_led()
by Sasha Levin 18 Jun '20

18 Jun '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit 49c22696348d6e7c8a2ecfd7e60fddfe188ded82 ] At the first time update_mute_led() gets called, if channels are already muted, the temp value equals to led_value as 0, skipping the following LED setting. So set led_value to -1 as an uninitialized state, to update the correct LED status at first time usage. Fixes: 5d43001ae436 ("ASoC: SOF: acpi led support for switch controls") Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Acked-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200430091139.7003-1-kai.heng.feng@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/control.c | 4 ++-- sound/soc/sof/sof-audio.h | 2 +- sound/soc/sof/topology.c | 2 ++ 3 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index dfc412e2d956..6d63768d42aa 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -19,8 +19,8 @@ static void update_mute_led(struct snd_sof_control *scontrol, struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - unsigned int temp = 0; - unsigned int mask; + int temp = 0; + int mask; int i; mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index bf65f31af858..875a5fc13297 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -56,7 +56,7 @@ struct snd_sof_pcm { struct snd_sof_led_control { unsigned int use_led; unsigned int direction; - unsigned int led_value; + int led_value; }; /* ALSA SOF Kcontrol device */ diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index fe8ba3e05e08..ab2b69de1d4d 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1203,6 +1203,8 @@ static int sof_control_load(struct snd_soc_component *scomp, int index, return ret; } + scontrol->led_ctl.led_value = -1; + dobj->private = scontrol; list_add(&scontrol->list, &sdev->kcontrol_list); return ret; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 157/388] ASoC: rt5682: fix I2C/Soundwire dependencies
by Sasha Levin 18 Jun '20

18 Jun '20

From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit fd443a20c2f0950f3c31765a08f7dd49b3bc69cb ] If one of the two is a loadable module, the combined driver must not be built-in: aarch64-linux-ld: sound/soc/codecs/rt5682.o: in function `rt5682_sdw_hw_free': rt5682.c:(.text+0xb34): undefined reference to `sdw_stream_remove_slave' aarch64-linux-ld: sound/soc/codecs/rt5682.o: in function `rt5682_sdw_hw_params': rt5682.c:(.text+0xe78): undefined reference to `sdw_stream_add_slave' In particular, the soundwire driver must not be built-in if CONFIG_I2C=m. Fixes: 5549ea647997 ("ASoC: rt5682: fix unmet dependencies") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: https://lore.kernel.org/r/20200428214642.3925004-1-arnd@arndb.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7d2cbed55a9d..4c643d2e6d6b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1136,10 +1136,13 @@ config SND_SOC_RT5677_SPI config SND_SOC_RT5682 tristate depends on I2C || SOUNDWIRE + depends on SOUNDWIRE || !SOUNDWIRE + depends on I2C || !I2C config SND_SOC_RT5682_SDW tristate "Realtek RT5682 Codec - SDW" depends on SOUNDWIRE + depends on I2C || !I2C select SND_SOC_RT5682 select REGMAP_SOUNDWIRE -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 155/388] ASoC: component: suppress uninitialized-variable warning
by Sasha Levin 18 Jun '20

18 Jun '20

From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit be16a0f0dc8fab8e25d9cdbeb4f8f28afc9186d2 ] Old versions of gcc (tested on gcc-4.8) produce a warning for correct code: sound/soc/soc-compress.c: In function 'soc_compr_open': sound/soc/soc-compress.c:75:28: error: 'component' is used uninitialized in this function [-Werror=uninitialized] struct snd_soc_component *component, *save = NULL; Change the for_each_rtd_components() macro to ensure 'component' gets initialized to a value the compiler does not complain about. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Link: https://lore.kernel.org/r/20200428214754.3925368-1-arnd@arndb.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 946f88a6c63d..e0371e70242d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1177,7 +1177,7 @@ struct snd_soc_pcm_runtime { #define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->num_cpus] #define for_each_rtd_components(rtd, i, component) \ - for ((i) = 0; \ + for ((i) = 0, component = NULL; \ ((i) < rtd->num_components) && ((component) = rtd->components[i]);\ (i)++) #define for_each_rtd_cpu_dais(rtd, i, dai) \ -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 147/388] slimbus: ngd: get drvdata from correct device
by Sasha Levin 18 Jun '20

18 Jun '20

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit b58c663059b484f7ff547d076a34cf6d7a302e56 ] Get drvdata directly from parent instead of ngd dev, as ngd dev can probe defer and previously set drvdata will become null. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20200417093618.7929-1-srinivas.kandagatla@linaro.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index fc2575fef51b..7426b5884218 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1361,7 +1361,6 @@ static int of_qcom_slim_ngd_register(struct device *parent, ngd->pdev->driver_override = QCOM_SLIM_NGD_DRV_NAME; ngd->pdev->dev.of_node = node; ctrl->ngd = ngd; - platform_set_drvdata(ngd->pdev, ctrl); platform_device_add(ngd->pdev); ngd->base = ctrl->base + ngd->id * data->offset + @@ -1376,12 +1375,13 @@ static int of_qcom_slim_ngd_register(struct device *parent, static int qcom_slim_ngd_probe(struct platform_device *pdev) { - struct qcom_slim_ngd_ctrl *ctrl = platform_get_drvdata(pdev); struct device *dev = &pdev->dev; + struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev->parent); int ret; ctrl->ctrl.dev = dev; + platform_set_drvdata(pdev, ctrl); pm_runtime_use_autosuspend(dev); pm_runtime_set_autosuspend_delay(dev, QCOM_SLIM_NGD_AUTOSUSPEND); pm_runtime_set_suspended(dev); -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 139/388] ALSA: usb-audio: Fix racy list management in output queue
by Sasha Levin 18 Jun '20

18 Jun '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5b6cc38f3f3f37109ce72b60bda215a5f6892c0b ] The linked list entry from FIFO is peeked at queue_pending_output_urbs() but the actual element pop-out is performed outside the spinlock, and it's potentially racy. Do delete the link at the right place inside the spinlock. Fixes: 8fdff6a319e7 ("ALSA: snd-usb: implement new endpoint streaming model") Link: https://lore.kernel.org/r/20200424074016.14301-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/endpoint.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index d8dc7cb56d43..50104f658ed4 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -346,17 +346,17 @@ static void queue_pending_output_urbs(struct snd_usb_endpoint *ep) ep->next_packet_read_pos %= MAX_URBS; /* take URB out of FIFO */ - if (!list_empty(&ep->ready_playback_urbs)) + if (!list_empty(&ep->ready_playback_urbs)) { ctx = list_first_entry(&ep->ready_playback_urbs, struct snd_urb_ctx, ready_list); + list_del_init(&ctx->ready_list); + } } spin_unlock_irqrestore(&ep->lock, flags); if (ctx == NULL) return; - list_del_init(&ctx->ready_list); - /* copy over the length information */ for (i = 0; i < packet->packets; i++) ctx->packet_size[i] = packet->packet_size[i]; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 138/388] ALSA: usb-audio: Improve frames size computation
by Sasha Levin 18 Jun '20

18 Jun '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ] For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 4 ++++ sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++----- sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 2 ++ 4 files changed, 45 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 395403a2d33f..820e564656ed 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -84,6 +84,10 @@ struct snd_usb_endpoint { dma_addr_t sync_dma; /* DMA address of syncbuf */ unsigned int pipe; /* the data i/o pipe */ + unsigned int framesize[2]; /* small/large frame sizes in samples */ + unsigned int sample_rem; /* remainder from division fs/fps */ + unsigned int sample_accum; /* sample accumulator */ + unsigned int fps; /* frames per second */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ int freqshift; /* how much to shift the feedback value to get Q16.16 */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 4a9a2f6ef5a4..d8dc7cb56d43 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -124,12 +124,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) /* * For streaming based on information derived from sync endpoints, - * prepare_outbound_urb_sizes() will call next_packet_size() to + * prepare_outbound_urb_sizes() will call slave_next_packet_size() to * determine the number of samples to be sent in the next packet. * - * For implicit feedback, next_packet_size() is unused. + * For implicit feedback, slave_next_packet_size() is unused. */ -int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -146,6 +146,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) return ret; } +/* + * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes() + * will call next_packet_size() to determine the number of samples to be + * sent in the next packet. + */ +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +{ + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + ep->sample_accum += ep->sample_rem; + if (ep->sample_accum >= ep->fps) { + ep->sample_accum -= ep->fps; + ret = ep->framesize[1]; + } else { + ret = ep->framesize[0]; + } + + return ret; +} + static void retire_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *urb_ctx) { @@ -190,6 +213,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep, if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); @@ -874,10 +899,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->maxpacksize = fmt->maxpacksize; ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); - if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) { ep->freqn = get_usb_full_speed_rate(rate); - else + ep->fps = 1000; + } else { ep->freqn = get_usb_high_speed_rate(rate); + ep->fps = 8000; + } + + ep->sample_rem = rate % ep->fps; + ep->framesize[0] = rate / ep->fps; + ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps; /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -936,6 +968,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; + ep->sample_accum = 0; snd_usb_endpoint_start_quirk(ep); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 63a39d4fa8d8..d23fa0a8c11b 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -28,6 +28,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a4e4064f9aee..b50965ab3b3a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1579,6 +1579,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 124/388] ALSA: usb-audio: RME Babyface Pro mixer patch
by Sasha Levin 18 Jun '20

18 Jun '20

From: Thomas Ebeling <penguins(a)bollie.de> [ Upstream commit 3e8f3bd047163d30fb1ad32ca7e4628921555c09 ] Added mixer quirks to allow controlling the internal DSP of the RME Babyface Pro and its successor Babyface Pro FS. Signed-off-by: Thomas Ebeling <penguins(a)bollie.de> Link: https://lore.kernel.org/r/20200414211019.qprg7whepg2y7nei@bollie.ca9.eu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_quirks.c | 418 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 418 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index a5f65a9a0254..bdff8674942e 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2185,6 +2185,421 @@ static int snd_rme_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * RME Babyface Pro (FS) + * + * These devices exposes a couple of DSP functions via request to EP0. + * Switches are available via control registers, while routing is controlled + * by controlling the volume on each possible crossing point. + * Volume control is linear, from -inf (dec. 0) to +6dB (dec. 46341) with + * 0dB being at dec. 32768. + */ +enum { + SND_BBFPRO_CTL_REG1 = 0, + SND_BBFPRO_CTL_REG2 +}; + +#define SND_BBFPRO_CTL_REG_MASK 1 +#define SND_BBFPRO_CTL_IDX_MASK 0xff +#define SND_BBFPRO_CTL_IDX_SHIFT 1 +#define SND_BBFPRO_CTL_VAL_MASK 1 +#define SND_BBFPRO_CTL_VAL_SHIFT 9 +#define SND_BBFPRO_CTL_REG1_CLK_MASTER 0 +#define SND_BBFPRO_CTL_REG1_CLK_OPTICAL 1 +#define SND_BBFPRO_CTL_REG1_SPDIF_PRO 7 +#define SND_BBFPRO_CTL_REG1_SPDIF_EMPH 8 +#define SND_BBFPRO_CTL_REG1_SPDIF_OPTICAL 10 +#define SND_BBFPRO_CTL_REG2_48V_AN1 0 +#define SND_BBFPRO_CTL_REG2_48V_AN2 1 +#define SND_BBFPRO_CTL_REG2_SENS_IN3 2 +#define SND_BBFPRO_CTL_REG2_SENS_IN4 3 +#define SND_BBFPRO_CTL_REG2_PAD_AN1 4 +#define SND_BBFPRO_CTL_REG2_PAD_AN2 5 + +#define SND_BBFPRO_MIXER_IDX_MASK 0x1ff +#define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff +#define SND_BBFPRO_MIXER_VAL_SHIFT 9 +#define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf +#define SND_BBFPRO_MIXER_VAL_MAX 46341 // +6dB + +#define SND_BBFPRO_USBREQ_CTL_REG1 0x10 +#define SND_BBFPRO_USBREQ_CTL_REG2 0x17 +#define SND_BBFPRO_USBREQ_MIXER 0x12 + +static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg, + u8 index, u8 value) +{ + int err; + u16 usb_req, usb_idx, usb_val; + struct snd_usb_audio *chip = mixer->chip; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + if (reg == SND_BBFPRO_CTL_REG1) { + usb_req = SND_BBFPRO_USBREQ_CTL_REG1; + if (index == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) { + usb_idx = 3; + usb_val = value ? 3 : 0; + } else { + usb_idx = 1 << index; + usb_val = value ? usb_idx : 0; + } + } else { + usb_req = SND_BBFPRO_USBREQ_CTL_REG2; + usb_idx = 1 << index; + usb_val = value ? usb_idx : 0; + } + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), usb_req, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + usb_val, usb_idx, 0, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 reg, idx, val; + int pv; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + val = kcontrol->private_value >> SND_BBFPRO_CTL_VAL_SHIFT; + + if ((reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) || + (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4))) { + ucontrol->value.enumerated.item[0] = val; + } else { + ucontrol->value.integer.value[0] = val; + } + return 0; +} + +static int snd_bbfpro_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + u8 reg, idx; + int pv; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + + if (reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) { + static const char * const texts[2] = { + "AutoSync", + "Internal" + }; + return snd_ctl_enum_info(uinfo, 1, 2, texts); + } else if (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4)) { + static const char * const texts[2] = { + "-10dBV", + "+4dBu" + }; + return snd_ctl_enum_info(uinfo, 1, 2, texts); + } + + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + return 0; +} + +static int snd_bbfpro_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int err; + u8 reg, idx; + int old_value, pv, val; + + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + old_value = (pv >> SND_BBFPRO_CTL_VAL_SHIFT) & SND_BBFPRO_CTL_VAL_MASK; + + if ((reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) || + (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4))) { + val = ucontrol->value.enumerated.item[0]; + } else { + val = ucontrol->value.integer.value[0]; + } + + if (val > 1) + return -EINVAL; + + if (val == old_value) + return 0; + + kcontrol->private_value = reg + | ((idx & SND_BBFPRO_CTL_IDX_MASK) << SND_BBFPRO_CTL_IDX_SHIFT) + | ((val & SND_BBFPRO_CTL_VAL_MASK) << SND_BBFPRO_CTL_VAL_SHIFT); + + err = snd_bbfpro_ctl_update(mixer, reg, idx, val); + return err < 0 ? err : 1; +} + +static int snd_bbfpro_ctl_resume(struct usb_mixer_elem_list *list) +{ + u8 reg, idx; + int value, pv; + + pv = list->kctl->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + value = (pv >> SND_BBFPRO_CTL_VAL_SHIFT) & SND_BBFPRO_CTL_VAL_MASK; + + return snd_bbfpro_ctl_update(list->mixer, reg, idx, value); +} + +static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index, + u32 value) +{ + struct snd_usb_audio *chip = mixer->chip; + int err; + u16 idx; + u16 usb_idx, usb_val; + u32 v; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + idx = index & SND_BBFPRO_MIXER_IDX_MASK; + // 18 bit linear volume, split so 2 bits end up in index. + v = value & SND_BBFPRO_MIXER_VAL_MASK; + usb_idx = idx | (v & 0x3) << 14; + usb_val = (v >> 2) & 0xffff; + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), + SND_BBFPRO_USBREQ_MIXER, + USB_DIR_OUT | USB_TYPE_VENDOR | + USB_RECIP_DEVICE, + usb_val, usb_idx, 0, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = + kcontrol->private_value >> SND_BBFPRO_MIXER_VAL_SHIFT; + return 0; +} + +static int snd_bbfpro_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = SND_BBFPRO_MIXER_VAL_MIN; + uinfo->value.integer.max = SND_BBFPRO_MIXER_VAL_MAX; + return 0; +} + +static int snd_bbfpro_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int err; + u16 idx; + u32 new_val, old_value, uvalue; + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + uvalue = ucontrol->value.integer.value[0]; + idx = kcontrol->private_value & SND_BBFPRO_MIXER_IDX_MASK; + old_value = kcontrol->private_value >> SND_BBFPRO_MIXER_VAL_SHIFT; + + if (uvalue > SND_BBFPRO_MIXER_VAL_MAX) + return -EINVAL; + + if (uvalue == old_value) + return 0; + + new_val = uvalue & SND_BBFPRO_MIXER_VAL_MASK; + + kcontrol->private_value = idx + | (new_val << SND_BBFPRO_MIXER_VAL_SHIFT); + + err = snd_bbfpro_vol_update(mixer, idx, new_val); + return err < 0 ? err : 1; +} + +static int snd_bbfpro_vol_resume(struct usb_mixer_elem_list *list) +{ + int pv = list->kctl->private_value; + u16 idx = pv & SND_BBFPRO_MIXER_IDX_MASK; + u32 val = (pv >> SND_BBFPRO_MIXER_VAL_SHIFT) + & SND_BBFPRO_MIXER_VAL_MASK; + return snd_bbfpro_vol_update(list->mixer, idx, val); +} + +// Predfine elements +static const struct snd_kcontrol_new snd_bbfpro_ctl_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_ctl_info, + .get = snd_bbfpro_ctl_get, + .put = snd_bbfpro_ctl_put +}; + +static const struct snd_kcontrol_new snd_bbfpro_vol_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_vol_info, + .get = snd_bbfpro_vol_get, + .put = snd_bbfpro_vol_put +}; + +static int snd_bbfpro_ctl_add(struct usb_mixer_interface *mixer, u8 reg, + u8 index, char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_ctl_control; + + knew.name = name; + knew.private_value = (reg & SND_BBFPRO_CTL_REG_MASK) + | ((index & SND_BBFPRO_CTL_IDX_MASK) + << SND_BBFPRO_CTL_IDX_SHIFT); + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_ctl_resume, + &knew, NULL); +} + +static int snd_bbfpro_vol_add(struct usb_mixer_interface *mixer, u16 index, + char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_vol_control; + + knew.name = name; + knew.private_value = index & SND_BBFPRO_MIXER_IDX_MASK; + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_vol_resume, + &knew, NULL); +} + +static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i, o; + char name[48]; + + static const char * const input[] = { + "AN1", "AN2", "IN3", "IN4", "AS1", "AS2", "ADAT3", + "ADAT4", "ADAT5", "ADAT6", "ADAT7", "ADAT8"}; + + static const char * const output[] = { + "AN1", "AN2", "PH3", "PH4", "AS1", "AS2", "ADAT3", "ADAT4", + "ADAT5", "ADAT6", "ADAT7", "ADAT8"}; + + for (o = 0 ; o < 12 ; ++o) { + for (i = 0 ; i < 12 ; ++i) { + // Line routing + snprintf(name, sizeof(name), + "%s-%s-%s Playback Volume", + (i < 2 ? "Mic" : "Line"), + input[i], output[o]); + err = snd_bbfpro_vol_add(mixer, (26 * o + i), name); + if (err < 0) + return err; + + // PCM routing... yes, it is output remapping + snprintf(name, sizeof(name), + "PCM-%s-%s Playback Volume", + output[i], output[o]); + err = snd_bbfpro_vol_add(mixer, (26 * o + 12 + i), + name); + if (err < 0) + return err; + } + } + + // Control Reg 1 + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_CLK_OPTICAL, + "Sample Clock Source"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_PRO, + "IEC958 Pro Mask"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_EMPH, + "IEC958 Emphasis"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_OPTICAL, + "IEC958 Switch"); + if (err < 0) + return err; + + // Control Reg 2 + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_48V_AN1, + "Mic-AN1 48V"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_48V_AN2, + "Mic-AN2 48V"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_SENS_IN3, + "Line-IN3 Sens."); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_SENS_IN4, + "Line-IN4 Sens."); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_PAD_AN1, + "Mic-AN1 PAD"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_PAD_AN2, + "Mic-AN2 PAD"); + if (err < 0) + return err; + + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2286,6 +2701,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ err = snd_sc1810_init_mixer(mixer); break; + case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ + err = snd_bbfpro_controls_create(mixer); + break; } return err; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 103/388] ASoC: meson: add missing free_irq() in error path
by Sasha Levin 18 Jun '20

