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

[PATCH] dt-bindings: sound: sgtl5000: Convert to json-schema
by Thierry Reding 22 Jul '20

22 Jul '20

From: Thierry Reding <treding(a)nvidia.com> Convert the Freescale SGTL5000 device tree bindings from free-form text format to json-schema. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- .../devicetree/bindings/sound/sgtl5000.txt | 60 ---------- .../devicetree/bindings/sound/sgtl5000.yaml | 103 ++++++++++++++++++ .../devicetree/bindings/trivial-devices.yaml | 2 - 3 files changed, 103 insertions(+), 62 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/sgtl5000.txt create mode 100644 Documentation/devicetree/bindings/sound/sgtl5000.yaml diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt deleted file mode 100644 index 9d9ff5184939..000000000000 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ /dev/null @@ -1,60 +0,0 @@ -* Freescale SGTL5000 Stereo Codec - -Required properties: -- compatible : "fsl,sgtl5000". - -- reg : the I2C address of the device - -- #sound-dai-cells: must be equal to 0 - -- clocks : the clock provider of SYS_MCLK - -- VDDA-supply : the regulator provider of VDDA - -- VDDIO-supply: the regulator provider of VDDIO - -Optional properties: - -- VDDD-supply : the regulator provider of VDDD - -- micbias-resistor-k-ohms : the bias resistor to be used in kOhms - The resistor can take values of 2k, 4k or 8k. - If set to 0 it will be off. - If this node is not mentioned or if the value is unknown, then - micbias resistor is set to 4K. - -- micbias-voltage-m-volts : the bias voltage to be used in mVolts - The voltage can take values from 1.25V to 3V by 250mV steps - If this node is not mentioned or the value is unknown, then - the value is set to 1.25V. - -- lrclk-strength: the LRCLK pad strength. Possible values are: -0, 1, 2 and 3 as per the table below: - -VDDIO 1.8V 2.5V 3.3V -0 = Disable -1 = 1.66 mA 2.87 mA 4.02 mA -2 = 3.33 mA 5.74 mA 8.03 mA -3 = 4.99 mA 8.61 mA 12.05 mA - -- sclk-strength: the SCLK pad strength. Possible values are: -0, 1, 2 and 3 as per the table below: - -VDDIO 1.8V 2.5V 3.3V -0 = Disable -1 = 1.66 mA 2.87 mA 4.02 mA -2 = 3.33 mA 5.74 mA 8.03 mA -3 = 4.99 mA 8.61 mA 12.05 mA - -Example: - -sgtl5000: codec@a { - compatible = "fsl,sgtl5000"; - reg = <0x0a>; - #sound-dai-cells = <0>; - clocks = <&clks 150>; - micbias-resistor-k-ohms = <2>; - micbias-voltage-m-volts = <2250>; - VDDA-supply = <&reg_3p3v>; - VDDIO-supply = <&reg_3p3v>; -}; diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.yaml b/Documentation/devicetree/bindings/sound/sgtl5000.yaml new file mode 100644 index 000000000000..4f29b63c54d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sgtl5000.yaml @@ -0,0 +1,103 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sgtl5000.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale SGTL5000 Stereo Codec + +maintainers: + - Fabio Estevam <festevam(a)gmail.com> + +properties: + compatible: + const: fsl,sgtl5000 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: the clock provider of SYS_MCLK + + VDDA-supply: + description: the regulator provider of VDDA + + VDDIO-supply: + description: the regulator provider of VDDIO + + VDDD-supply: + description: the regulator provider of VDDD + + micbias-resistor-k-ohms: + description: The bias resistor to be used in kOhms. The resistor can take + values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not + mentioned or if the value is unknown, then micbias resistor is set to + 4k. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 0, 2, 4, 8 ] + + micbias-voltage-m-volts: + description: The bias voltage to be used in mVolts. The voltage can take + values from 1.25V to 3V by 250mV steps. If this node is not mentioned + or the value is unknown, then the value is set to 1.25V. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 1250, 1500, 1750, 2000, 2250, 2500, 2750, 3000 ] + + lrclk-strength: + description: | + The LRCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the + table below: + + VDDIO 1.8V 2.5V 3.3V + 0 = Disable + 1 = 1.66 mA 2.87 mA 4.02 mA + 2 = 3.33 mA 5.74 mA 8.03 mA + 3 = 4.99 mA 8.61 mA 12.05 mA + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 0, 1, 2, 3 ] + + sclk-strength: + description: | + The SCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the + table below: + + VDDIO 1.8V 2.5V 3.3V + 0 = Disable + 1 = 1.66 mA 2.87 mA 4.02 mA + 2 = 3.33 mA 5.74 mA 8.03 mA + 3 = 4.99 mA 8.61 mA 12.05 mA + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 0, 1, 2, 3 ] + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - VDDA-supply + - VDDIO-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@a { + compatible = "fsl,sgtl5000"; + reg = <0x0a>; + #sound-dai-cells = <0>; + clocks = <&clks 150>; + micbias-resistor-k-ohms = <2>; + micbias-voltage-m-volts = <2250>; + VDDA-supply = <&reg_3p3v>; + VDDIO-supply = <&reg_3p3v>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/trivial-devices.yaml b/Documentation/devicetree/bindings/trivial-devices.yaml index 4165352a590a..b7e94fe8643f 100644 --- a/Documentation/devicetree/bindings/trivial-devices.yaml +++ b/Documentation/devicetree/bindings/trivial-devices.yaml @@ -80,8 +80,6 @@ properties: - fsl,mpl3115 # MPR121: Proximity Capacitive Touch Sensor Controller - fsl,mpr121 - # SGTL5000: Ultra Low-Power Audio Codec - - fsl,sgtl5000 # G751: Digital Temperature Sensor and Thermal Watchdog with Two-Wire Interface - gmt,g751 # Infineon IR38064 Voltage Regulator -- 2.24.1