18 Jun '20

From: "Pavel Machek (CIP)" <pavel(a)denx.de> [ Upstream commit 3b8a299a58b2afce464ae11324b59dcf0f1d10a7 ] free_irq() is missing in case of error, fix that. Signed-off-by: Pavel Machek (CIP) <pavel(a)denx.de> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200606153103.GA17905@amd Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/axg-fifo.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 2e9b56b29d31..b2e867113226 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -249,7 +249,7 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Enable pclk to access registers and clock the fifo ip */ ret = clk_prepare_enable(fifo->pclk); if (ret) - return ret; + goto free_irq; /* Setup status2 so it reports the memory pointer */ regmap_update_bits(fifo->map, FIFO_CTRL1, @@ -269,8 +269,14 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Take memory arbitror out of reset */ ret = reset_control_deassert(fifo->arb); if (ret) - clk_disable_unprepare(fifo->pclk); + goto free_clk; + + return 0; +free_clk: + clk_disable_unprepare(fifo->pclk); +free_irq: + free_irq(fifo->irq, ss); return ret; } EXPORT_SYMBOL_GPL(axg_fifo_pcm_open); -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 094/388] ASoC: meson: fix memory leak of links if allocation of ldata fails
by Sasha Levin 18 Jun '20

18 Jun '20

From: Colin Ian King <colin.king(a)canonical.com> [ Upstream commit 6e801dc411329aff592fbd48fb116183d0acdb00 ] Currently if the allocation of ldata fails the error return path does not kfree the allocated links object. Fix this by adding an error exit return path that performs the necessary kfree'ing. Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Addresses-Coverity: ("Resource leak") Link: https://lore.kernel.org/r/20200604171216.60043-1-colin.king@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/meson-card-utils.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 2ca8c98e204f..5a4a91c88734 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -49,19 +49,26 @@ int meson_card_reallocate_links(struct snd_soc_card *card, links = krealloc(priv->card.dai_link, num_links * sizeof(*priv->card.dai_link), GFP_KERNEL | __GFP_ZERO); + if (!links) + goto err_links; + ldata = krealloc(priv->link_data, num_links * sizeof(*priv->link_data), GFP_KERNEL | __GFP_ZERO); - - if (!links || !ldata) { - dev_err(priv->card.dev, "failed to allocate links\n"); - return -ENOMEM; - } + if (!ldata) + goto err_ldata; priv->card.dai_link = links; priv->link_data = ldata; priv->card.num_links = num_links; return 0; + +err_ldata: + kfree(links); +err_links: + dev_err(priv->card.dev, "failed to allocate links\n"); + return -ENOMEM; + } EXPORT_SYMBOL_GPL(meson_card_reallocate_links); -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 085/388] ASoC: qcom: q6asm-dai: kCFI fix
by Sasha Levin 18 Jun '20

18 Jun '20

From: John Stultz <john.stultz(a)linaro.org> [ Upstream commit a6b675a89e51a1cdad0481b809b7840d3f86e4b5 ] Fixes the following kCFI crash seen on db845c, caused by the function prototypes not matching the callback function prototype. [ 82.585661] Unable to handle kernel NULL pointer dereference at virtual address 0000000000000001 [ 82.595387] Mem abort info: [ 82.599463] ESR = 0x96000005 [ 82.602658] EC = 0x25: DABT (current EL), IL = 32 bits [ 82.608177] SET = 0, FnV = 0 [ 82.611829] EA = 0, S1PTW = 0 [ 82.615369] Data abort info: [ 82.618751] ISV = 0, ISS = 0x00000005 [ 82.622641] CM = 0, WnR = 0 [ 82.625774] user pgtable: 4k pages, 39-bit VAs, pgdp=0000000174259000 [ 82.632292] [0000000000000001] pgd=0000000000000000, pud=0000000000000000 [ 82.639167] Internal error: Oops: 96000005 [#1] PREEMPT SMP [ 82.644795] Modules linked in: hci_uart btqca xhci_plat_hcd xhci_pci_renesas xhci_pci xhci_hcd wcn36xx wcnss_ctrl wcd934x vctrl_regulator ufs_qcom syscon_reboot_e [ 82.644927] qcom_apcs_ipc_mailbox q6asm_dai q6routing q6asm q6afe_dai q6adm q6afe q6core q6dsp_common pm8941_pwrkey pm8916_wdt platform_mhu pinctrl_spmi_mpp pine [ 82.812982] CPU: 3 PID: 240 Comm: kworker/u16:4 Tainted: G W 5.6.0-rc7-mainline-00960-g0c34353d11b9-dirty #1 [ 82.824201] Hardware name: Thundercomm Dragonboard 845c (DT) [ 82.829937] Workqueue: qcom_apr_rx apr_rxwq [apr] [ 82.834698] pstate: 80c00005 (Nzcv daif +PAN +UAO) [ 82.839553] pc : __cfi_check_fail+0x4/0x1c [q6asm_dai] [ 82.844754] lr : __cfi_check+0x3a8/0x3b0 [q6asm_dai] [ 82.849767] sp : ffffffc0105f3c20 [ 82.853123] x29: ffffffc0105f3c30 x28: 0000000000000020 [ 82.858489] x27: ffffff80f4588400 x26: ffffff80f458ec94 [ 82.863854] x25: ffffff80f458ece8 x24: ffffffe3670c7000 [ 82.869220] x23: ffffff8094bb7b34 x22: ffffffe367137000 [ 82.874585] x21: bd07909b332eada6 x20: 0000000000000001 [ 82.879950] x19: ffffffe36713863c x18: ffffff80f8df4430 [ 82.885316] x17: 0000000000000001 x16: ffffffe39d15e660 [ 82.890681] x15: 0000000000000001 x14: 0000000000000027 [ 82.896047] x13: 0000000000000000 x12: ffffffe39e6465a0 [ 82.901413] x11: 0000000000000051 x10: 000000000000ffff [ 82.906779] x9 : 000ffffffe366c19 x8 : c3c5f18762d1ceef [ 82.912145] x7 : 0000000000000000 x6 : ffffffc010877698 [ 82.917511] x5 : ffffffc0105f3c00 x4 : 0000000000000000 [ 82.922877] x3 : 0000000000000000 x2 : 0000000000000001 [ 82.928243] x1 : ffffffe36713863c x0 : 0000000000000001 [ 82.933610] Call trace: [ 82.936099] __cfi_check_fail+0x4/0x1c [q6asm_dai] [ 82.940955] q6asm_srvc_callback+0x22c/0x618 [q6asm] [ 82.945973] apr_rxwq+0x1a8/0x27c [apr] [ 82.949861] process_one_work+0x2e8/0x54c [ 82.953919] worker_thread+0x27c/0x4d4 [ 82.957715] kthread+0x144/0x154 [ 82.960985] ret_from_fork+0x10/0x18 [ 82.964603] Code: a8c37bfd f85f8e5e d65f03c0 b40000a0 (39400008) [ 82.970762] ---[ end trace 410accb839617143 ]--- [ 82.975429] Kernel panic - not syncing: Fatal exception Signed-off-by: John Stultz <john.stultz(a)linaro.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Stephan Gerhold <stephan(a)gerhold.net> Cc: Sami Tolvanen <samitolvanen(a)google.com> Cc: Todd Kjos <tkjos(a)google.com> Cc: Alistair Delva <adelva(a)google.com> Cc: Amit Pundir <amit.pundir(a)linaro.org> Cc: Sumit Semwal <sumit.semwal(a)linaro.org> Cc: alsa-devel(a)alsa-project.org Link: https://lore.kernel.org/r/20200529213823.98812-1-john.stultz@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 125af00bba53..4640804aab7f 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -176,7 +176,7 @@ static const struct snd_compr_codec_caps q6asm_compr_caps = { }; static void event_handler(uint32_t opcode, uint32_t token, - uint32_t *payload, void *priv) + void *payload, void *priv) { struct q6asm_dai_rtd *prtd = priv; struct snd_pcm_substream *substream = prtd->substream; @@ -490,7 +490,7 @@ static int q6asm_dai_hw_params(struct snd_soc_component *component, } static void compress_event_handler(uint32_t opcode, uint32_t token, - uint32_t *payload, void *priv) + void *payload, void *priv) { struct q6asm_dai_rtd *prtd = priv; struct snd_compr_stream *substream = prtd->cstream; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 026/388] ALSA: isa/wavefront: prevent out of bounds write in ioctl
by Sasha Levin 18 Jun '20

18 Jun '20

From: Dan Carpenter <dan.carpenter(a)oracle.com> [ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ] The "header->number" comes from the ioctl and it needs to be clamped to prevent out of bounds writes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/wavefront/wavefront_synth.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index c5b1d5900eed..d6420d224d09 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1171,7 +1171,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) "alias for %d\n", header->number, header->hdr.a.OriginalSample); - + + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + munge_int32 (header->number, &alias_hdr[0], 2); munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2); munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset), @@ -1202,6 +1205,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) int num_samples; unsigned char *msample_hdr; + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL); if (! msample_hdr) return -ENOMEM; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 025/388] ALSA: hda/realtek - Introduce polarity for micmute LED GPIO
by Sasha Levin 18 Jun '20

18 Jun '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit dbd13179780555ecd3c992dea1222ca31920e892 ] Currently mute LED and micmute LED share the same GPIO polarity. So split the polarity for mute and micmute, in case they have different polarities. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200430083255.5093-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2c4575909441..e057ecb5a904 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -81,6 +81,7 @@ struct alc_spec { /* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */ int mute_led_polarity; + int micmute_led_polarity; hda_nid_t mute_led_nid; hda_nid_t cap_mute_led_nid; @@ -4080,11 +4081,9 @@ static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec, /* update LED status via GPIO */ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, - bool enabled) + int polarity, bool enabled) { - struct alc_spec *spec = codec->spec; - - if (spec->mute_led_polarity) + if (polarity) enabled = !enabled; alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */ } @@ -4095,7 +4094,8 @@ static void alc_fixup_gpio_mute_hook(void *private_data, int enabled) struct hda_codec *codec = private_data; struct alc_spec *spec = codec->spec; - alc_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); + alc_update_gpio_led(codec, spec->gpio_mute_led_mask, + spec->mute_led_polarity, enabled); } /* turn on/off mic-mute LED via GPIO per capture hook */ @@ -4104,6 +4104,7 @@ static void alc_gpio_micmute_update(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_update_gpio_led(codec, spec->gpio_mic_led_mask, + spec->micmute_led_polarity, spec->gen.micmute_led.led_value); } @@ -5808,7 +5809,8 @@ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, snd_hda_gen_hp_automute(codec, jack); /* mute_led_polarity is set to 0, so we pass inverted value here */ - alc_update_gpio_led(codec, 0x10, !spec->gen.hp_jack_present); + alc_update_gpio_led(codec, 0x10, spec->mute_led_polarity, + !spec->gen.hp_jack_present); } /* Manage GPIOs for HP EliteBook Folio 9480m. -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 020/388] ASoC: codecs: wm97xx: fix ac97 dependency
by Sasha Levin 18 Jun '20

18 Jun '20

From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit ee2cbe06935bfa58f1fe07dc2a2283945f4b97dc ] A recent build fix got the dependency slightly wrong, breaking builds with CONFIG_AC97_BUS_NEW: WARNING: unmet direct dependencies detected for SND_SOC_WM9713 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AC97_BUS [=n] Selected by [m]: - SND_SOC_ZYLONITE [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_PXA2XX_SOC [=m] && MACH_ZYLONITE [=y] && AC97_BUS [=n]=n WARNING: unmet direct dependencies detected for SND_SOC_WM9712 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AC97_BUS [=n] Selected by [m]: - SND_PXA2XX_SOC_EM_X270 [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_PXA2XX_SOC [=m] && (MACH_EM_X270 [=n] || MACH_EXEDA [=n] || MACH_CM_X300 [=y]) && AC97_BUS [=n]=n Change the dependency to allow either version of the AC97 library code. Fixes: 5a309875787d ("ASoC: Fix SND_SOC_ALL_CODECS imply ac97 fallout") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200428212721.2877627-1-arnd@arndb.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e60e0b6a689c..8cdc68c141dc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1620,19 +1620,19 @@ config SND_SOC_WM9090 config SND_SOC_WM9705 tristate - depends on SND_SOC_AC97_BUS + depends on SND_SOC_AC97_BUS || AC97_BUS_NEW select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9712 tristate - depends on SND_SOC_AC97_BUS + depends on SND_SOC_AC97_BUS || AC97_BUS_NEW select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9713 tristate - depends on SND_SOC_AC97_BUS + depends on SND_SOC_AC97_BUS || AC97_BUS_NEW select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 016/388] ASoC: davinci-mcasp: Fix dma_chan refcnt leak when getting dma type
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit a697ae6ea56e23397341b027098c1b11d9ab13da ] davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a reference of the specified dma_chan object to "chan" with increased refcnt. When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in one exception handling path of davinci_mcasp_get_dma_type(). When chan device is NULL, the function forgets to decrease the refcnt increased by dma_request_chan(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when chan device is NULL. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 734ffe925c4d..7a7db743dc5b 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1896,8 +1896,10 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) PTR_ERR(chan)); return PTR_ERR(chan); } - if (WARN_ON(!chan->device || !chan->device->dev)) + if (WARN_ON(!chan->device || !chan->device->dev)) { + dma_release_channel(chan); return -EINVAL; + } if (chan->device->dev->of_node) ret = of_property_read_string(chan->device->dev->of_node, -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 013/388] ASoC: fsl_esai: Disable exception interrupt before scheduling tasklet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 1fecbb71fe0e46b886f84e3b6decca6643c3af6d ] Disable exception interrupt before scheduling tasklet, otherwise if the tasklet isn't handled immediately, there will be endless xrun interrupt. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/a8f2ad955aac9e52587beedc1133b3efbe746895.15879688… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7a49d03463a..84290be778f0 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -87,6 +87,10 @@ static irqreturn_t esai_isr(int irq, void *devid) if ((saisr & (ESAI_SAISR_TUE | ESAI_SAISR_ROE)) && esai_priv->reset_at_xrun) { dev_dbg(&pdev->dev, "reset module for xrun\n"); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xEIE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xEIE_MASK, 0); tasklet_schedule(&esai_priv->task); } -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 007/388] ASoC: SOF: imx8: Fix randbuild error
by Sasha Levin 18 Jun '20

18 Jun '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit fe17e6cdc0fefca96ba9659be4b2b07487cbf0c5 ] when do randconfig like this: CONFIG_SND_SOC_SOF_IMX8_SUPPORT=y CONFIG_SND_SOC_SOF_IMX8=y CONFIG_SND_SOC_SOF_OF=y CONFIG_IMX_DSP=m CONFIG_IMX_SCU=y there is a link error: sound/soc/sof/imx/imx8.o: In function 'imx8_send_msg': imx8.c:(.text+0x380): undefined reference to 'imx_dsp_ring_doorbell' Select IMX_DSP in SND_SOC_SOF_IMX8_SUPPORT to fix this Fixes: f9ad75468453 ("ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Link: https://lore.kernel.org/r/20200409071832.2039-2-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index bae4f7bf5f75..812749064ca8 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -14,7 +14,7 @@ if SND_SOC_SOF_IMX_TOPLEVEL config SND_SOC_SOF_IMX8_SUPPORT bool "SOF support for i.MX8" depends on IMX_SCU - depends on IMX_DSP + select IMX_DSP help This adds support for Sound Open Firmware for NXP i.MX8 platforms Say Y if you have such a device. -- 2.25.1

1 0

Move ownership for repositories in Github related to gobject-introspection
by Takashi Sakamoto 18 Jun '20

18 Jun '20

Hi Jaroslav, Now I'm going to prepare for official release of alsa-gobject v0.1.0. At the same time, Rust binding is also OK to be released. Currently It's in my private repository[1] but I think it better to put under alsa-project project account in Github. Github supports transferring ownership, and this time I'd like to utilize the feature if you don't mind it. After transferring. some arrangement for the repository are perhaps required; e.g. to put it to 'gobject-introspection' team[3]. Would I ask it to you? Additionally, I'd like to move repositories for libhinawa[4][5][6] as well, which is required to make up control service in user space for devices supported by ALSA firewire stack[7]. [1] https://github.com/takaswie/alsa-gobject-rs/ [2] https://help.github.com/en/github/administering-a-repository/transferring-a… [3] https://github.com/orgs/alsa-project/teams/gobject-introspection [4] https://github.com/takaswie/libhinawa [5] https://github.com/takaswie/libhinawa-docs [6] https://github.com/takaswie/hinawa-rs [7] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/168550.html Regards Takashi Sakamoto

2 6

[PATCH 0/3] ASoC: SOF: Intel: update PCI IDs
by Pierre-Louis Bossart 17 Jun '20

17 Jun '20

Clean-up CometLake and add missing PCI IDs. Changes for the legacy driver are sent separately. Pierre-Louis Bossart (3): 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 sound/hda/intel-dsp-config.c | 4 +--- sound/soc/intel/boards/Kconfig | 4 ++-- sound/soc/sof/intel/Kconfig | 29 ++++++++--------------------- sound/soc/sof/sof-pci-dev.c | 24 ++++++++++++++---------- 4 files changed, 25 insertions(+), 36 deletions(-) -- 2.20.1

2 4

[PATCH] ASoC: hdac_hda: fix memleak with regmap not freed on remove
by Pierre-Louis Bossart 17 Jun '20

17 Jun '20

kmemleak throws error reports on module load/unload tests, add snd_hdac_regmap_exit() in .remove(). While we are at it, also fix the error handling flow in .probe() to use snd_hdac_regmap_exit() if needed. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/codecs/hdac_hda.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index de003acb1951..473efe9ef998 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -441,13 +441,13 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_set_name(hcodec, hcodec->preset->name); if (ret < 0) { dev_err(&hdev->dev, "name failed %s\n", hcodec->preset->name); - goto error; + goto error_pm; } ret = snd_hdac_regmap_init(&hcodec->core); if (ret < 0) { dev_err(&hdev->dev, "regmap init failed\n"); - goto error; + goto error_pm; } patch = (hda_codec_patch_t)hcodec->preset->driver_data; @@ -455,7 +455,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = patch(hcodec); if (ret < 0) { dev_err(&hdev->dev, "patch failed %d\n", ret); - goto error; + goto error_regmap; } } else { dev_dbg(&hdev->dev, "no patch file found\n"); @@ -467,7 +467,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) ret = snd_hda_codec_parse_pcms(hcodec); if (ret < 0) { dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); - goto error; + goto error_regmap; } /* HDMI controls need to be created in machine drivers */ @@ -476,7 +476,7 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) if (ret < 0) { dev_err(&hdev->dev, "unable to create controls %d\n", ret); - goto error; + goto error_regmap; } } @@ -496,7 +496,9 @@ static int hdac_hda_codec_probe(struct snd_soc_component *component) return 0; -error: +error_regmap: + snd_hdac_regmap_exit(hdev); +error_pm: pm_runtime_put(&hdev->dev); error_no_pm: snd_hdac_ext_bus_link_put(hdev->bus, hlink); @@ -518,6 +520,8 @@ static void hdac_hda_codec_remove(struct snd_soc_component *component) pm_runtime_disable(&hdev->dev); snd_hdac_ext_bus_link_put(hdev->bus, hlink); + + snd_hdac_regmap_exit(hdev); } static const struct snd_soc_dapm_route hdac_hda_dapm_routes[] = { -- 2.20.1

2 1

sof-soundwire: initial UCM2 version
by GitHub pull_request - opened 17 Jun '20

17 Jun '20

alsa-project/alsa-ucm-conf pull request #36 was opened from plbossart: Add support for sof-soundwire driver, with conditional inclusion of sections for RT700, RT711, RT5682, RT1308, RT715. Required test files for the ucm-validation are provided at https://github.com/alsa-project/alsa-tests/pull/10 Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/36 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/36.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] regmap: fix memory leak with map->patch
by Pierre-Louis Bossart 17 Jun '20

17 Jun '20

kmemleak throws the following error on devices using regmap_register_patch(). map->patch is allocated dynamically with krealloc() but never freed. root@Zotac:~/kmod# cat /sys/kernel/debug/kmemleak unreferenced object 0xffff8880641c4e08 (size 96): comm "systemd-udevd", pid 266, jiffies 4294736541 (age 414.914s) hex dump (first 32 bytes): 3d 01 00 00 00 36 00 00 00 00 00 00 12 01 00 00 =....6.......... a8 0a 00 00 00 00 00 00 14 01 00 00 aa 0a 00 00 ................ backtrace: [<00000000f2841faf>] krealloc+0x63/0xc0 [<00000000ed643fb3>] regmap_register_patch+0x68/0x1d0 [<000000001806a5c3>] 0xffffffffc088a5f9 [<00000000f4781e26>] i2c_device_probe+0x462/0x4b0 [<000000009dd7f5f4>] really_probe+0x17f/0x520 [<00000000be42920d>] driver_probe_device+0x114/0x170 [<00000000e5840d3f>] device_driver_attach+0x82/0x90 [<0000000027828801>] __driver_attach+0xa9/0x190 [<0000000085195e73>] bus_for_each_dev+0xe1/0x140 [<0000000070edb38c>] bus_add_driver+0x224/0x2e0 [<000000002d41b24a>] driver_register+0xd3/0x150 [<0000000055dbb3fb>] i2c_register_driver+0x6b/0xd0 [<00000000bf84b954>] do_one_initcall+0x86/0x2a0 [<000000001d939871>] do_init_module+0xf8/0x350 [<00000000267b474f>] load_module+0x41a9/0x44c0 [<000000007d8014dc>] __do_sys_finit_module+0x121/0x1b0 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/base/regmap/regmap.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index c472f624382d..46f4b6322e47 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -1357,6 +1357,7 @@ void regmap_exit(struct regmap *map) if (map->hwlock) hwspin_lock_free(map->hwlock); kfree_const(map->name); + kfree(map->patch); kfree(map); } EXPORT_SYMBOL_GPL(regmap_exit); -- 2.20.1

2 1

[PATCH] ASoC: meson: imply acodec glue on axg sound card
by Jerome Brunet 17 Jun '20

17 Jun '20

When axg card driver support is enabled, lets enable the related internal DAC glue by default. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 8b6295283989..363dc3b1bbe4 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -68,6 +68,7 @@ config SND_MESON_AXG_SOUND_CARD imply SND_MESON_AXG_SPDIFOUT imply SND_MESON_AXG_SPDIFIN imply SND_MESON_AXG_PDM + imply SND_MESON_G12A_TOACODEC imply SND_MESON_G12A_TOHDMITX if DRM_MESON_DW_HDMI help Select Y or M to add support for the AXG SoC sound card -- 2.25.4

2 1

[PATCH v2 1/2] ASoC: bindings: fsl-asoc-card: Add compatible string for MQS
by Shengjiu Wang 17 Jun '20

17 Jun '20

Add compatible string "fsl,imx-audio-mqs" for MQS, and move "audio-routing" property to be optional for MQS doesn't need such property. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - Move "audio-routing" to optional. .../devicetree/bindings/sound/fsl-asoc-card.txt | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index c60a5732d29c..ca9a3a43adfd 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -34,6 +34,8 @@ The compatible list for this generic sound card currently: "fsl,imx-audio-wm8960" + "fsl,imx-audio-mqs" + Required properties: - compatible : Contains one of entries in the compatible list. @@ -44,6 +46,11 @@ Required properties: - audio-codec : The phandle of an audio codec +Optional properties: + + - audio-asrc : The phandle of ASRC. It can be absent if there's no + need to add ASRC support via DPCM. + - audio-routing : A 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 @@ -60,11 +67,6 @@ Required properties: coexisting in order to support the old bindings of wm8962 and sgtl5000. -Optional properties: - - - audio-asrc : The phandle of ASRC. It can be absent if there's no - need to add ASRC support via DPCM. - Optional unless SSI is selected as a CPU DAI: - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) -- 2.21.0

3 3

[PATCH v3 1/2] ASoC: bindings: fsl_spdif: Add new compatible string for imx6sx
by Shengjiu Wang 17 Jun '20

17 Jun '20

Add new compatible string "fsl,imx6sx-spdif" in the binding document. And add compatible string "fsl,vf610-spdif" which was missed before. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.txt | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index 8b324f82a782..e1365b0ee1e9 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -6,7 +6,11 @@ a fibre cable. Required properties: - - compatible : Compatible list, must contain "fsl,imx35-spdif". + - compatible : Compatible list, should contain one of the following + compatibles: + "fsl,imx35-spdif", + "fsl,vf610-spdif", + "fsl,imx6sx-spdif", - reg : Offset and length of the register set for the device. -- 2.21.0

2 2

[PATCH] ASoC: tegra: Fix reference count leaks.
by wu000273＠umn.edu 17 Jun '20

17 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> Calling pm_runtime_get_sync increments the counter even in case of failure, causing incorrect ref count if pm_runtime_put is not called in error handling paths. Call pm_runtime_put if pm_runtime_get_sync fails. Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> --- sound/soc/tegra/tegra30_ahub.c | 4 +++- sound/soc/tegra/tegra30_i2s.c | 4 +++- 2 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 635eacbd28d4..156e3b9d613c 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -643,8 +643,10 @@ static int tegra30_ahub_resume(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put(dev); return ret; + } ret = regcache_sync(ahub->regmap_ahub); ret |= regcache_sync(ahub->regmap_apbif); pm_runtime_put(dev); diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index d59882ec48f1..db5a8587bfa4 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -567,8 +567,10 @@ static int tegra30_i2s_resume(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put(dev); return ret; + } ret = regcache_sync(i2s->regmap); pm_runtime_put(dev); -- 2.17.1

3 2

pcm: direct: correctly apply existing interval settings
by GitHub pull_request - edited 17 Jun '20

17 Jun '20

alsa-project/alsa-lib pull request #62 was edited from aditpape: Feature 'variable periodsize' allows to extend user period size up to buffer_size/2 independent of slave period. On enlargement of the settings for period_time.max and period_size.max the setting for openmax was not updated. This lead to the effect, that if the slave period itself had openmax set it was still set on the extended size. Configuration of a period matching half buffer size was thus rejected. Example for failure: period size of 384 (half buffer size) is requested which is rejected and rounded down to 352: <pre> root@oracle-virtualbox:~# arecord -DDSNOOP_32 -r48000 -fS16_LE -c2 --period-size=382 -vv --dump-hw-params > /dev/null Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo HW Params of device "DSNOOP_32": -------------------- ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: (666 7334) PERIOD_SIZE: [32 352] PERIOD_BYTES: [128 1408] PERIODS: [2 24] BUFFER_TIME: (1333 16000] BUFFER_SIZE: [64 768] BUFFER_BYTES: [256 3072] TICK_TIME: ALL -------------------- Direct Snoop PCM Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 704 period_size : 352 period_time : 7333 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 352 period_event : 0 start_threshold : 1 stop_threshold : 704 silence_threshold: 0 silence_size : 0 boundary : 6341068275337658368 Hardware PCM card 0 'Intel 82801AA-ICH' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 32 period_time : 666 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 32 period_event : 0 start_threshold : 1 stop_threshold : 6917529027641081856 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 224 When correctly applying the openmax setting: root@oracle-virtualbox:~# arecord -DDSNOOP_32 -r48000 -fS16_LE -c2 --period-size=382 -vv --dump-hw-params > /dev/null Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo HW Params of device "DSNOOP_32": -------------------- ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: (666 8000] PERIOD_SIZE: [32 384] PERIOD_BYTES: [128 1536] PERIODS: [2 24] BUFFER_TIME: (1333 16000] BUFFER_SIZE: [64 768] BUFFER_BYTES: [256 3072] TICK_TIME: ALL -------------------- Direct Snoop PCM Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 384 period_time : 8000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 384 period_event : 0 start_threshold : 1 stop_threshold : 768 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 Hardware PCM card 0 'Intel 82801AA-ICH' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 32 period_time : 666 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 32 period_event : 0 start_threshold : 1 stop_threshold : 6917529027641081856 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 224 </pre> Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/62 Patch URL : https://github.com/alsa-project/alsa-lib/pull/62.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

pcm: direct: correctly apply existing interval settings
by GitHub pull_request - opened 17 Jun '20

17 Jun '20

alsa-project/alsa-lib pull request #62 was opened from aditpape: Feature 'variable periodsize' allows to extend user period size up to buffer_size/2 independent of slave period. On enlargement of the settings for period_time.max and period_size.max the setting for openmax was not updated. This lead to the effect, that if the slave period itself had openmax set it was still set on the extended size. Configuration of a period matching half buffer size was thus rejected. Example for failure: period size of 384 (half buffer size) is requested which is rejected and rounded down to 352: root@oracle-virtualbox:~# arecord -DDSNOOP_32 -r48000 -fS16_LE -c2 --period-size=382 -vv --dump-hw-params > /dev/null Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo HW Params of device "DSNOOP_32": -------------------- ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: (666 7334) PERIOD_SIZE: [32 352] PERIOD_BYTES: [128 1408] PERIODS: [2 24] BUFFER_TIME: (1333 16000] BUFFER_SIZE: [64 768] BUFFER_BYTES: [256 3072] TICK_TIME: ALL -------------------- Direct Snoop PCM Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 704 period_size : 352 period_time : 7333 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 352 period_event : 0 start_threshold : 1 stop_threshold : 704 silence_threshold: 0 silence_size : 0 boundary : 6341068275337658368 Hardware PCM card 0 'Intel 82801AA-ICH' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 32 period_time : 666 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 32 period_event : 0 start_threshold : 1 stop_threshold : 6917529027641081856 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 224 When correctly applying the openmax setting: root@oracle-virtualbox:~# arecord -DDSNOOP_32 -r48000 -fS16_LE -c2 --period-size=382 -vv --dump-hw-params > /dev/null Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo HW Params of device "DSNOOP_32": -------------------- ACCESS: MMAP_INTERLEAVED MMAP_NONINTERLEAVED RW_INTERLEAVED RW_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: (666 8000] PERIOD_SIZE: [32 384] PERIOD_BYTES: [128 1536] PERIODS: [2 24] BUFFER_TIME: (1333 16000] BUFFER_SIZE: [64 768] BUFFER_BYTES: [256 3072] TICK_TIME: ALL -------------------- Direct Snoop PCM Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 384 period_time : 8000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 384 period_event : 0 start_threshold : 1 stop_threshold : 768 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 Hardware PCM card 0 'Intel 82801AA-ICH' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 768 period_size : 32 period_time : 666 tstamp_mode : ENABLE tstamp_type : MONOTONIC period_step : 1 avail_min : 32 period_event : 0 start_threshold : 1 stop_threshold : 6917529027641081856 silence_threshold: 0 silence_size : 0 boundary : 6917529027641081856 appl_ptr : 0 hw_ptr : 224 Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/62 Patch URL : https://github.com/alsa-project/alsa-lib/pull/62.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 1/2] ALSA: hda/realtek: Add COEF controlled micmute LED support
by Kai-Heng Feng 17 Jun '20