2 1

[PATCH v3 1/2] dt-bindings: tas2770: Convert tas2770 binding to yaml
by Dan Murphy 22 Jul '20

22 Jul '20

Convert the tas2770 binding to yaml format. Add in the reset-gpio to the binding as it is in the code but not documented in the binding. Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 --------- .../devicetree/bindings/sound/tas2770.yaml | 76 +++++++++++++++++++ 2 files changed, 76 insertions(+), 37 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2770.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt deleted file mode 100644 index ede6bb3d9637..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2770.txt +++ /dev/null @@ -1,37 +0,0 @@ -Texas Instruments TAS2770 Smart PA - -The TAS2770 is a mono, digital input Class-D audio amplifier optimized for -efficiently driving high peak power into small loudspeakers. -Integrated speaker voltage and current sense provides for -real time monitoring of loudspeaker behavior. - -Required properties: - - - compatible: - Should contain "ti,tas2770". - - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. - - ti,imon-slot-no:- TDM TX current sense time slot. - - ti,vmon-slot-no:- TDM TX voltage sense time slot. - -Optional properties: - -- interrupt-parent: the phandle to the interrupt controller which provides - the interrupt. -- interrupts: interrupt specification for data-ready. - -Examples: - - tas2770@4c { - compatible = "ti,tas2770"; - reg = <0x4c>; - #address-cells = <1>; - #size-cells = <0>; - interrupt-parent = <&msm_gpio>; - interrupts = <97 0>; - ti,asi-format = <0>; - ti,imon-slot-no = <0>; - ti,vmon-slot-no = <2>; - }; - diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml new file mode 100644 index 000000000000..8192450d72dc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019-20 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2770.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2770 Smart PA + +maintainers: + - Shi Fu <shifu0704(a)thundersoft.com> + +description: | + The TAS2770 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2770 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + reset-gpio: + description: GPIO used to reset the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + ti,asi-format: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Sets TDM RX capture edge. + enum: + - 0 # Rising edge + - 1 # Falling edge + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpio = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + -- 2.27.0

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[PATCH 1/2] dt-bindings: tas2770: Convert tas2770 binding to yaml
by Dan Murphy 22 Jul '20

22 Jul '20

Convert the tas2770 binding to yaml format. Add in the reset-gpio to the binding as it is in the code but not documented in the binding. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 --------- .../devicetree/bindings/sound/tas2770.yaml | 76 +++++++++++++++++++ 2 files changed, 76 insertions(+), 37 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2770.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt deleted file mode 100644 index ede6bb3d9637..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2770.txt +++ /dev/null @@ -1,37 +0,0 @@ -Texas Instruments TAS2770 Smart PA - -The TAS2770 is a mono, digital input Class-D audio amplifier optimized for -efficiently driving high peak power into small loudspeakers. -Integrated speaker voltage and current sense provides for -real time monitoring of loudspeaker behavior. - -Required properties: - - - compatible: - Should contain "ti,tas2770". - - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. - - ti,imon-slot-no:- TDM TX current sense time slot. - - ti,vmon-slot-no:- TDM TX voltage sense time slot. - -Optional properties: - -- interrupt-parent: the phandle to the interrupt controller which provides - the interrupt. -- interrupts: interrupt specification for data-ready. - -Examples: - - tas2770@4c { - compatible = "ti,tas2770"; - reg = <0x4c>; - #address-cells = <1>; - #size-cells = <0>; - interrupt-parent = <&msm_gpio>; - interrupts = <97 0>; - ti,asi-format = <0>; - ti,imon-slot-no = <0>; - ti,vmon-slot-no = <2>; - }; - diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml new file mode 100644 index 000000000000..8c667fd37a57 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019-20 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2770.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2770 Smart PA + +maintainers: + - Shi Fu <shifu0704(a)thundersoft.com> + +description: | + The TAS2770 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2770 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + reset-gpio: + description: GPIO used to reset the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + ti,asi-format: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Sets TDM RX capture edge. + enum: + - 0 # Rising edge + - 1 # Falling edge + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpio = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + -- 2.26.2