17 Jun '20

Currently, HDA codec LED class can only used by by GPIO controlled LED. However, there are some new systems that control LED via COEF instead of GPIO. In order to support those systems, create a new helper that can be facilitated by both COEF controlled and GPIO controlled LED, and use generic call_micmute_led_update() to control both case. In addition to that, add LED_CORE_SUSPENDRESUME flag since some systems don't restore the LED properly after suspend. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v2: - Prevent platforms like Dell, Lenovoe and Huawei create double LED class devices. sound/pci/hda/hda_generic.c | 3 ++- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/patch_realtek.c | 29 ++++++++++++++++------------- 3 files changed, 19 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f4e9d9445e18..9c3f1990c621 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3897,7 +3897,7 @@ enum { MICMUTE_LED_FOLLOW_MUTE, }; -static void call_micmute_led_update(struct hda_codec *codec) +void call_micmute_led_update(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; unsigned int val; @@ -3924,6 +3924,7 @@ static void call_micmute_led_update(struct hda_codec *codec) if (spec->micmute_led.update) spec->micmute_led.update(codec); } +EXPORT_SYMBOL_GPL(call_micmute_led_update); static void update_micmute_led(struct hda_codec *codec, struct snd_kcontrol *kcontrol, diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index fb9f1a90238b..9f08a9ee9112 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -353,6 +353,7 @@ unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, void snd_hda_gen_stream_pm(struct hda_codec *codec, hda_nid_t nid, bool on); int snd_hda_gen_fix_pin_power(struct hda_codec *codec, hda_nid_t pin); +void call_micmute_led_update(struct hda_codec *codec); int snd_hda_gen_add_micmute_led(struct hda_codec *codec, void (*hook)(struct hda_codec *)); void snd_hda_gen_fixup_micmute_led(struct hda_codec *codec, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d73f8beadb6..9691af8241cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4114,10 +4114,10 @@ static int micmute_led_set(struct led_classdev *led_cdev, enum led_brightness brightness) { struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent); - struct alc_spec *spec = codec->spec; + struct hda_gen_spec *spec = codec->spec; - alc_update_gpio_led(codec, spec->gpio_mic_led_mask, - spec->micmute_led_polarity, !!brightness); + spec->micmute_led.led_mode = !brightness; + call_micmute_led_update(codec); return 0; } @@ -4126,7 +4126,17 @@ static struct led_classdev micmute_led_cdev = { .max_brightness = 1, .brightness_set_blocking = micmute_led_set, .default_trigger = "audio-micmute", + .flags = LED_CORE_SUSPENDRESUME, }; + +static void alc_register_micmute_led(struct hda_codec *codec) +{ + micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + if (devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev)) + codec_warn(codec, "failed to register micmute LED\n"); +} +#else +static inline void alc_register_micmute_led(struct hda_codec *codec) {} #endif /* setup mute and mic-mute GPIO bits, add hooks appropriately */ @@ -4136,9 +4146,6 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, unsigned int micmute_mask) { struct alc_spec *spec = codec->spec; -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - int err; -#endif alc_fixup_gpio(codec, action, mute_mask | micmute_mask); @@ -4151,13 +4158,7 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, if (micmute_mask) { spec->gpio_mic_led_mask = micmute_mask; snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); - -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); - err = devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev); - if (err) - codec_warn(codec, "failed to register micmute LED\n"); -#endif + alc_register_micmute_led(codec); } } @@ -4305,6 +4306,7 @@ static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec, spec->mic_led_coefbit_on = 1<<13; spec->mic_led_coefbit_off = 0; snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + alc_register_micmute_led(codec); } } @@ -4319,6 +4321,7 @@ static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec, spec->mic_led_coefbit_on = 2<<2; spec->mic_led_coefbit_off = 1<<2; snd_hda_gen_add_micmute_led(codec, alc_hp_cap_micmute_update); + alc_register_micmute_led(codec); } } -- 2.17.1

2 2

[PATCH v2 1/2] ASoC: bindings: fsl_spdif: Add new compatible string for imx6sx
by Shengjiu Wang 17 Jun '20

17 Jun '20

Add new compatible string "fsl,imx6sx-spdif" in the binding document. And add compatible string "fsl,vf610-spdif" which was missed before. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.txt | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index 8b324f82a782..e1365b0ee1e9 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -6,7 +6,11 @@ a fibre cable. Required properties: - - compatible : Compatible list, must contain "fsl,imx35-spdif". + - compatible : Compatible list, should contain one of the following + compatibles: + "fsl,imx35-spdif", + "fsl,vf610-spdif", + "fsl,imx6sx-spdif", - reg : Offset and length of the register set for the device. -- 2.21.0

3 3

alsa-gobject: timer/seq: use const qualifier to self argument for getter API
by GitHub pull_request - opened 17 Jun '20

17 Jun '20

alsa-project/alsa-gobject pull request #40 was opened from takaswie: In ALSATimer and ALSASeq, some methods have prototype such that self argument don't have const qualifier even if it's getter method. This patchset adds const qualifier to the methods. Request URL : https://github.com/alsa-project/alsa-gobject/pull/40 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/40.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 1/2] ASoC: bindings: fsl_spdif: Add new compatible string for imx6sx
by Shengjiu Wang 17 Jun '20

17 Jun '20

Add new compatible string "fsl,imx6sx-spdif" in the binding document. And add compatible string "fsl,vf610-spdif" which was missed before. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,spdif.txt | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,spdif.txt b/Documentation/devicetree/bindings/sound/fsl,spdif.txt index 8b324f82a782..e1365b0ee1e9 100644 --- a/Documentation/devicetree/bindings/sound/fsl,spdif.txt +++ b/Documentation/devicetree/bindings/sound/fsl,spdif.txt @@ -6,7 +6,11 @@ a fibre cable. Required properties: - - compatible : Compatible list, must contain "fsl,imx35-spdif". + - compatible : Compatible list, should contain one of the following + compatibles: + "fsl,imx35-spdif", + "fsl,vf610-spdif", + "fsl,imx6sx-spdif", - reg : Offset and length of the register set for the device. -- 2.21.0

3 3

[PATCH 1/2] ASoC: bindings: fsl-asoc-card: Add compatible string for MQS
by Shengjiu Wang 17 Jun '20

17 Jun '20

Add compatible string "fsl,imx-audio-mqs" for MQS Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index c60a5732d29c..6417c5dffa9a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -34,6 +34,8 @@ The compatible list for this generic sound card currently: "fsl,imx-audio-wm8960" + "fsl,imx-audio-mqs" + Required properties: - compatible : Contains one of entries in the compatible list. -- 2.21.0

3 3

[PATCH alsa-lib] control: ctlparse - use type-specific bound on element count
by John Keeping 16 Jun '20

16 Jun '20

Using a fixed bound of 128 means that too many values may be set for an INTEGER64 type and that any elements past 128 are out of reach for BYTE type controls. Derive the maximum number of elements from the type so that the full range is parsed for all types. Signed-off-by: John Keeping <john(a)metanate.com> --- src/control/ctlparse.c | 24 +++++++++++++++++++++++- 1 file changed, 23 insertions(+), 1 deletion(-) diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 1ac31bb9..ee1e0602 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -304,6 +304,25 @@ static int get_ctl_enum_item_index(snd_ctl_t *handle, return -1; } +static unsigned int get_ctl_type_max_elements(snd_ctl_elem_type_t type) +{ + struct snd_ctl_elem_value value; + + switch (type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + case SND_CTL_ELEM_TYPE_INTEGER: + return ARRAY_SIZE(value.value.integer.value); + case SND_CTL_ELEM_TYPE_INTEGER64: + return ARRAY_SIZE(value.value.integer64.value); + case SND_CTL_ELEM_TYPE_ENUMERATED: + return ARRAY_SIZE(value.value.enumerated.item); + case SND_CTL_ELEM_TYPE_BYTES: + return ARRAY_SIZE(value.value.bytes.data); + default: + return 0; + } +} + /** * \brief parse ASCII string as CTL element value * \param handle CTL handle @@ -331,8 +350,11 @@ int snd_ctl_ascii_value_parse(snd_ctl_t *handle, type = snd_ctl_elem_info_get_type(info); count = snd_ctl_elem_info_get_count(info); snd_ctl_elem_value_set_id(dst, &myid); + + if (count > get_ctl_type_max_elements(type)) + count = get_ctl_type_max_elements(type); - for (idx = 0; idx < count && idx < 128 && ptr && *ptr; idx++) { + for (idx = 0; idx < count && ptr && *ptr; idx++) { if (*ptr == ',') goto skip; switch (type) { -- 2.27.0

2 1

[RESENDx2][PATCH] sound: qcom: Kconfig: Tweak dependencies on SND_SOC_SDM845
by John Stultz 16 Jun '20

16 Jun '20

CROS_EC isn't strictly required for audio to work on other SDM845 platforms (like the Dragonboard 845c). So lets remove the dependency and select the related CROS_EC options if CROS_EC is already enabled. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Rohit kumar <rohitkr(a)codeaurora.org> Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: John Stultz <john.stultz(a)linaro.org> --- sound/soc/qcom/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index f51b28d1b94d..0434f2c7d746 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -99,12 +99,12 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR && CROS_EC && I2C && SOUNDWIRE + depends on QCOM_APR && I2C && SOUNDWIRE select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 select SND_SOC_MAX98927 - select SND_SOC_CROS_EC_CODEC + select SND_SOC_CROS_EC_CODEC if CROS_EC help To add support for audio on Qualcomm Technologies Inc. SDM845 SoC-based systems. -- 2.17.1

3 4

pcm: dmic: assembly: add x32 support.
by GitHub pull_request - opened 16 Jun '20

16 Jun '20

alsa-project/alsa-lib pull request #61 was opened from hvdijk: x32 is the x86_64 ABI that uses 32-bit pointers, so requires loading addresses into edi/esi/ebx rather than rdi/rsi/rbx. Note that instructions such as movl %eax, (%rdi) do not require updating, as loading an address into %edi zeroes the high bits, causing the full %rdi register to hold a valid address. Signed-off-by: Harald van Dijk <harald(a)gigawatt.nl> Without this, even aplay segfaults when building with `-mx32`. Request URL : https://github.com/alsa-project/alsa-lib/pull/61 Patch URL : https://github.com/alsa-project/alsa-lib/pull/61.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] soundwire: Replace 'objs' by 'y'
by Vinod Koul 16 Jun '20

16 Jun '20

`-objs` is designed for building host programs, change to `-y`, more straightforward for device drivers. See Documentation/kbuild/makefiles.rst Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- drivers/soundwire/Makefile | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/Makefile b/drivers/soundwire/Makefile index b5871612613b..7c53ffae9f50 100644 --- a/drivers/soundwire/Makefile +++ b/drivers/soundwire/Makefile @@ -4,22 +4,22 @@ # #Bus Objs -soundwire-bus-objs := bus_type.o bus.o master.o slave.o mipi_disco.o stream.o \ +soundwire-bus-y := bus_type.o bus.o master.o slave.o mipi_disco.o stream.o \ sysfs_slave.o sysfs_slave_dpn.o obj-$(CONFIG_SOUNDWIRE) += soundwire-bus.o ifdef CONFIG_DEBUG_FS -soundwire-bus-objs += debugfs.o +soundwire-bus-y += debugfs.o endif #Cadence Objs -soundwire-cadence-objs := cadence_master.o +soundwire-cadence-y := cadence_master.o obj-$(CONFIG_SOUNDWIRE_CADENCE) += soundwire-cadence.o #Intel driver -soundwire-intel-objs := intel.o intel_init.o +soundwire-intel-y := intel.o intel_init.o obj-$(CONFIG_SOUNDWIRE_INTEL) += soundwire-intel.o #Qualcomm driver -soundwire-qcom-objs := qcom.o +soundwire-qcom-y := qcom.o obj-$(CONFIG_SOUNDWIRE_QCOM) += soundwire-qcom.o -- 2.26.2

1 0

[PATCH] ALSA: hda/realtek - Add quirk for MSI GE63 laptop
by Takashi Iwai 16 Jun '20

16 Jun '20

MSI GE63 laptop with ALC1220 codec requires the very same quirk (ALC1220_FIXUP_CLEVO_P950) as other MSI devices for the proper sound output. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208057 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d1b67d75f954..3ca93421e209 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2461,6 +2461,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1458, 0xa0cd, "Gigabyte X570 Aorus Master", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), -- 2.25.0

1 0

[PATCH] ALSA: usb-audio: Fix potential use-after-free of streams
by Takashi Iwai 16 Jun '20

16 Jun '20

With the recent full-duplex support of implicit feedback streams, an endpoint can be still running after closing the capture stream as long as the playback stream with the sync-endpoint is running. In such a state, the URBs are still be handled and they may call retire_data_urb callback, which tries to transfer the data from the PCM buffer. Since the PCM stream gets closed, this may lead to use-after-free. This patch adds the proper clearance of the callback at stopping the capture stream for addressing the possible UAF above. Fixes: 10ce77e4817f ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 84c0ae431936..a777d36c4f5a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1787,6 +1787,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream return 0; case SNDRV_PCM_TRIGGER_STOP: stop_endpoints(subs); + subs->data_endpoint->retire_data_urb = NULL; subs->running = 0; return 0; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -- 2.25.0

1 0

[PATCH v2] ASoC: rt1015: Flush DAC data before playback.
by jack.yu＠realtek.com 16 Jun '20

16 Jun '20

From: Jack Yu <jack.yu(a)realtek.com> Flush DAC data before playback. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 18 ++++++++++++++---- sound/soc/codecs/rt1015.h | 5 +++++ 2 files changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 2cccb310fa96..1f61b4aa4353 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -493,7 +493,7 @@ static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, if (!rt1015->dac_is_used) { rt1015->bypass_boost = ucontrol->value.integer.value[0]; - if (rt1015->bypass_boost == 1) { + if (rt1015->bypass_boost == RT1015_Bypass_Boost) { snd_soc_component_write(component, RT1015_PWR4, 0x00b2); snd_soc_component_write(component, @@ -549,7 +549,7 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: rt1015->dac_is_used = 1; - if (rt1015->bypass_boost == 0) { + if (rt1015->bypass_boost == RT1015_Enable_Boost) { snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f7); snd_soc_component_write(component, @@ -566,8 +566,17 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, } break; + case SND_SOC_DAPM_POST_PMU: + if (rt1015->bypass_boost == RT1015_Bypass_Boost) { + regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x00a8); + regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x0597); + regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x05f7); + regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x0028); + } + break; + case SND_SOC_DAPM_POST_PMD: - if (rt1015->bypass_boost == 0) { + if (rt1015->bypass_boost == RT1015_Enable_Boost) { snd_soc_component_write(component, RT1015_PWR9, 0xa800); snd_soc_component_write(component, @@ -617,7 +626,8 @@ static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("DAC", NULL, RT1015_PWR1, RT1015_PWR_DAC_BIT, 0, - r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUTPUT("SPO"), }; diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 8169962935a5..7bd159e8f958 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -368,6 +368,11 @@ enum { FIXED_ADAPTIVE, }; +enum { + RT1015_Enable_Boost = 0, + RT1015_Bypass_Boost, +}; + struct rt1015_priv { struct snd_soc_component *component; struct regmap *regmap; -- 2.25.1

2 1

[PATCH v3] ASoC: fsl_ssi: Fix bclk calculation for mono channel
by Shengjiu Wang 16 Jun '20

16 Jun '20

For mono channel, SSI will switch to Normal mode. In Normal mode and Network mode, the Word Length Control bits control the word length divider in clock generator, which is different with I2S Master mode (the word length is fixed to 32bit), it should be the value of params_width(hw_params). The condition "slots == 2" is not good for I2S Master mode, because for Network mode and Normal mode, the slots can also be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) to check if it is I2S Master mode. So we refine the formula for mono channel, otherwise there will be sound issue for S24_LE. Fixes: b0a7043d5c2c ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3 - update according to Nicolin's comments changes in v2 - refine patch for Network mode and Normal mode. sound/soc/fsl/fsl_ssi.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bad89b0d129e..1a2fa7f18142 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -678,8 +678,9 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct regmap *regs = ssi->regs; u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; unsigned long clkrate, baudrate, tmprate; - unsigned int slots = params_channels(hw_params); - unsigned int slot_width = 32; + unsigned int channels = params_channels(hw_params); + unsigned int slot_width = params_width(hw_params); + unsigned int slots = 2; u64 sub, savesub = 100000; unsigned int freq; bool baudclk_is_used; @@ -688,10 +689,14 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, /* Override slots and slot_width if being specifically set... */ if (ssi->slots) slots = ssi->slots; - /* ...but keep 32 bits if slots is 2 -- I2S Master mode */ - if (ssi->slot_width && slots != 2) + if (ssi->slot_width) slot_width = ssi->slot_width; + /* ...but force 32 bits for stereo audio using I2S Master Mode */ + if (channels == 2 && + (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) == SSI_SCR_I2S_MODE_MASTER) + slot_width = 32; + /* Generate bit clock based on the slot number and slot width */ freq = slots * slot_width * params_rate(hw_params); -- 2.21.0

3 2

[PATCH 1/2] ALSA: hda: Make codec controlled LED support more generic
by Kai-Heng Feng 16 Jun '20

16 Jun '20

Currently, only HDA codec GPIO controlled LED class is supported, and only via platform specific quirk. There are systems that control LED via COEF instead of GPIO, and to support those systems, move the LED class registration to snd_hda_gen_add_micmute_led(), so all systems can facilitate the same interface. In addition to that, add LED_CORE_SUSPENDRESUME flag since some systems don't restore the LED properly after suspend. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_generic.c | 28 ++++++++++++++++++++++++++++ sound/pci/hda/patch_realtek.c | 30 ------------------------------ 2 files changed, 28 insertions(+), 30 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f4e9d9445e18..4242407734c0 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4006,6 +4006,28 @@ static const struct snd_kcontrol_new micmute_led_mode_ctl = { * * Returns 0 if the hook is established or a negative error code. */ + +#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) +static int micmute_led_set(struct led_classdev *led_cdev, + enum led_brightness brightness) +{ + struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent); + struct hda_gen_spec *spec = codec->spec; + + spec->micmute_led.led_mode = !brightness; + call_micmute_led_update(codec); + return 0; +} + +static struct led_classdev micmute_led_cdev = { + .name = "hda::micmute", + .max_brightness = 1, + .brightness_set_blocking = micmute_led_set, + .default_trigger = "audio-micmute", + .flags = LED_CORE_SUSPENDRESUME, +}; +#endif + int snd_hda_gen_add_micmute_led(struct hda_codec *codec, void (*hook)(struct hda_codec *)) { @@ -4019,6 +4041,12 @@ int snd_hda_gen_add_micmute_led(struct hda_codec *codec, spec->cap_sync_hook = update_micmute_led; if (!snd_hda_gen_add_kctl(spec, NULL, &micmute_led_mode_ctl)) return -ENOMEM; + +#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) + micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + if (devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev)) + codec_warn(codec, "failed to register micmute LED\n"); +#endif return 0; } EXPORT_SYMBOL_GPL(snd_hda_gen_add_micmute_led); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d73f8beadb6..cead44a6c6cd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4109,26 +4109,6 @@ static void alc_gpio_micmute_update(struct hda_codec *codec) spec->gen.micmute_led.led_value); } -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) -static int micmute_led_set(struct led_classdev *led_cdev, - enum led_brightness brightness) -{ - struct hda_codec *codec = dev_to_hda_codec(led_cdev->dev->parent); - struct alc_spec *spec = codec->spec; - - alc_update_gpio_led(codec, spec->gpio_mic_led_mask, - spec->micmute_led_polarity, !!brightness); - return 0; -} - -static struct led_classdev micmute_led_cdev = { - .name = "hda::micmute", - .max_brightness = 1, - .brightness_set_blocking = micmute_led_set, - .default_trigger = "audio-micmute", -}; -#endif - /* setup mute and mic-mute GPIO bits, add hooks appropriately */ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, int action, @@ -4136,9 +4116,6 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, unsigned int micmute_mask) { struct alc_spec *spec = codec->spec; -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - int err; -#endif alc_fixup_gpio(codec, action, mute_mask | micmute_mask); @@ -4151,13 +4128,6 @@ static void alc_fixup_hp_gpio_led(struct hda_codec *codec, if (micmute_mask) { spec->gpio_mic_led_mask = micmute_mask; snd_hda_gen_add_micmute_led(codec, alc_gpio_micmute_update); - -#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) - micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); - err = devm_led_classdev_register(&codec->core.dev, &micmute_led_cdev); - if (err) - codec_warn(codec, "failed to register micmute LED\n"); -#endif } } -- 2.17.1

2 3

low volume libalsa 1.2.3
by GitHub issues - edited 16 Jun '20

16 Jun '20

alsa-project/alsa-lib issue #60 was edited from texstar: Sound is about 30% lower with alsa 1.2.3 than with alsa 1.2.2 using the same volume setting. Backing down to 1.2.2 to restore order as the natives are getting stabby. Issue URL : https://github.com/alsa-project/alsa-lib/issues/60 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

low volume libalsa 1.2.3
by GitHub issues - opened 16 Jun '20

16 Jun '20

alsa-project/alsa-lib issue #60 was opened from texstar: Sound is about 30% lower with alsa 1.2.3 than with alsa 1.2.2 using the same volume settings. Backing down to 1.2.2 to restore order as the natives are getting stabby. Issue URL : https://github.com/alsa-project/alsa-lib/issues/60 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] Add implicit feedback quirk for SSL2.
by Laurence Tratt 16 Jun '20

16 Jun '20

As expected, this requires the same quirk as the SSL2+ in order for the clock to sync. This was suggested by, and tested on an SSL2, by Dmitry. Suggested-by: Dmitry <dpavlushko(a)gmail.com> Signed-off-by: Laurence Tratt <laurie(a)tratt.net> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git sound/usb/pcm.c sound/usb/pcm.c index 84c0ae4319..dc1608bdf6 100644 --- sound/usb/pcm.c +++ sound/usb/pcm.c @@ -367,6 +367,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 0; goto add_sync_ep_from_ifnum; case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ ep = 0x81; ifnum = 2; -- 2.27.0

1 0

[PATCH] Add implicit feedback quirk for SSL2.
by Laurence Tratt 16 Jun '20

16 Jun '20

As expected, this requires the same quirk as the SSL2+ in order for the clock to sync. This was suggested by, and tested on an SSL2, by Dmitry. Signed-off-by: Dmitry <dpavlushko(a)gmail.com> Signed-off-by: Laurence Tratt <laurie(a)tratt.net> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git sound/usb/pcm.c sound/usb/pcm.c index 84c0ae4319..dc1608bdf6 100644 --- sound/usb/pcm.c +++ sound/usb/pcm.c @@ -367,6 +367,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 0; goto add_sync_ep_from_ifnum; case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ ep = 0x81; ifnum = 2; -- 2.27.0

1 0

[PATCH v2] ASoC: fsl_ssi: Fix bclk calculation for mono channel
by Shengjiu Wang 16 Jun '20

16 Jun '20

For mono channel, SSI will switch to Normal mode. In Normal mode and Network mode, the Word Length Control bits control the word length divider in clock generator, which is different with I2S Master mode (the word length is fixed to 32bit), it should be the value of params_width(hw_params). The condition "slots == 2" is not good for I2S Master mode, because for Network mode and Normal mode, the slots can also be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) to check if it is I2S Master mode. So we refine the famula for mono channel, otherwise there will be sound issue for S24_LE. Fixes: b0a7043d5c2c ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - refine patch for Network mode and Normal mode. sound/soc/fsl/fsl_ssi.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bad89b0d129e..cbf67d132fda 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -678,7 +678,8 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, struct regmap *regs = ssi->regs; u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; unsigned long clkrate, baudrate, tmprate; - unsigned int slots = params_channels(hw_params); + unsigned int channels = params_channels(hw_params); + unsigned int slots; unsigned int slot_width = 32; u64 sub, savesub = 100000; unsigned int freq; @@ -688,9 +689,15 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, /* Override slots and slot_width if being specifically set... */ if (ssi->slots) slots = ssi->slots; - /* ...but keep 32 bits if slots is 2 -- I2S Master mode */ - if (ssi->slot_width && slots != 2) - slot_width = ssi->slot_width; + else + /* Apply two slots for mono channel, because DC = 2 */ + slots = (channels == 1) ? 2 : channels; + + /* ...but keep 32 bits if I2S Master mode */ + if ((ssi->i2s_net & SSI_SCR_I2S_MODE_MASK) != SSI_SCR_I2S_MODE_MASTER || + channels == 1) + slot_width = ssi->slot_width ? ssi->slot_width : + params_width(hw_params); /* Generate bit clock based on the slot number and slot width */ freq = slots * slot_width * params_rate(hw_params); -- 2.21.0

3 4

[PATCH 0/3] ASoC: fsl_easrc: Fix several warnings
by Shengjiu Wang 16 Jun '20

16 Jun '20

Fix several warnings with "make W=1" Shengjiu Wang (3): ASoC: fsl_easrc: Fix -Wmissing-prototypes warning ASoC: fsl_easrc: Fix -Wunused-but-set-variable ASoC: fsl_easrc: Fix "Function parameter not described" warnings sound/soc/fsl/fsl_easrc.c | 42 +++++++++++++++++---------------------- 1 file changed, 18 insertions(+), 24 deletions(-) -- 2.21.0

3 7

[PATCH v2 0/2] ASoC: mediatek: mt6358: support DMIC one-wire mode
by Jiaxin Yu 16 Jun '20

16 Jun '20

v2 changes: 1. Uses a DT property to select DMIC mode instead of a mixer control. v1 changes: 1. Uses a mixer control to select DMIC mode. 2. patchwork list: https://patchwork.kernel.org/patch/11578309 Jiaxin Yu (2): ASoC: mediatek: mt6358: support DMIC one-wire mode ASoC: dt-bindings: mediatek: mt6358: add dmic-mode property Documentation/devicetree/bindings/sound/mt6358.txt | 6 ++++++ sound/soc/codecs/mt6358.c | 23 +++++++++++++++++++++- 2 files changed, 28 insertions(+), 1 deletion(-) -- 1.8.1.1.dirty

3 4

[PATCH] ASoC: img-parallel-out: Fix a reference count leak
by wu000273＠umn.edu 16 Jun '20

16 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> pm_runtime_get_sync() increments the runtime PM usage counter even when it returns an error code, causing incorrect ref count if pm_runtime_put_noidle() is not called in error handling paths. Thus call pm_runtime_put_noidle() if pm_runtime_get_sync() fails. Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> --- sound/soc/img/img-parallel-out.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index 5ddbe3a31c2e..4da49a42e854 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -163,8 +163,10 @@ static int img_prl_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) } ret = pm_runtime_get_sync(prl->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(prl->dev); return ret; + } reg = img_prl_out_readl(prl, IMG_PRL_OUT_CTL); reg = (reg & ~IMG_PRL_OUT_CTL_EDGE_MASK) | control_set; -- 2.17.1

2 1

[V7 PATCH] dt-bindings: Added device tree binding for max98390
by Steve Lee 16 Jun '20

16 Jun '20

Add DT binding of max98390 amplifier driver. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- Changed since V6: * Re-confirm yaml dt binding check * Add minimum and maximum value for each temperature_calib and r0_calib * Add maxim prefix for naming. Changed since V5: * Change txt to yaml and fix up the examples. Changed since V4: * No changes. Changed since V3: * No changes. Changed since V2: * No changes. Changed since V1: * Modified sample text in example .../bindings/sound/maxim,max98390.yaml | 51 +++++++++++++++++++ 1 file changed, 51 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98390.yaml diff --git a/Documentation/devicetree/bindings/sound/maxim,max98390.yaml b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml new file mode 100644 index 000000000000..e5ac35280da3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98390.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98390.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98390 Speaker Amplifier with Integrated Dynamic Speaker Management + +maintainers: + - Steve Lee <steves.lee(a)maximintegrated.com> + +properties: + compatible: + const: maxim,max98390 + + reg: + maxItems: 1 + description: I2C address of the device. + + maxim,temperature_calib: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + description: The calculated temperature data was measured while doing the calibration. + minimum: 0 + maximum: 65535 + + maxim,r0_calib: + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + description: This is r0 calibration data which was measured in factory mode. + minimum: 1 + maximum: 8388607 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98390: amplifier@38 { + compatible = "maxim,max98390"; + reg = <0x38>; + maxim,temperature_calib = <1024>; + maxim,r0_calib = <100232>; + }; + }; -- 2.17.1

3 2

[PATCH 1/2] ASoC: bindings: wm8960: Add property for headphone detection
by Shengjiu Wang 16 Jun '20

16 Jun '20

Add two properties for headphone detection. wlf,hp-cfg: A list of headphone jack detect configuration register values wlf,gpio-cfg: A list of GPIO configuration register values Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/wm8960.txt | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt index 6d29ac3750ee..85d3b287108c 100644 --- a/Documentation/devicetree/bindings/sound/wm8960.txt +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -21,6 +21,17 @@ Optional properties: enabled and disabled together with HP_L and HP_R pins in response to jack detect events. + - wlf,hp-cfg: A list of headphone jack detect configuration register values. + The list must be 3 entries long. + hp-cfg[0]: HPSEL[1:0] of R48 (Additional Control 4). + hp-cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2). + hp-cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1). + + - wlf,gpio-cfg: A list of GPIO configuration register values. + The list must be 2 entries long. + gpio-cfg[0]: ALRCGPIO of R9 (Audio interface) + gpio-cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4). + Example: wm8960: codec@1a { -- 2.21.0

3 3

[PATCH 0/2] ASoC: max98357a: support MAX98360A in OF
by Tzung-Bi Shih 16 Jun '20

16 Jun '20

Commit 1a0f2433d738 ("ASoC: max98357a: Add ACPI HID MAX98360A") supports MAX98360A in ACPI world. This series supports MAX98360A in OF world. Tzung-Bi Shih (2): ASoC: max98357a: add compatible string for MAX98360A ASoC: dt-bindings: add compatible string for MAX98360A .../devicetree/bindings/sound/max98357a.txt | 12 +++++++++--- sound/soc/codecs/max98357a.c | 1 + 2 files changed, 10 insertions(+), 3 deletions(-) -- 2.27.0.278.ge193c7cf3a9-goog

3 4

[PATCH v3 0/3] ASoC: Intel: Add KeemBay ASoC platform driver
by Sia Jee Heng 16 Jun '20

16 Jun '20

The below series of patches support the KeemBay ASoC platform driver. The platform driver initialize the i2s to capture and playback the pcm data on the ARM. The i2s is running in polling mode. There is no DSP in the KeemBay SoC. Users are rely on the Gstreamer plugin to perform Audio preprocessing. Audio graph card is used to connect the platform driver with the tlv320aic3204 codec. Change History: v3: - Adjusted header format. - Use Audio graph card instead of custom sound card. - Use if-else instead of conditional operator. - Enabled .set_fmt to configure master clock. v2: - Corrected I2S naming for DT binding. v1: - Initial version. Sia Jee Heng (3): ASoC: Intel: Add KeemBay platform driver ASoC: Intel: Add makefiles and kconfig changes for KeemBay dt-bindings: sound: Add documentation for KeemBay i2s .../bindings/sound/intel,keembay-i2s.yaml | 68 +++ sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/keembay/Makefile | 4 + sound/soc/intel/keembay/kmb_platform.c | 667 +++++++++++++++++++++ sound/soc/intel/keembay/kmb_platform.h | 145 +++++ 6 files changed, 892 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml create mode 100644 sound/soc/intel/keembay/Makefile create mode 100644 sound/soc/intel/keembay/kmb_platform.c create mode 100644 sound/soc/intel/keembay/kmb_platform.h -- 1.9.1

4 8

[PATCH 1/5] ASoC: rt5682: Enable Vref2 under using PLL2
by derek.fang＠realtek.com 16 Jun '20

16 Jun '20

From: "derek.fang" <derek.fang(a)realtek.com> Enable Vref2 under long term using PLL2 to avoid clock unstable. Signed-off-by: derek.fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 24 +++++++++++++++++------- 1 file changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index d324512..ca8e1f2 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -960,13 +960,12 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); - else + RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); + RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); @@ -1601,8 +1600,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), /* ASRC */ @@ -2485,6 +2483,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw) snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, RT5682_PWR_MB); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2"); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2, + RT5682_PWR_VREF2); + usleep_range(55000, 60000); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_FV2, RT5682_PWR_FV2); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B"); @@ -2510,9 +2517,12 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw) snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2"); if (!rt5682->jack_type) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_FV2 | RT5682_PWR_MB, 0); + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F"); snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B"); -- 2.7.4

2 6

[PATCH v5 0/3] ASoC: Intel: Add KeemBay ASoC platform driver
by Sia Jee Heng 16 Jun '20

16 Jun '20

The below series of patches support the KeemBay ASoC platform driver. The platform driver initialize the i2s to capture and playback the pcm data on the ARM. The i2s is running in polling mode. There is no DSP in the KeemBay SoC. Users are rely on the Gstreamer plugin to perform Audio preprocessing. Audio graph card is used to connect the platform driver with the tlv320aic3204 codec. Change History: v5: - Remove OF dependency from Kconfig as OF is shifted to audio graph card. v4: - Reduce if-otology at the tx/rx function. - Fix indentation. - specify .rate directly v3: - Adjusted header format. - Use Audio graph card instead of custom sound card. - Use if-else instead of conditional operator. - Enabled .set_fmt to configure master clock. v2: - Corrected I2S naming for DT binding. v1: - Initial version. Sia Jee Heng (3): ASoC: Intel: Add KeemBay platform driver ASoC: Intel: Add makefiles and kconfig changes for KeemBay dt-bindings: sound: Add documentation for KeemBay i2s .../bindings/sound/intel,keembay-i2s.yaml | 68 +++ sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/keembay/Makefile | 4 + sound/soc/intel/keembay/kmb_platform.c | 654 +++++++++++++++++++++ sound/soc/intel/keembay/kmb_platform.h | 145 +++++ 6 files changed, 879 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml create mode 100644 sound/soc/intel/keembay/Makefile create mode 100644 sound/soc/intel/keembay/kmb_platform.c create mode 100644 sound/soc/intel/keembay/kmb_platform.h -- 1.9.1

2 4

[v3] ASoC: AMD: Use mixer control to switch between DMICs
by Akshu Agrawal 16 Jun '20

16 Jun '20

Having mixer control to switch between DMICs prevents user to initiate capture simultaneously on both the DMIcs. Earlier 2 separate devices, one for each DMIC, gave an option of using them simultaneously, which is not supported. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Modified "Front Mic" to "DMIC Switch" v3: Changed to using of Mux sound/soc/amd/acp3x-rt5682-max9836.c | 58 +++++++++++++++------------- 1 file changed, 31 insertions(+), 27 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index e499c00e0c66..f745b42dfd23 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -188,25 +188,27 @@ static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_BT_INSTANCE; snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (dmic_sel) - gpiod_set_value(dmic_sel, 0); return rt5682_clk_enable(substream); } -static int acp3x_ec_dmic1_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); +static int dmic_switch; - machine->cap_i2s_instance = I2S_BT_INSTANCE; - snd_soc_dai_set_bclk_ratio(codec_dai, 64); - if (dmic_sel) - gpiod_set_value(dmic_sel, 1); +static int dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = dmic_switch; + return 0; +} - return rt5682_clk_enable(substream); +static int dmic_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (dmic_sel) { + dmic_switch = ucontrol->value.integer.value[0]; + gpiod_set_value(dmic_sel, dmic_switch); + } + return 0; } static void rt5682_shutdown(struct snd_pcm_substream *substream) @@ -229,11 +231,6 @@ static const struct snd_soc_ops acp3x_ec_cap0_ops = { .shutdown = rt5682_shutdown, }; -static const struct snd_soc_ops acp3x_ec_cap1_ops = { - .startup = acp3x_ec_dmic1_startup, - .shutdown = rt5682_shutdown, -}; - SND_SOC_DAILINK_DEF(acp3x_i2s, DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.0"))); SND_SOC_DAILINK_DEF(acp3x_bt, @@ -279,21 +276,26 @@ static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { .ops = &acp3x_ec_cap0_ops, SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), }, - { - .name = "acp3x-ec-dmic1-capture", - .stream_name = "Capture DMIC1", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBS_CFS, - .dpcm_capture = 1, - .ops = &acp3x_ec_cap1_ops, - SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), - }, }; +static const char * const dmic_mux_text[] = { + "Front Mic", + "Rear Mic", +}; + +static SOC_ENUM_SINGLE_DECL( + acp3x_dmic_enum, SND_SOC_NOPM, 0, dmic_mux_text); + +static const struct snd_kcontrol_new acp3x_dmic_mux_control = + SOC_DAPM_ENUM_EXT("DMIC Select Mux", acp3x_dmic_enum, + dmic_get, dmic_set); + static const struct snd_soc_dapm_widget acp3x_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MUX("Dmic Mux", SND_SOC_NOPM, 0, 0, + &acp3x_dmic_mux_control), }; static const struct snd_soc_dapm_route acp3x_audio_route[] = { @@ -301,6 +303,8 @@ static const struct snd_soc_dapm_route acp3x_audio_route[] = { {"Headphone Jack", NULL, "HPOR"}, {"IN1P", NULL, "Headset Mic"}, {"Spk", NULL, "Speaker"}, + {"Dmic Mux", "Front Mic", "DMIC"}, + {"Dmic Mux", "Rear Mic", "DMIC"}, }; static const struct snd_kcontrol_new acp3x_mc_controls[] = { -- 2.20.1