3 3

[PATCH for-next v2 1/2] ASoC: tas2770: Fix reset gpio property name
by Dan Murphy 22 Jul '20

22 Jul '20

Fix the reset property name when allocating the GPIO descriptor. The gpiod_get_optional appends either the -gpio or -gpios suffix to the name. Fixes: 1a476abc723e6 ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2770.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 54c8135fe43c..cf071121c839 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -758,8 +758,7 @@ static int tas2770_i2c_probe(struct i2c_client *client, } } - tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev, - "reset-gpio", + tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev, "reset", GPIOD_OUT_HIGH); if (IS_ERR(tas2770->reset_gpio)) { if (PTR_ERR(tas2770->reset_gpio) == -EPROBE_DEFER) { -- 2.27.0

2 2

[PATCH v7] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function
by Harsha Priya 21 Jul '20

21 Jul '20

kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs. The hw parameters are changed based on the codec DAI, the stream is intended for. The earlier approach to get snd_soc_dpcm was using container_of() macro on snd_pcm_hw_params. The structures have been modified over time and snd_soc_dpcm does not have snd_pcm_hw_params as a reference but as a copy. This causes the current driver to crash when used. This patch changes the way snd_soc_dpcm is extracted. The snd_soc_pcm_runtime holds 2 dpcm instances (one for playback and one for capture). The 2 codecs on this SSP are dmic and speakers. One is for capture and one is for playback respectively. Based on the direction of the stream, the snd_soc_dpcm is extracted from the snd_soc_pcm_runtime structure. Tested for all use cases of the driver. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> --- v1 -> v2: - Extract dmic from SSP0 as every BE should have own fixup function. v2 -> v3: - Restore naming in the dapm route table to not confuse with other drivers - Fixed indentations v3 -> v4: - Updated code and commit description according to solution proposed by Harsha v4 -> v5: - Cosmetic Changes v5 -> v6: - Dmic regression seen with v4 fixed - Using available routines for obtaining dpcm information v6 -> v7: - Updated comments - initilize rtd_dpcm variable - added break statement in the loop --- --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 38 ++++++++++++++----- 1 file changed, 29 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 584e4f9cedc2..9f4b949cc39c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -379,22 +379,42 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_interval *chan = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm = container_of( - params, struct snd_soc_dpcm, hw_params); - struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; - struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; + + /* + * The following loop will be called only for playback stream + * In this platform, there is only one playback device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * This following loop will be called only for capture stream + * In this platform, there is only one capture device on every SSP + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { + rtd_dpcm = dpcm; + break; + } + + /* + * The above 2 loops are mutually exclusive based on the strem direction, + * thus rtd_dpcm variable will never be overwritten + */ /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ - if (!strcmp(fe_dai_link->name, "Kbl Audio Port") || - !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) { + if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || + !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { rate->min = rate->max = 48000; chan->min = chan->max = 2; snd_mask_none(fmt); snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) { + } else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) { if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) chan->min = chan->max = 2; @@ -405,7 +425,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, * The speaker on the SSP0 supports S16_LE and not S24_LE. * thus changing the mask here */ - if (!strcmp(be_dai_link->name, "SSP0-Codec")) + if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); return 0; -- 2.17.1

2 1

Speaker pops with max98357a on rk3399-gru-kevin since v5.7
by Alper Nebi Yasak 21 Jul '20

21 Jul '20

Hi, I have been getting "pop" sounds from the speaker on my rk3399-gru-kevin for a while, and bisected it to 128f825aeab7 ("ASoC: max98357a: move control of SD_MODE to DAPM"), but looks like the pops were somewhat expected: On 12 Feb 2020 13:55:15 +0800, Tzung-Bi Shih wrote [1]: > Possible drawback: may generate pop noise when BCLK=on but LRCLK=off. > As the datasheet[2] mentioned: > > Do not remove LRCLK while BCLK is present. > > Removing LRCLK while BCLK is present can cause unexpected output behavior > > including a large DC output voltage. [1] https://lore.kernel.org/alsa-devel/20200212055517.122728-1-tzungbi@google.c… As of v5.8-rc5 I'm still getting the speaker pops. More info below, but not all pops coincide with "set sdmode" messages, and vice versa. Reverting that commit stops the pops, but then the "Speakers Switch" can no longer mute the speakers. I don't really know much about these things, could anyone have a look at it? I'll try to read relevant documentation and code to make sense of it, but wanted to send an email in case there is an immediately obvious solution (besides reverting) and maybe just nobody had the time to implement it yet. Thanks in advance. --- I've experimented a bit on a Debian userspace (with #define DEBUG 1 to get sdmode messages), here's what happens without PulseAudio: - When playback starts, speaker pops with "set sdmode to 1". - During playback, toggling "Speakers Switch" sets sdmode to 0/1 and mutes/unmutes speakers without any pops. - Pausing/resuming playback (like seeking on a media player) make pops without changing sdmode. Also pops when playback ends (e.g. when "speaker-test -l 1" quits). - A few seconds after playback ends, speaker pops for the last time with "set sdmode to 0". - Within that few seconds, toggling "Speakers Switch" sets sdmode to 0/1 and makes a pop each time. It's mostly the same with PulseAudio but: - Pause/resume during playback doesn't make any pops. - The penultimate pop happens a few seconds after playback stops, when PA says it's suspending the max98357a device (no sdmode changes). - The final pop comes a few seconds after that (still with "set sdmode to 0"). Even without any playback it makes several pops during boot when e.g. PulseAudio starts/stops running in the display manager or in the user session.

2 4

[PATCH 1/2] ASoC: qcom: dt-bindings: Add sc7180 machine bindings
by Cheng-Yi Chiang 21 Jul '20

21 Jul '20

Add devicetree bindings documentation file for sc7180 sound card. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- .../bindings/sound/qcom,sc7180.yaml | 123 ++++++++++++++++++ 1 file changed, 123 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,sc7180.yaml b/Documentation/devicetree/bindings/sound/qcom,sc7180.yaml new file mode 100644 index 000000000000..d60d2880d991 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,sc7180.yaml @@ -0,0 +1,123 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,sc7180.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm Technologies Inc. SC7180 ASoC sound card driver + +maintainers: + - Rohit kumar <rohitkr(a)codeaurora.org> + - Cheng-Yi Chiang <cychiang(a)chromium.org> + +description: | + This binding describes the SC7180 sound card, which uses LPASS for audio. + +definitions: + + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + dai: + type: object + properties: + sound-dai: + maxItems: 1 + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandle array of the codec or CPU DAI + + required: + - sound-dai + + unidirectional: + description: Specify direction of unidirectional dai link. + 0 for playback only. 1 for capture only. + $ref: /schemas/types.yaml#/definitions/uint32 + +properties: + compatible: + contains: + enum: + - qcom,sc7180-sndcard + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: |- + 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 source. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + +patternProperties: + "^dai-link-[0-9]+$": + description: | + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + $ref: "#/definitions/link-name" + + unidirectional: + $ref: "#/definitions/unidirectional" + + cpu: + $ref: "#/definitions/dai" + + codec: + $ref: "#/definitions/dai" + + required: + - link-name + - cpu + - codec + + additionalProperties: false + +examples: + + - | + snd { + compatible = "qcom,sc7180-sndcard"; + model = "sc7180-snd-card"; + + pinctrl-names = "default"; + pinctrl-0 = <&sec_mi2s_active &sec_mi2s_dout_active + &sec_mi2s_ws_active &pri_mi2s_active + &pri_mi2s_dout_active &pri_mi2s_ws_active + &pri_mi2s_din_active &pri_mi2s_mclk_active>; + + audio-routing = + "Headphone Jack", "HPOL", + "Headphone Jack", "HPOR"; + + dai-link-0 { + link-name = "MultiMedia0"; + cpu { + sound-dai = <&lpass_cpu 0>; + }; + + codec { + sound-dai = <&alc5682 0>; + }; + }; + + dai-link-1 { + link-name = "MultiMedia1"; + unidirectional = <0>; + cpu { + sound-dai = <&lpass_cpu 1>; + }; + + codec { + sound-dai = <&max98357a>; + }; + }; + }; -- 2.28.0.rc0.105.gf9edc3c819-goog