2 1

[PATCH 0/3] ASoC: improve core dmesg logs and verbosity
by Pierre-Louis Bossart 16 Jun '20

16 Jun '20

Try to both reduce useless verbosity and keep useful error reports. Pierre-Louis Bossart (3): ASoC: soc-core: reduce verbosity of BE override message ASoC: soc-pcm: improve error messages in soc_pcm_new() ASoC: soc-pcm/compress: reduce verbosity on mapping ok messages sound/soc/soc-compress.c | 4 ++-- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-pcm.c | 13 +++++++------ 3 files changed, 11 insertions(+), 10 deletions(-) -- 2.20.1

2 4

[PATCH] ASoC: img: Fix a reference count leak in img_i2s_in_set_fmt
by wu000273＠umn.edu 16 Jun '20

16 Jun '20

From: Qiushi Wu <wu000273(a)umn.edu> pm_runtime_get_sync() increments the runtime PM usage counter even when it returns an error code, causing incorrect ref count if pm_runtime_put_noidle() is not called in error handling paths. Thus call pm_runtime_put_noidle() if pm_runtime_get_sync() fails. Signed-off-by: Qiushi Wu <wu000273(a)umn.edu> --- sound/soc/img/img-i2s-in.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index e30b66b94bf6..0843235d73c9 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -343,8 +343,10 @@ static int img_i2s_in_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) chan_control_mask = IMG_I2S_IN_CH_CTL_CLK_TRANS_MASK; ret = pm_runtime_get_sync(i2s->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(i2s->dev); return ret; + } for (i = 0; i < i2s->active_channels; i++) img_i2s_in_ch_disable(i2s, i); -- 2.17.1

2 1

[PATCH v2 00/12] ASoC: soc-component: collect component functions
by Kuninori Morimoto 16 Jun '20

16 Jun '20

Hi Mark We have soc-component.c now, but still many component related functions are implemented many place. This patch-set collect these into soc-component.c. v1 -> v2 - remove soc-compress.c exchange (But I have plan to repost it) - fixup loop break issue on some functions - direct return on some functions Link: https://lore.kernel.org/r/87a71nzhy2.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (12): ASoC: soc-component: add soc_component_pin() and share code ASoC: soc-component: move snd_soc_component_xxx_regmap() to soc-component ASoC: soc-component: move snd_soc_component_initialize() to soc-component.c ASoC: soc-component: add soc_component_err() ASoC: soc-component: add snd_soc_pcm_component_prepare() ASoC: soc-component: add snd_soc_pcm_component_hw_params() ASoC: soc-component: add snd_soc_pcm_component_hw_free() ASoC: soc-component: add snd_soc_pcm_component_trigger() ASoC: soc-component: add snd_soc_component_init() ASoC: soc-component: merge soc-io.c into soc-component.c ASoC: soc-component: merge soc_pcm_trigger_start/stop() ASoC: soc-component: tidyup Copyright include/sound/soc-component.h | 29 +- sound/soc/Makefile | 2 +- sound/soc/soc-component.c | 666 ++++++++++++++++++++++++---------- sound/soc/soc-core.c | 102 +----- sound/soc/soc-io.c | 202 ----------- sound/soc/soc-pcm.c | 114 ++---- 6 files changed, 531 insertions(+), 584 deletions(-) delete mode 100644 sound/soc/soc-io.c -- 2.17.1

3 14

[PATCH 1/3] dt-bindings: sound: Document wm8994 endpoints
by Jonathan Bakker 16 Jun '20

16 Jun '20

The wm8994 exposes several inputs and outputs that can be used by machine drivers in their routing. Add them to the documention so they don't have been duplicated in any machine drivers bindings. Signed-off-by: Jonathan Bakker <xc-racer2(a)live.ca> --- .../devicetree/bindings/sound/wm8994.txt | 23 +++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8994.txt b/Documentation/devicetree/bindings/sound/wm8994.txt index 367b58ce1bb9..8fa947509c10 100644 --- a/Documentation/devicetree/bindings/sound/wm8994.txt +++ b/Documentation/devicetree/bindings/sound/wm8994.txt @@ -68,6 +68,29 @@ Optional properties: - wlf,csnaddr-pd : If present enable the internal pull-down resistor on the CS/ADDR pin. +Pins on the device (for linking into audio routes): + + * IN1LN + * IN1LP + * IN2LN + * IN2LP:VXRN + * IN1RN + * IN1RP + * IN2RN + * IN2RP:VXRP + * SPKOUTLP + * SPKOUTLN + * SPKOUTRP + * SPKOUTRN + * HPOUT1L + * HPOUT1R + * HPOUT2P + * HPOUT2N + * LINEOUT1P + * LINEOUT1N + * LINEOUT2P + * LINEOUT2N + Example: wm8994: codec@1a { -- 2.20.1

2 2

[PATCH 0/4] Fix potential issues with x86 PCM SG-buffer
by Takashi Iwai 15 Jun '20

15 Jun '20

Hi, it turned out that x86 SG-pages that are used for PCM buffers aren't safe on x86 in all cases, and we need some limitation and workarounds for non-standard configurations. This patch set is an attempt to paper over it in a minimalistic way. Takashi === Takashi Iwai (4): ALSA: pcm: Use dma_mmap_coherent() on x86, too ALSA: memalloc: Initialize all fields of snd_dma_buffer properly ALSA: memalloc: Make SG-buffer helper usable for continuous buffer, too ALSA: pcm: Use SG-buffer only when direct DMA is available include/sound/memalloc.h | 9 ++++++++- sound/core/memalloc.c | 7 +++---- sound/core/pcm_memory.c | 13 +++++++++++++ sound/core/pcm_native.c | 2 -- sound/core/sgbuf.c | 3 +++ 5 files changed, 27 insertions(+), 7 deletions(-) -- 2.16.4

1 4

[PATCH] ASoC: soc-pcm: fix checks for multi-cpu FE dailinks
by Pierre-Louis Bossart 15 Jun '20

15 Jun '20

soc_dpcm_fe_runtime_update() is called for all dailinks, and we want to first discard all back-ends, then deal with front-ends. The existing code first reports an error with multi-cpu front-ends, and that check needs to be moved after we know that we are dealing with a front-end. BugLink: https://github.com/thesofproject/linux/issues/1970 Fixes: 6e1276a5e613d ('ASoC: Return error if the function does not support multi-cpu') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/soc-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c114b4542ce..c517064f5391 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2630,15 +2630,15 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) int count, paths; int ret; + if (!fe->dai_link->dynamic) + return 0; + if (fe->num_cpus > 1) { dev_err(fe->dev, "%s doesn't support Multi CPU yet\n", __func__); return -EINVAL; } - if (!fe->dai_link->dynamic) - return 0; - /* only check active links */ if (!snd_soc_dai_active(asoc_rtd_to_cpu(fe, 0))) return 0; -- 2.20.1

2 1

[PATCH 1/2] ASoC: rt1015: Update rt1015 default register value according to spec modification.
by jack.yu＠realtek.com 15 Jun '20

15 Jun '20

From: Jack Yu <jack.yu(a)realtek.com> Update rt1015 default register value according to spec modification. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 124 +++++++++++++++++++++++--------------- sound/soc/codecs/rt1015.h | 15 ++++- 2 files changed, 89 insertions(+), 50 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 67e2e944d21b..2cccb310fa96 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -34,30 +34,32 @@ static const struct reg_default rt1015_reg[] = { { 0x0000, 0x0000 }, { 0x0004, 0xa000 }, { 0x0006, 0x0003 }, - { 0x000a, 0x0802 }, - { 0x000c, 0x0020 }, + { 0x000a, 0x081e }, + { 0x000c, 0x0006 }, { 0x000e, 0x0000 }, { 0x0010, 0x0000 }, { 0x0012, 0x0000 }, + { 0x0014, 0x0000 }, + { 0x0016, 0x0000 }, + { 0x0018, 0x0000 }, { 0x0020, 0x8000 }, - { 0x0022, 0x471b }, - { 0x006a, 0x0000 }, - { 0x006c, 0x4020 }, + { 0x0022, 0x8043 }, { 0x0076, 0x0000 }, { 0x0078, 0x0000 }, - { 0x007a, 0x0000 }, + { 0x007a, 0x0002 }, { 0x007c, 0x10ec }, { 0x007d, 0x1015 }, { 0x00f0, 0x5000 }, - { 0x00f2, 0x0774 }, - { 0x00f3, 0x8400 }, + { 0x00f2, 0x004c }, + { 0x00f3, 0xecfe }, { 0x00f4, 0x0000 }, + { 0x00f6, 0x0400 }, { 0x0100, 0x0028 }, { 0x0102, 0xff02 }, - { 0x0104, 0x8232 }, + { 0x0104, 0xa213 }, { 0x0106, 0x200c }, - { 0x010c, 0x002f }, - { 0x010e, 0xc000 }, + { 0x010c, 0x0000 }, + { 0x010e, 0x0058 }, { 0x0111, 0x0200 }, { 0x0112, 0x0400 }, { 0x0114, 0x0022 }, @@ -65,38 +67,46 @@ static const struct reg_default rt1015_reg[] = { { 0x0118, 0x0000 }, { 0x011a, 0x0123 }, { 0x011c, 0x4567 }, - { 0x0300, 0xdddd }, - { 0x0302, 0x0000 }, - { 0x0311, 0x9330 }, - { 0x0313, 0x0000 }, - { 0x0314, 0x0000 }, + { 0x0300, 0x203d }, + { 0x0302, 0x001e }, + { 0x0311, 0x0000 }, + { 0x0313, 0x6014 }, + { 0x0314, 0x00a2 }, { 0x031a, 0x00a0 }, { 0x031c, 0x001f }, { 0x031d, 0xffff }, { 0x031e, 0x0000 }, { 0x031f, 0x0000 }, + { 0x0320, 0x0000 }, { 0x0321, 0x0000 }, - { 0x0322, 0x0000 }, - { 0x0328, 0x0000 }, - { 0x0329, 0x0000 }, - { 0x032a, 0x0000 }, - { 0x032b, 0x0000 }, - { 0x032c, 0x0000 }, - { 0x032d, 0x0000 }, - { 0x032e, 0x030e }, - { 0x0330, 0x0080 }, + { 0x0322, 0xd7df }, + { 0x0328, 0x10b2 }, + { 0x0329, 0x0175 }, + { 0x032a, 0x36ad }, + { 0x032b, 0x7e55 }, + { 0x032c, 0x0520 }, + { 0x032d, 0xaa00 }, + { 0x032e, 0x570e }, + { 0x0330, 0xe180 }, { 0x0332, 0x0034 }, - { 0x0334, 0x0000 }, - { 0x0336, 0x0000 }, + { 0x0334, 0x0001 }, + { 0x0336, 0x0010 }, + { 0x0338, 0x0000 }, + { 0x04fa, 0x0030 }, + { 0x04fc, 0x35c8 }, + { 0x04fe, 0x0800 }, + { 0x0500, 0x0400 }, + { 0x0502, 0x1000 }, + { 0x0504, 0x0000 }, { 0x0506, 0x04ff }, - { 0x0508, 0x0030 }, - { 0x050a, 0x0018 }, - { 0x0519, 0x307f }, - { 0x051a, 0xffff }, - { 0x051b, 0x4000 }, + { 0x0508, 0x0010 }, + { 0x050a, 0x001a }, + { 0x0519, 0x1c68 }, + { 0x051a, 0x0ccc }, + { 0x051b, 0x0666 }, { 0x051d, 0x0000 }, { 0x051f, 0x0000 }, - { 0x0536, 0x1000 }, + { 0x0536, 0x061c }, { 0x0538, 0x0000 }, { 0x053a, 0x0000 }, { 0x053c, 0x0000 }, @@ -110,19 +120,18 @@ static const struct reg_default rt1015_reg[] = { { 0x0544, 0x0000 }, { 0x0568, 0x0000 }, { 0x056a, 0x0000 }, - { 0x1000, 0x0000 }, - { 0x1002, 0x6505 }, + { 0x1000, 0x0040 }, + { 0x1002, 0x5405 }, { 0x1006, 0x5515 }, - { 0x1007, 0x003f }, - { 0x1009, 0x770f }, - { 0x100a, 0x01ff }, - { 0x100c, 0x0000 }, + { 0x1007, 0x05f7 }, + { 0x1009, 0x0b0a }, + { 0x100a, 0x00ef }, { 0x100d, 0x0003 }, { 0x1010, 0xa433 }, { 0x1020, 0x0000 }, - { 0x1200, 0x3d02 }, - { 0x1202, 0x0813 }, - { 0x1204, 0x0211 }, + { 0x1200, 0x5a01 }, + { 0x1202, 0x6524 }, + { 0x1204, 0x1f00 }, { 0x1206, 0x0000 }, { 0x1208, 0x0000 }, { 0x120a, 0x0000 }, @@ -130,16 +139,16 @@ static const struct reg_default rt1015_reg[] = { { 0x120e, 0x0000 }, { 0x1210, 0x0000 }, { 0x1212, 0x0000 }, - { 0x1300, 0x0701 }, - { 0x1302, 0x12f9 }, - { 0x1304, 0x3405 }, + { 0x1300, 0x10a1 }, + { 0x1302, 0x12ff }, + { 0x1304, 0x0400 }, { 0x1305, 0x0844 }, - { 0x1306, 0x1611 }, + { 0x1306, 0x4611 }, { 0x1308, 0x555e }, { 0x130a, 0x0000 }, - { 0x130c, 0x2400}, - { 0x130e, 0x7700 }, - { 0x130f, 0x0000 }, + { 0x130c, 0x2000 }, + { 0x130e, 0x0100 }, + { 0x130f, 0x0001 }, { 0x1310, 0x0000 }, { 0x1312, 0x0000 }, { 0x1314, 0x0000 }, @@ -209,6 +218,9 @@ static bool rt1015_volatile_register(struct device *dev, unsigned int reg) case RT1015_DC_CALIB_CLSD7: case RT1015_DC_CALIB_CLSD8: case RT1015_S_BST_TIMING_INTER1: + case RT1015_OSCK_STA: + case RT1015_MONO_DYNA_CTRL1: + case RT1015_MONO_DYNA_CTRL5: return true; default: @@ -224,6 +236,12 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg) case RT1015_CLK3: case RT1015_PLL1: case RT1015_PLL2: + case RT1015_DUM_RW1: + case RT1015_DUM_RW2: + case RT1015_DUM_RW3: + case RT1015_DUM_RW4: + case RT1015_DUM_RW5: + case RT1015_DUM_RW6: case RT1015_CLK_DET: case RT1015_SIL_DET: case RT1015_CUSTOMER_ID: @@ -235,6 +253,7 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg) case RT1015_PAD_DRV2: case RT1015_GAT_BOOST: case RT1015_PRO_ALT: + case RT1015_OSCK_STA: case RT1015_MAN_I2C: case RT1015_DAC1: case RT1015_DAC2: @@ -272,6 +291,13 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg) case RT1015_SMART_BST_CTRL2: case RT1015_ANA_CTRL1: case RT1015_ANA_CTRL2: + case RT1015_PWR_STATE_CTRL: + case RT1015_MONO_DYNA_CTRL: + case RT1015_MONO_DYNA_CTRL1: + case RT1015_MONO_DYNA_CTRL2: + case RT1015_MONO_DYNA_CTRL3: + case RT1015_MONO_DYNA_CTRL4: + case RT1015_MONO_DYNA_CTRL5: case RT1015_SPK_VOL: case RT1015_SHORT_DETTOP1: case RT1015_SHORT_DETTOP2: diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 6fbe802082c4..8169962935a5 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -21,6 +21,12 @@ #define RT1015_CLK3 0x0006 #define RT1015_PLL1 0x000a #define RT1015_PLL2 0x000c +#define RT1015_DUM_RW1 0x000e +#define RT1015_DUM_RW2 0x0010 +#define RT1015_DUM_RW3 0x0012 +#define RT1015_DUM_RW4 0x0014 +#define RT1015_DUM_RW5 0x0016 +#define RT1015_DUM_RW6 0x0018 #define RT1015_CLK_DET 0x0020 #define RT1015_SIL_DET 0x0022 #define RT1015_CUSTOMER_ID 0x0076 @@ -32,6 +38,7 @@ #define RT1015_PAD_DRV2 0x00f2 #define RT1015_GAT_BOOST 0x00f3 #define RT1015_PRO_ALT 0x00f4 +#define RT1015_OSCK_STA 0x00f6 #define RT1015_MAN_I2C 0x0100 #define RT1015_DAC1 0x0102 #define RT1015_DAC2 0x0104 @@ -70,7 +77,13 @@ #define RT1015_ANA_CTRL1 0x0334 #define RT1015_ANA_CTRL2 0x0336 #define RT1015_PWR_STATE_CTRL 0x0338 -#define RT1015_SPK_VOL 0x0506 +#define RT1015_MONO_DYNA_CTRL 0x04fa +#define RT1015_MONO_DYNA_CTRL1 0x04fc +#define RT1015_MONO_DYNA_CTRL2 0x04fe +#define RT1015_MONO_DYNA_CTRL3 0x0500 +#define RT1015_MONO_DYNA_CTRL4 0x0502 +#define RT1015_MONO_DYNA_CTRL5 0x0504 +#define RT1015_SPK_VOL 0x0506 #define RT1015_SHORT_DETTOP1 0x0508 #define RT1015_SHORT_DETTOP2 0x050a #define RT1015_SPK_DC_DETECT1 0x0519 -- 2.25.1

2 4

[PATCH v3] ASoC: SOF: Intel: hda: Clear RIRB status before reading WP
by Brent Lu 15 Jun '20

15 Jun '20

Port commit 6d011d5057ff ("ALSA: hda: Clear RIRB status before reading WP") from legacy HDA driver to fix the get response timeout issue. Current SOF driver does not suffer from this issue because sync write is enabled in hda_init. The issue will come back if the sync write is disabled for some reason. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/sof/intel/hda-stream.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 7f65dcc..1bda14c 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -653,11 +653,16 @@ irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) if (status & AZX_INT_CTRL_EN) { rirb_status = snd_hdac_chip_readb(bus, RIRBSTS); if (rirb_status & RIRB_INT_MASK) { + /* + * Clearing the interrupt status here ensures + * that no interrupt gets masked after the RIRB + * wp is read in snd_hdac_bus_update_rirb. + */ + snd_hdac_chip_writeb(bus, RIRBSTS, + RIRB_INT_MASK); active = true; if (rirb_status & RIRB_INT_RESPONSE) snd_hdac_bus_update_rirb(bus); - snd_hdac_chip_writeb(bus, RIRBSTS, - RIRB_INT_MASK); } } #endif -- 2.7.4

5 4

[PATCH 1/2] ASoc: q6afe: add support to get port direction
by Srinivas Kandagatla 15 Jun '20

15 Jun '20

This patch adds support to q6afe_is_rx_port() to get direction of DSP BE dai port, this is useful for setting dailink directions correctly. Fixes: c25e295cd77b (ASoC: qcom: Add support to parse common audio device nodes") Reported-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe.c | 8 ++++++++ sound/soc/qcom/qdsp6/q6afe.h | 1 + 2 files changed, 9 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index e0945f7a58c8..0ce4eb60f984 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -800,6 +800,14 @@ int q6afe_get_port_id(int index) } EXPORT_SYMBOL_GPL(q6afe_get_port_id); +int q6afe_is_rx_port(int index) +{ + if (index < 0 || index >= AFE_PORT_MAX) + return -EINVAL; + + return port_maps[index].is_rx; +} +EXPORT_SYMBOL_GPL(q6afe_is_rx_port); static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt, struct q6afe_port *port) { diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index c7ed5422baff..1a0f80a14afe 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -198,6 +198,7 @@ int q6afe_port_start(struct q6afe_port *port); int q6afe_port_stop(struct q6afe_port *port); void q6afe_port_put(struct q6afe_port *port); int q6afe_get_port_id(int index); +int q6afe_is_rx_port(int index); void q6afe_hdmi_port_prepare(struct q6afe_port *port, struct q6afe_hdmi_cfg *cfg); void q6afe_slim_port_prepare(struct q6afe_port *port, -- 2.21.0

4 5

[PATCH 0/2] ASoC: topology: fix use-after-free when removing components
by Pierre-Louis Bossart 15 Jun '20

15 Jun '20

This patchset fixes a memory allocation issue and removes a 100% reproducible use-after-free report thrown by KASAN in automated module removal tests across multiple platforms. All the credit goes to Bard Liao for root-causing the issue. DAIs may be registered at the same time as a component, or when the topology is loaded. This two-step registration causes the memory for topology-based DAIs to allocated last, and conversely to be released first by devres, before the component is released and the DAIs removed from the component DAI list with snd_soc_unregister_dais(). When we remove a component, by the time we walk through its dai list to unregister all dais, the dais allocated by the topology have been freed already by devres and the list is corrupted with pointers that are no longer valid. The suggestion is to add an explicit devm_ based registration for topology-based dais, so that each dai is cleanly removed from the component dai list in the release operation before devres releases the allocated memory. Pierre-Louis Bossart (2): ASoC: soc-devres: add devm_snd_soc_register_dai() ASoC: soc-topology: use devm_snd_soc_register_dai() include/sound/soc.h | 4 ++++ sound/soc/soc-devres.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/soc-topology.c | 3 +-- 3 files changed, 42 insertions(+), 2 deletions(-) -- 2.20.1

2 3

alsa-gobject: seq/timer: add constructure for status object
by GitHub pull_request - opened 15 Jun '20

15 Jun '20

alsa-project/alsa-gobject pull request #39 was opened from takaswie: In patchset at #38, I forgot to add constructors for issued status object because no function call returns the instance of object. This patchset adds the constructors. Request URL : https://github.com/alsa-project/alsa-gobject/pull/39 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ALSA: usb-audio: Set 48 kHz rate for Rodecaster
by Christopher Swenson 15 Jun '20

15 Jun '20

>From 22581d37726a17ed9c400b8c9ff8ade52c17f65a Mon Sep 17 00:00:00 2001 From: Christopher Swenson <swenson(a)swenson.io> Date: Sun, 14 Jun 2020 15:19:41 -0700 Subject: [PATCH] ALSA: usb-audio: Set 48 kHz rate for Rodecaster Like the Line6 devices, the Rode Rodecaster Pro does not support UAC2_CS_RANGE and only supports a sample rate of 48 kHz. Tested against a Rode Rodecaster Pro. Tested-by: Christopher Swenson <swenson(a)swenson.io> Signed-off-by: Christopher Swenson <swenson(a)swenson.io> --- sound/usb/format.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 5ffb457cc88c..1b28d01d1f4c 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -394,8 +394,9 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, return nr_rates; } -/* Line6 Helix series don't support the UAC2_CS_RANGE usb function - * call. Return a static table of known clock rates. +/* Line6 Helix series and the Rode Rodecaster Pro don't support the + * UAC2_CS_RANGE usb function call. Return a static table of known + * clock rates. */ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, struct audioformat *fp) @@ -408,6 +409,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */ case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ case USB_ID(0x0e41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ + case USB_ID(0x19f7, 0x0011): /* Rode Rodecaster Pro */ return set_fixed_rate(fp, 48000, SNDRV_PCM_RATE_48000); } -- 2.26.0

2 1

[patch for-5.8] dma-pool: decouple DMA_REMAP from DMA_COHERENT_POOL
by David Rientjes 15 Jun '20

15 Jun '20

DMA_REMAP is an unnecessary requirement for AMD SEV, which requires DMA_COHERENT_POOL, so avoid selecting it when it is otherwise unnecessary. The only other requirement for DMA coherent pools is DMA_DIRECT_REMAP, so ensure that properly selects the config option when needed. Fixes: 82fef0ad811f ("x86/mm: unencrypted non-blocking DMA allocations use coherent pools") Suggested-by: Christoph Hellwig <hch(a)lst.de> Signed-off-by: David Rientjes <rientjes(a)google.com> --- kernel/dma/Kconfig | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/kernel/dma/Kconfig b/kernel/dma/Kconfig --- a/kernel/dma/Kconfig +++ b/kernel/dma/Kconfig @@ -73,18 +73,18 @@ config SWIOTLB config DMA_NONCOHERENT_MMAP bool +config DMA_COHERENT_POOL + bool + config DMA_REMAP + bool depends on MMU select GENERIC_ALLOCATOR select DMA_NONCOHERENT_MMAP - bool - -config DMA_COHERENT_POOL - bool - select DMA_REMAP config DMA_DIRECT_REMAP bool + select DMA_REMAP select DMA_COHERENT_POOL config DMA_CMA

3 2

alsa-gobject: seq/timer: use (inout) annotation for API to retrieve status
by GitHub pull_request - opened 15 Jun '20

15 Jun '20

alsa-project/alsa-gobject pull request #38 was opened from takaswie: It's probable to retrieve status for something several times in runtime, and it's convenient for the runtime to use pre-allocated instance for the status. Request URL : https://github.com/alsa-project/alsa-gobject/pull/38 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/38.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

sof-hda-dsp: add initial kcontrol values
by GitHub pull_request - opened 15 Jun '20

15 Jun '20

alsa-project/alsa-ucm-conf pull request #35 was opened from libinyang: This patch adds the initial values of sof-hda-dsp card. Signed-off-by: Libin Yang <libin.yang(a)intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/35 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/35.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 3/3] ASoC: samsung: Add driver for Aries boards
by Jonathan Bakker 14 Jun '20