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[PATCH v2 0/9] soundwire: intel: revisit SHIM programming
by Bard Liao 21 Jul '20

21 Jul '20

This series does some cleanup, revisits SHIM programming sequences, and merges Soundwire interrupt handlers/threads. changes in v2: - Resume M device instead of S device in intel_master_process_wakeen_event function. See the comment in intel_master_process_wakeen_event() in detail. Bard Liao (2): soundwire: intel/cadence: merge Soundwire interrupt handlers/threads Soundwire: intel_init: save Slave(s) _ADR info in sdw_intel_ctx Pierre-Louis Bossart (6): soundwire: intel: reuse code for wait loops to set/clear bits soundwire: intel: revisit SHIM programming sequences. soundwire: intel: introduce a helper to arm link synchronization soundwire: intel: introduce helper for link synchronization soundwire: intel_init: add implementation of sdw_intel_enable_irq() soundwire: intel_init: use EXPORT_SYMBOL_NS Rander Wang (1): soundwire: intel: add wake interrupt support drivers/soundwire/cadence_master.c | 18 +- drivers/soundwire/cadence_master.h | 4 + drivers/soundwire/intel.c | 381 +++++++++++++++++++++------- drivers/soundwire/intel.h | 9 + drivers/soundwire/intel_init.c | 101 +++++++- include/linux/soundwire/sdw_intel.h | 2 + 6 files changed, 417 insertions(+), 98 deletions(-) -- 2.17.1

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[PATCH v2 0/2] snd-usb-6fire: firmware load and pulseaudio assumption
by René Herman 21 Jul '20

21 Jul '20

Hi Takashi. v2 of the earlier today sent out "set" for snd-usb-6fire, although now only two patches left, incorportaing the requested changes. The snd-usb-6fire driver for the TerraTec DMX 6Fire USB soundcard has been failing its firmware upload due to a non DMA-capable buffer on the stack. First of the patches kmallocs said bufffer instead and fixes the firmware upload; also makes it more alsa-generic by using the "goto out" structure. Note that it only looks a bit ungeneric as a patch due to also needing to at the same time unify its failure path: it's obvious when looked at post-apply. After that first patch the driver nominally works again but still has Pulseaudio crap out due to struct snd_pcm_hardware.channels_min=1 causing it to recognize it as a mono device only. Comparing with e.g. the TerraTec Aureon 7.1 Universe driver it seems that the solution is to simply set channels_min=2 as per the second patch. With these changes the card works again. Driver author Torsten Schenk has seen these and is fine with them: maintains an external driver with more options. I or he might time permitting start integrating more into the kernel driver over time. Regards, René René Herman (2): Move DMA-buffer off the stack Pulseaudio needs snd_pcm_hardware.channels_min > 1 sound/usb/6fire/firmware.c | 95 ++++++++++++++++++-------------------- sound/usb/6fire/pcm.c | 2 +- 2 files changed, 47 insertions(+), 50 deletions(-) -- 2.17.1

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[PATCH 0/3] snd-usb-6fire: firmware load and pulseaudio assumption
by René Herman 21 Jul '20

21 Jul '20

Hi Takashi. I sent these out a week ago to alsa-devel (only) but am not sure they got anywhere, perhaps due to at least the alsa-devel archive scrubbing the text/x-patch attachments as "non-text". The snd-usb-6fire driver for the TerraTec DMX 6Fire USB soundcard has been failing its firmware upload due to a non DMA-capable buffer on the stack. First of the patches kmallocs said bufffer instead and fixes the firmware upload. After that first patch the driver nominally works again but still has Pulseaudio crap out due to struct snd_pcm_hardware.channels_min=1 causing it to recognize it as a mono device only. Comparing with e.g. the TerraTec Aureon 7.1 Universe driver it seems that the solution is to simply set channels_min=2 as per the second patch. Third patch unmarks the snd_pcm_hardware struct const as it it is in fact changed in usb6fire_pcm_open(), even though through a pointer, and is supposedly trivial. With these changes the card works again. Driver author Torsten Schenk has seen these and is fine with them: maintains an external driver with more options. I or he might time permitting start integrating more into the kernel driver over time. Regards, René René Herman (3): snd-usb-6fire: Move DMA-buffer off of the stack snd-usb-6fire: Pulseaudio needs snd_pcm_hardware.channels_min > 1 snd-usb-6fire: Unmark struct snd_pcm_hardware const firmware.c | 95 ++++++++++++++++++++++++++---------------------------- pcm.c | 4 +-- 2 files changed, 48 insertions(+), 51 deletions(-) -- 2.17.1

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