14 Jun '20

Samsung Aries boards have a WM8994 codec connected to the Samsung I2S controller, the BT codec, and the cellular modem. Jack detection is done by a combination of an ADC, GPIOs, and an extcon device for the USB dock. There is also a GPIO for selection between the Mic path and the TV out path on the headphone jack. There are two main variants, one with an FM radio and where the modem is the master and one without a radio and the modem is the slave. Signed-off-by: Jonathan Bakker <xc-racer2(a)live.ca> --- sound/soc/samsung/Kconfig | 13 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/aries_wm8994.c | 695 +++++++++++++++++++++++++++++++ 3 files changed, 710 insertions(+) create mode 100644 sound/soc/samsung/aries_wm8994.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 112911dc271b..505087d2a50d 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -212,4 +212,17 @@ config SND_SOC_SAMSUNG_TM2_WM5110 help Say Y if you want to add support for SoC audio on the TM2 board. +config SND_SOC_SAMSUNG_ARIES_WM8994 + tristate "SoC I2S Audio support for WM8994 on Aries" + depends on SND_SOC_SAMSUNG && MFD_WM8994 && IIO && EXTCON + select SND_SOC_BT_SCO + select SND_SOC_WM8994 + select SND_SAMSUNG_I2S + help + Say Y if you want to add support for SoC audio on Aries boards, + which has a WM8994 codec connected to a BT codec, a cellular + modem, and the Samsung I2S controller. Jack detection is done + via ADC, GPIOs, and an extcon device. Switching between the Mic + and TV-Out path is also handled. + endif #SND_SOC_SAMSUNG diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 8f5dfe20b9f1..22259f7818f0 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -41,6 +41,7 @@ snd-soc-bells-objs := bells.o snd-soc-odroid-objs := odroid.o snd-soc-arndale-objs := arndale.o snd-soc-tm2-wm5110-objs := tm2_wm5110.o +snd-soc-aries-wm8994-objs := aries_wm8994.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -64,3 +65,4 @@ obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o obj-$(CONFIG_SND_SOC_ODROID) += snd-soc-odroid.o obj-$(CONFIG_SND_SOC_ARNDALE) += snd-soc-arndale.o obj-$(CONFIG_SND_SOC_SAMSUNG_TM2_WM5110) += snd-soc-tm2-wm5110.o +obj-$(CONFIG_SND_SOC_SAMSUNG_ARIES_WM8994) += snd-soc-aries-wm8994.o diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c new file mode 100644 index 000000000000..8579c87dcae8 --- /dev/null +++ b/sound/soc/samsung/aries_wm8994.c @@ -0,0 +1,695 @@ +// SPDX-License-Identifier: GPL-2.0+ +#include <linux/extcon.h> +#include <linux/iio/consumer.h> +#include <linux/iio/iio.h> +#include <linux/input-event-codes.h> +#include <linux/mfd/wm8994/registers.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "i2s.h" +#include "../codecs/wm8994.h" + +#define ARIES_MCLK1_FREQ 24000000 + +struct aries_wm8994_variant { + unsigned int modem_dai_fmt; + bool has_fm_radio; +}; + +struct aries_wm8994_data { + struct extcon_dev *usb_extcon; + struct regulator *reg_main_micbias; + struct regulator *reg_headset_micbias; + struct gpio_desc *gpio_headset_detect; + struct gpio_desc *gpio_headset_key; + struct gpio_desc *gpio_earpath_sel; + struct iio_channel *adc; + const struct aries_wm8994_variant *variant; +}; + +/* USB dock */ +static struct snd_soc_jack aries_dock; + +static struct snd_soc_jack_pin dock_pins[] = { + { + .pin = "LINE", + .mask = SND_JACK_LINEOUT, + }, +}; + +static int aries_extcon_notifier(struct notifier_block *this, + unsigned long connected, void *_cmd) +{ + if (connected) + snd_soc_jack_report(&aries_dock, SND_JACK_LINEOUT, + SND_JACK_LINEOUT); + else + snd_soc_jack_report(&aries_dock, 0, SND_JACK_LINEOUT); + + return NOTIFY_DONE; +} + +static struct notifier_block aries_extcon_notifier_block = { + .notifier_call = aries_extcon_notifier, +}; + +/* Headset jack */ +static struct snd_soc_jack aries_headset; + +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "HP", + .mask = SND_JACK_HEADPHONE, + }, { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static struct snd_soc_jack_zone headset_zones[] = { + { + .min_mv = 0, + .max_mv = 241, + .jack_type = SND_JACK_HEADPHONE, + }, { + .min_mv = 242, + .max_mv = 2980, + .jack_type = SND_JACK_HEADSET, + }, { + .min_mv = 2981, + .max_mv = UINT_MAX, + .jack_type = SND_JACK_HEADPHONE, + }, +}; + +static irqreturn_t headset_det_irq_thread(int irq, void *data) +{ + struct aries_wm8994_data *priv = (struct aries_wm8994_data *) data; + int ret = 0; + int time_left_ms = 300; + int adc; + + while (time_left_ms > 0) { + if (!gpiod_get_value(priv->gpio_headset_detect)) { + snd_soc_jack_report(&aries_headset, 0, + SND_JACK_HEADSET); + gpiod_set_value(priv->gpio_earpath_sel, 0); + return IRQ_HANDLED; + } + msleep(20); + time_left_ms -= 20; + } + + /* Temporarily enable micbias and earpath selector */ + ret = regulator_enable(priv->reg_headset_micbias); + if (ret) + pr_err("%s failed to enable micbias: %d", __func__, ret); + + gpiod_set_value(priv->gpio_earpath_sel, 1); + + ret = iio_read_channel_processed(priv->adc, &adc); + if (ret < 0) { + /* failed to read ADC, so assume headphone */ + pr_err("%s failed to read ADC, assuming headphones", __func__); + snd_soc_jack_report(&aries_headset, SND_JACK_HEADPHONE, + SND_JACK_HEADSET); + } else { + snd_soc_jack_report(&aries_headset, + snd_soc_jack_get_type(&aries_headset, adc), + SND_JACK_HEADSET); + } + + ret = regulator_disable(priv->reg_headset_micbias); + if (ret) + pr_err("%s failed disable micbias: %d", __func__, ret); + + /* Disable earpath selector when no mic connected */ + if (!(aries_headset.status & SND_JACK_MICROPHONE)) + gpiod_set_value(priv->gpio_earpath_sel, 0); + + return IRQ_HANDLED; +} + +static int headset_button_check(void *data) +{ + struct aries_wm8994_data *priv = (struct aries_wm8994_data *) data; + + /* Filter out keypresses when 4 pole jack not detected */ + if (gpiod_get_value_cansleep(priv->gpio_headset_key) && + aries_headset.status & SND_JACK_MICROPHONE) + return SND_JACK_BTN_0; + + return 0; +} + +static struct snd_soc_jack_gpio headset_button_gpio[] = { + { + .name = "Media Button", + .report = SND_JACK_BTN_0, + .debounce_time = 30, + .jack_status_check = headset_button_check, + }, +}; + +static int aries_spk_cfg(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_component *component; + int ret = 0; + + rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); + component = asoc_rtd_to_codec(rtd, 0)->component; + + /** + * We have an odd setup - the SPKMODE pin is pulled up so + * we only have access to the left side SPK configs, + * but SPKOUTR isn't bridged so when playing back in + * stereo, we only get the left hand channel. The only + * option we're left with is to force the AIF into mono + * mode. + */ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + ret = snd_soc_component_update_bits(component, + WM8994_AIF1_DAC1_FILTERS_1, + WM8994_AIF1DAC1_MONO, WM8994_AIF1DAC1_MONO); + break; + case SND_SOC_DAPM_PRE_PMD: + ret = snd_soc_component_update_bits(component, + WM8994_AIF1_DAC1_FILTERS_1, + WM8994_AIF1DAC1_MONO, 0); + break; + } + + return ret; +} + +static int aries_main_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct aries_wm8994_data *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(priv->reg_main_micbias); + break; + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable(priv->reg_main_micbias); + break; + } + + return ret; +} + +static int aries_headset_bias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct aries_wm8994_data *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(priv->reg_headset_micbias); + break; + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable(priv->reg_headset_micbias); + break; + } + + return ret; +} + +static const struct snd_kcontrol_new aries_controls[] = { + SOC_DAPM_PIN_SWITCH("Modem In"), + SOC_DAPM_PIN_SWITCH("Modem Out"), +}; + +static const struct snd_soc_dapm_widget aries_dapm_widgets[] = { + SND_SOC_DAPM_HP("HP", NULL), + + SND_SOC_DAPM_SPK("SPK", aries_spk_cfg), + SND_SOC_DAPM_SPK("RCV", NULL), + + SND_SOC_DAPM_LINE("LINE", NULL), + + SND_SOC_DAPM_MIC("Main Mic", aries_main_bias), + SND_SOC_DAPM_MIC("Headset Mic", aries_headset_bias), + + SND_SOC_DAPM_MIC("Bluetooth Mic", NULL), + SND_SOC_DAPM_SPK("Bluetooth SPK", NULL), + + SND_SOC_DAPM_LINE("Modem In", NULL), + SND_SOC_DAPM_LINE("Modem Out", NULL), + + /* This must be last as it is conditionally not used */ + SND_SOC_DAPM_LINE("FM In", NULL), +}; + +static int aries_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + unsigned int pll_out; + int ret; + + /* AIF1CLK should be >=3MHz for optimal performance */ + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else if (params_rate(params) == 8000 || params_rate(params) == 11025) + pll_out = params_rate(params) * 512; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + ARIES_MCLK1_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static int aries_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + int ret; + + /* Switch sysclk to MCLK1 */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, + ARIES_MCLK1_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Stop PLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + ARIES_MCLK1_FREQ, 0); + if (ret < 0) + return ret; + + return 0; +} + +/* + * Main DAI operations + */ +static struct snd_soc_ops aries_ops = { + .hw_params = aries_hw_params, + .hw_free = aries_hw_free, +}; + +static int aries_baseband_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + unsigned int pll_out; + int ret; + + pll_out = 8000 * 512; + + /* Set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, WM8994_FLL_SRC_MCLK1, + ARIES_MCLK1_FREQ, pll_out); + if (ret < 0) + return ret; + + /* Set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static int aries_late_probe(struct snd_soc_card *card) +{ + struct aries_wm8994_data *priv = snd_soc_card_get_drvdata(card); + int ret, irq; + + ret = snd_soc_card_jack_new(card, "Dock", SND_JACK_LINEOUT, + &aries_dock, dock_pins, ARRAY_SIZE(dock_pins)); + if (ret) + return ret; + + ret = devm_extcon_register_notifier(card->dev, + priv->usb_extcon, EXTCON_JACK_LINE_OUT, + &aries_extcon_notifier_block); + if (ret) + return ret; + + if (extcon_get_state(priv->usb_extcon, + EXTCON_JACK_LINE_OUT) > 0) + snd_soc_jack_report(&aries_dock, SND_JACK_LINEOUT, + SND_JACK_LINEOUT); + else + snd_soc_jack_report(&aries_dock, 0, SND_JACK_LINEOUT); + + ret = snd_soc_card_jack_new(card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &aries_headset, + jack_pins, ARRAY_SIZE(jack_pins)); + if (ret) + return ret; + + ret = snd_soc_jack_add_zones(&aries_headset, ARRAY_SIZE(headset_zones), + headset_zones); + if (ret) + return ret; + + irq = gpiod_to_irq(priv->gpio_headset_detect); + if (irq < 0) { + dev_err(card->dev, "Failed to map headset detect gpio to irq"); + return -EINVAL; + } + + ret = devm_request_threaded_irq(card->dev, irq, NULL, + headset_det_irq_thread, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | + IRQF_ONESHOT, "headset_detect", priv); + if (ret) { + dev_err(card->dev, "Failed to request headset detect irq"); + return ret; + } + + headset_button_gpio[0].data = priv; + headset_button_gpio[0].desc = priv->gpio_headset_key; + + snd_jack_set_key(aries_headset.jack, SND_JACK_BTN_0, KEY_MEDIA); + + return snd_soc_jack_add_gpios(&aries_headset, + ARRAY_SIZE(headset_button_gpio), headset_button_gpio); +} + +static const struct snd_soc_pcm_stream baseband_params = { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 1, + .channels_max = 1, +}; + +static const struct snd_soc_pcm_stream bluetooth_params = { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 1, + .channels_max = 2, +}; + +static const struct snd_soc_dapm_widget aries_modem_widgets[] = { + SND_SOC_DAPM_INPUT("Modem RX"), + SND_SOC_DAPM_OUTPUT("Modem TX"), +}; + +static const struct snd_soc_dapm_route aries_modem_routes[] = { + { "Modem Capture", NULL, "Modem RX" }, + { "Modem TX", NULL, "Modem Playback" }, +}; + +static const struct snd_soc_component_driver aries_component = { + .name = "aries-audio", + .dapm_widgets = aries_modem_widgets, + .num_dapm_widgets = ARRAY_SIZE(aries_modem_widgets), + .dapm_routes = aries_modem_routes, + .num_dapm_routes = ARRAY_SIZE(aries_modem_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static struct snd_soc_dai_driver aries_ext_dai[] = { + { + .name = "Voice call", + .playback = { + .stream_name = "Modem Playback", + .channels_min = 1, + .channels_max = 1, + .rate_min = 8000, + .rate_max = 8000, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Modem Capture", + .channels_min = 1, + .channels_max = 1, + .rate_min = 8000, + .rate_max = 8000, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, +}; + +SND_SOC_DAILINK_DEFS(aif1, + DAILINK_COMP_ARRAY(COMP_CPU(SAMSUNG_I2S_DAI)), + DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "wm8994-aif1")), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +SND_SOC_DAILINK_DEFS(baseband, + DAILINK_COMP_ARRAY(COMP_CPU("Voice call")), + DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "wm8994-aif2"))); + +SND_SOC_DAILINK_DEFS(bluetooth, + DAILINK_COMP_ARRAY(COMP_CPU("bt-sco-pcm")), + DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "wm8994-aif3"))); + +static struct snd_soc_dai_link aries_dai[] = { + { + .name = "WM8994 AIF1", + .stream_name = "HiFi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .ops = &aries_ops, + SND_SOC_DAILINK_REG(aif1), + }, + { + .name = "WM8994 AIF2", + .stream_name = "Baseband", + .init = &aries_baseband_init, + .params = &baseband_params, + .ignore_suspend = 1, + SND_SOC_DAILINK_REG(baseband), + }, + { + .name = "WM8994 AIF3", + .stream_name = "Bluetooth", + .params = &bluetooth_params, + .ignore_suspend = 1, + SND_SOC_DAILINK_REG(bluetooth), + }, +}; + +static struct snd_soc_card aries_card = { + .name = "ARIES", + .owner = THIS_MODULE, + .dai_link = aries_dai, + .num_links = ARRAY_SIZE(aries_dai), + .controls = aries_controls, + .num_controls = ARRAY_SIZE(aries_controls), + .dapm_widgets = aries_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aries_dapm_widgets), + .late_probe = aries_late_probe, +}; + +static const struct aries_wm8994_variant fascinate4g_variant = { + .modem_dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBS_CFS + | SND_SOC_DAIFMT_IB_NF, + .has_fm_radio = false, +}; + +static const struct aries_wm8994_variant aries_variant = { + .modem_dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_CBM_CFM + | SND_SOC_DAIFMT_IB_NF, + .has_fm_radio = true, +}; + +static const struct of_device_id samsung_wm8994_of_match[] = { + { + .compatible = "samsung,fascinate4g-wm8994", + .data = &fascinate4g_variant, + }, + { + .compatible = "samsung,aries-wm8994", + .data = &aries_variant, + }, + { /* sentinel */ }, +}; +MODULE_DEVICE_TABLE(of, samsung_wm8994_of_match); + +static int aries_audio_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *cpu, *codec, *extcon_np; + struct device *dev = &pdev->dev; + struct snd_soc_card *card = &aries_card; + struct aries_wm8994_data *priv; + struct snd_soc_dai_link *dai_link; + const struct of_device_id *match; + int ret, i; + + if (!np) + return -EINVAL; + + card->dev = dev; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + snd_soc_card_set_drvdata(card, priv); + + match = of_match_node(samsung_wm8994_of_match, np); + priv->variant = match->data; + + /* Remove FM widget if not present */ + if (!priv->variant->has_fm_radio) + card->num_dapm_widgets--; + + priv->reg_main_micbias = devm_regulator_get(dev, "main-micbias"); + if (IS_ERR(priv->reg_main_micbias)) { + dev_err(dev, "Failed to get main micbias regulator\n"); + return PTR_ERR(priv->reg_main_micbias); + } + + priv->reg_headset_micbias = devm_regulator_get(dev, "headset-micbias"); + if (IS_ERR(priv->reg_headset_micbias)) { + dev_err(dev, "Failed to get headset micbias regulator\n"); + return PTR_ERR(priv->reg_headset_micbias); + } + + priv->gpio_earpath_sel = devm_gpiod_get(dev, "earpath-sel", + GPIOD_OUT_LOW); + if (IS_ERR(priv->gpio_earpath_sel)) { + dev_err(dev, "Failed to get earpath selector gpio"); + return PTR_ERR(priv->gpio_earpath_sel); + } + + extcon_np = of_parse_phandle(np, "extcon", 0); + priv->usb_extcon = extcon_find_edev_by_node(extcon_np); + if (IS_ERR(priv->usb_extcon)) { + if (PTR_ERR(priv->usb_extcon) != -EPROBE_DEFER) + dev_err(dev, "Failed to get extcon device"); + return PTR_ERR(priv->usb_extcon); + } + of_node_put(extcon_np); + + priv->adc = devm_iio_channel_get(dev, "headset-detect"); + if (IS_ERR(priv->adc)) { + if (PTR_ERR(priv->adc) != -EPROBE_DEFER) + dev_err(dev, "Failed to get ADC channel"); + return PTR_ERR(priv->adc); + } + if (priv->adc->channel->type != IIO_VOLTAGE) + return -EINVAL; + + priv->gpio_headset_key = devm_gpiod_get(dev, "headset-key", + GPIOD_IN); + if (IS_ERR(priv->gpio_headset_key)) { + dev_err(dev, "Failed to get headset key gpio"); + return PTR_ERR(priv->gpio_headset_key); + } + + priv->gpio_headset_detect = devm_gpiod_get(dev, + "headset-detect", GPIOD_IN); + if (IS_ERR(priv->gpio_headset_detect)) { + dev_err(dev, "Failed to get headset detect gpio"); + return PTR_ERR(priv->gpio_headset_detect); + } + + /* Update card-name if provided through DT, else use default name */ + snd_soc_of_parse_card_name(card, "model"); + + ret = snd_soc_of_parse_audio_routing(card, "samsung,audio-routing"); + if (ret < 0) { + dev_err(dev, "Audio routing invalid/unspecified\n"); + return ret; + } + + aries_dai[1].dai_fmt = priv->variant->modem_dai_fmt; + + cpu = of_get_child_by_name(dev->of_node, "cpu"); + if (!cpu) + return -EINVAL; + + codec = of_get_child_by_name(dev->of_node, "codec"); + if (!codec) + return -EINVAL; + + for_each_card_prelinks(card, i, dai_link) { + dai_link->codecs->of_node = of_parse_phandle(codec, + "sound-dai", 0); + if (!dai_link->codecs->of_node) { + ret = -EINVAL; + goto out; + } + } + + /* Set CPU and platform of_node for main DAI */ + aries_dai[0].cpus->of_node = of_parse_phandle(cpu, + "sound-dai", 0); + if (!aries_dai[0].cpus->of_node) { + ret = -EINVAL; + goto out; + } + + aries_dai[0].platforms->of_node = aries_dai[0].cpus->of_node; + + /* Set CPU of_node for BT DAI */ + aries_dai[2].cpus->of_node = of_parse_phandle(cpu, + "sound-dai", 1); + if (!aries_dai[2].cpus->of_node) { + ret = -EINVAL; + goto out; + } + + ret = devm_snd_soc_register_component(dev, &aries_component, + aries_ext_dai, ARRAY_SIZE(aries_ext_dai)); + if (ret < 0) { + dev_err(dev, "Failed to register component: %d\n", ret); + goto out; + } + + ret = devm_snd_soc_register_card(dev, card); + if (ret) + dev_err(dev, "snd_soc_register_card() failed:%d\n", ret); + +out: + of_node_put(cpu); + of_node_put(codec); + + return ret; +} + +static struct platform_driver aries_audio_driver = { + .driver = { + .name = "aries-audio-wm8994", + .of_match_table = of_match_ptr(samsung_wm8994_of_match), + .pm = &snd_soc_pm_ops, + }, + .probe = aries_audio_probe, +}; + +module_platform_driver(aries_audio_driver); + +MODULE_DESCRIPTION("ALSA SoC ARIES WM8994"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:aries-audio-wm8994"); -- 2.20.1

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[PATCH 2/3] dt-bindings: sound: Add bindings for Samsung Aries audio complex
by Jonathan Bakker 14 Jun '20

14 Jun '20

The audio system on S5PV210 Aries boards have a wm8994 codec connected to the Samsung I2S0 DAI. Jack detection is done via GPIOs, an ADC, and an extcon device (fsa9480). There are two main variants, one with an FM radio and where the wm8994 is the master for the modem audio and the other without FM and the modem is the master. Signed-off-by: Jonathan Bakker <xc-racer2(a)live.ca> --- .../bindings/sound/samsung,aries-wm8994.yaml | 147 ++++++++++++++++++ 1 file changed, 147 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml new file mode 100644 index 000000000000..902a0b66628e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -0,0 +1,147 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,aries-wm8994.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Aries audio complex with WM8994 codec + +maintainers: + - Jonathan Bakker <xc-racer2(a)live.ca> + +properties: + compatible: + oneOf: + - const: samsung,aries-wm8994 + description: With FM radio and modem master + + - const: samsung,fascinate4g-wm8994 + description: Without FM radio and modem slave + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + minItems: 2 + maxItems: 2 + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + phandles to the I2S controller and bluetooth codec, + in that order + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle to the WM8994 CODEC + + samsung,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 for sources and sinks are the WM8994's pins (as + documented in its binding), and the jacks on the board - + For samsung,aries-wm8994: HP, SPK, RCV, LINE, Main Mic, Headset Mic, + or FM In + For samsung,fascinate4g-wm8994: HP, SPK, RCV, LINE, Main Mic, + or HeadsetMic + + extcon: + description: Extcon phandle for dock detection + + main-micbias-supply: + description: Supply for the micbias on the main mic + + headset-micbias-supply: + description: Supply for the micbias on the headset mic + + earpath-sel-gpios: + description: GPIO for switching between tv-out and mic paths + + headset-detect-gpios: + description: GPIO for detection of headset insertion + + headset-key-gpios: + description: GPIO for detection of headset key press + + io-channels: + maxItems: 1 + description: IO channel to read micbias voltage for headset detection + + io-channel-names: + const: headset-detect + +required: + - compatible + - model + - cpu + - codec + - samsung,audio-routing + - extcon + - main-micbias-supply + - headset-micbias-supply + - earpath-sel-gpios + - headset-detect-gpios + - headset-key-gpios + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + sound { + compatible = "samsung,fascinate4g-wm8994"; + + model = "Fascinate4G"; + + extcon = <&fsa9480>; + + main-micbias-supply = <&main_micbias_reg>; + headset-micbias-supply = <&headset_micbias_reg>; + + earpath-sel-gpios = <&gpj2 6 GPIO_ACTIVE_HIGH>; + + io-channels = <&adc 3>; + io-channel-names = "headset-detect"; + headset-detect-gpios = <&gph0 6 GPIO_ACTIVE_HIGH>; + headset-key-gpios = <&gph3 6 GPIO_ACTIVE_HIGH>; + + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + + "SPK", "SPKOUTLN", + "SPK", "SPKOUTLP", + + "RCV", "HPOUT2N", + "RCV", "HPOUT2P", + + "LINE", "LINEOUT2N", + "LINE", "LINEOUT2P", + + "IN1LP", "Main Mic", + "IN1LN", "Main Mic", + + "IN1RP", "Headset Mic", + "IN1RN", "Headset Mic"; + + pinctrl-names = "default"; + pinctrl-0 = <&headset_det &earpath_sel>; + + cpu { + sound-dai = <&i2s0>, <&bt_codec>; + }; + + codec { + sound-dai = <&wm8994>; + }; + }; + -- 2.20.1

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[PATCH 0/3] ASoC: samsung: Add machine driver for Aries boards
by Jonathan Bakker 14 Jun '20

14 Jun '20

This is a machine driver for Aries boards which are based on Samsung's S5PV210 platform. These are the first-gen Galaxy S devices. It consists of a WM8994 codec connected to I2S0 along with two codec-to-codec links to the modem and the bluetooth. --------- | | Modem | | --------- | AIF2 | ---------- --------- | | AIF1 | | I2S ------- wm8994 | | | | ---------- --------- | AIF3 | --------- | | BT | | --------- Jack detection is relatively complex with various GPIOs, an ADC, and an extcon device all involved. One oddity is that the headphone jack is shared between the mic and the analog tv-out and is controlled by a GPIO (earpath-sel) and the proper path should be selected based on the jack detection. There is also an oddity with how the speaker is connected. It's a mono speaker and the spkmode pin is pulled up, but the connections aren't bridged so if stereo playback is attempted, only the one channel will play. Therefore, when the speaker is enabled the entire AIF should be forced into mono mode. UCM2 config files are available at https://github.com/xc-racer99/aries-hw-files/tree/ucm2/usr/share/alsa/ucm2 Jonathan Bakker (3): dt-bindings: sound: Document wm8994 endpoints dt-bindings: sound: Add bindings for Samsung Aries audio complex ASoC: samsung: Add driver for Aries boards .../bindings/sound/samsung,aries-wm8994.yaml | 147 ++++ .../devicetree/bindings/sound/wm8994.txt | 23 + sound/soc/samsung/Kconfig | 13 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/aries_wm8994.c | 695 ++++++++++++++++++ 5 files changed, 880 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml create mode 100644 sound/soc/samsung/aries_wm8994.c -- 2.20.1

1 0

alsa usb problem
by GitHub issues - opened 14 Jun '20

14 Jun '20

alsa-project/alsa-lib issue #59 was opened from texstar: Plugging in a usb microphone causes pulse audio to crash with alsa-lib 1.2.3. Issue URL : https://github.com/alsa-project/alsa-lib/issues/59 Repository URL: https://github.com/alsa-project/alsa-lib

1 0