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

[PATCH] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 01 May '20

01 May '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> --- .../bindings/sound/audio-graph-card.txt | 337 -------------- .../bindings/sound/audio-graph-card.yaml | 416 ++++++++++++++++++ 2 files changed, 416 insertions(+), 337 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-card.txt deleted file mode 100644 index 269682619a70..000000000000 --- a/Documentation/devicetree/bindings/sound/audio-graph-card.txt +++ /dev/null @@ -1,337 +0,0 @@ -Audio Graph Card: - -Audio Graph Card specifies audio DAI connections of SoC <-> codec. -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 - -Below are same as Simple-Card. - -- label -- widgets -- routing -- dai-format -- frame-master -- bitclock-master -- bitclock-inversion -- frame-inversion -- mclk-fs -- hp-det-gpio -- mic-det-gpio -- dai-tdm-slot-num -- dai-tdm-slot-width -- clocks / system-clock-frequency - -Required properties: - -- compatible : "audio-graph-card"; -- dais : list of CPU DAI port{s} - -Optional properties: -- pa-gpios: GPIO used to control external amplifier. - ------------------------ -Example: Single DAI case ------------------------ - - sound_card { - compatible = "audio-graph-card"; - - dais = <&cpu_port>; - }; - - dai-controller { - ... - cpu_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - - dai-format = "left_j"; - ... - }; - }; - }; - - audio-codec { - ... - port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; - }; - ------------------------ -Example: Multi DAI case ------------------------ - - sound-card { - compatible = "audio-graph-card"; - - label = "sound-card"; - - dais = <&cpu_port0 - &cpu_port1 - &cpu_port2>; - }; - - audio-codec@0 { - ... - port { - codec0_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint0>; - }; - }; - }; - - audio-codec@1 { - ... - port { - codec1_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint1>; - }; - }; - }; - - audio-codec@2 { - ... - port { - codec2_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint2>; - }; - }; - }; - - dai-controller { - ... - ports { - cpu_port0: port@0 { - cpu_endpoint0: endpoint { - remote-endpoint = <&codec0_endpoint>; - - dai-format = "left_j"; - ... - }; - }; - cpu_port1: port@1 { - cpu_endpoint1: endpoint { - remote-endpoint = <&codec1_endpoint>; - - dai-format = "i2s"; - ... - }; - }; - cpu_port2: port@2 { - cpu_endpoint2: endpoint { - remote-endpoint = <&codec2_endpoint>; - - dai-format = "i2s"; - ... - }; - }; - }; - }; - - ------------------------ -Example: Sampling Rate Conversion ------------------------ - - sound_card { - compatible = "audio-graph-card"; - - label = "sound-card"; - prefix = "codec"; - routing = "codec Playback", "DAI0 Playback", - "DAI0 Capture", "codec Capture"; - convert-rate = <48000>; - - dais = <&cpu_port>; - }; - - audio-codec { - ... - port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; - }; - - dai-controller { - ... - cpu_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - - dai-format = "left_j"; - ... - }; - }; - }; - ------------------------ -Example: 2 CPU 1 Codec (Mixing) ------------------------ - - sound_card { - compatible = "audio-graph-card"; - - label = "sound-card"; - routing = "codec Playback", "DAI0 Playback", - "codec Playback", "DAI1 Playback", - "DAI0 Capture", "codec Capture"; - - dais = <&cpu_port>; - }; - - audio-codec { - ... - - audio-graph-card,prefix = "codec"; - audio-graph-card,convert-rate = <48000>; - port { - reg = <0>; - codec_endpoint0: endpoint@0 { - remote-endpoint = <&cpu_endpoint0>; - }; - codec_endpoint1: endpoint@1 { - remote-endpoint = <&cpu_endpoint1>; - }; - }; - }; - - dai-controller { - ... - cpu_port: port { - cpu_endpoint0: endpoint@0 { - remote-endpoint = <&codec_endpoint0>; - - dai-format = "left_j"; - ... - }; - cpu_endpoint1: endpoint@1 { - remote-endpoint = <&codec_endpoint1>; - - dai-format = "left_j"; - ... - }; - }; - }; - ------------------------ -Example: Multi DAI with DPCM ------------------------ - - CPU0 ------ ak4613 - CPU1 ------ HDMI - CPU2 ------ PCM3168A-p /* DPCM 1ch/2ch */ - CPU3 --/ /* DPCM 3ch/4ch */ - CPU4 --/ /* DPCM 5ch/6ch */ - CPU5 --/ /* DPCM 7ch/8ch */ - CPU6 ------ PCM3168A-c - - sound_card: sound { - compatible = "audio-graph-card"; - - label = "sound-card"; - - routing = "pcm3168a Playback", "DAI2 Playback", - "pcm3168a Playback", "DAI3 Playback", - "pcm3168a Playback", "DAI4 Playback", - "pcm3168a Playback", "DAI5 Playback"; - - dais = <&snd_port0 /* ak4613 */ - &snd_port1 /* HDMI0 */ - &snd_port2 /* pcm3168a playback */ - &snd_port3 /* pcm3168a capture */ - >; - }; - - ak4613: codec@10 { - ... - port { - ak4613_endpoint: endpoint { - remote-endpoint = <&rsnd_endpoint0>; - }; - }; - }; - - pcm3168a: audio-codec@44 { - ... - audio-graph-card,prefix = "pcm3168a"; - audio-graph-card,convert-channels = <8>; /* TDM Split */ - ports { - port@0 { - reg = <0>; - pcm3168a_endpoint_p1: endpoint@1 { - remote-endpoint = <&rsnd_endpoint2>; - ... - }; - pcm3168a_endpoint_p2: endpoint@2 { - remote-endpoint = <&rsnd_endpoint3>; - ... - }; - pcm3168a_endpoint_p3: endpoint@3 { - remote-endpoint = <&rsnd_endpoint4>; - ... - }; - pcm3168a_endpoint_p4: endpoint@4 { - remote-endpoint = <&rsnd_endpoint5>; - ... - }; - }; - port@1 { - reg = <1>; - pcm3168a_endpoint_c: endpoint { - remote-endpoint = <&rsnd_endpoint6>; - ... - }; - }; - }; - }; - - &sound { - ports { - snd_port0: port@0 { - rsnd_endpoint0: endpoint { - remote-endpoint = <&ak4613_endpoint>; - ... - }; - }; - snd_port1: port@1 { - rsnd_endpoint1: endpoint { - remote-endpoint = <&dw_hdmi0_snd_in>; - ... - }; - }; - snd_port2: port@2 { - #address-cells = <1>; - #size-cells = <0>; - rsnd_endpoint2: endpoint@2 { - remote-endpoint = <&pcm3168a_endpoint_p1>; - ... - }; - rsnd_endpoint3: endpoint@3 { - remote-endpoint = <&pcm3168a_endpoint_p2>; - ... - }; - rsnd_endpoint4: endpoint@4 { - remote-endpoint = <&pcm3168a_endpoint_p3>; - ... - }; - rsnd_endpoint5: endpoint@5 { - remote-endpoint = <&pcm3168a_endpoint_p4>; - ... - }; - }; - snd_port3: port@6 { - rsnd_endpoint6: endpoint { - remote-endpoint = <&pcm3168a_endpoint_c>; - ... - }; - }; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/audio-graph-card.yaml b/Documentation/devicetree/bindings/sound/audio-graph-card.yaml new file mode 100644 index 000000000000..fd0ac87ee2ed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/audio-graph-card.yaml @@ -0,0 +1,416 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/audio-graph-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio Graph Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +definitions: + endpoint: + type: object + properties: + remote-endpoint: + maxItems: 1 + mclk-fs: + $ref: "simple-card.yaml#/definitions/mclk-fs" + frame-inversion: + $ref: "simple-card.yaml#/definitions/frame-inversion" + bitclock-inversion: + $ref: "simple-card.yaml#/definitions/bitclock-inversion" + frame-master: + $ref: "simple-card.yaml#/definitions/frame-master" + bitclock-master: + $ref: "simple-card.yaml#/definitions/bitclock-master" + dai-format: + $ref: "simple-card.yaml#/definitions/format" + convert-rate: + $ref: "simple-card.yaml#/definitions/convert-rate" + convert-channels: + $ref: "simple-card.yaml#/definitions/convert-channels" + required: + - remote-endpoint + + port: + description: single OF-Graph subnode + type: object + properties: + prefix: + $ref: "simple-card.yaml#/definitions/prefix" + convert-rate: + $ref: "simple-card.yaml#/definitions/convert-rate" + convert-channels: + $ref: "simple-card.yaml#/definitions/convert-channels" + endpoint: + $ref: "#/definitions/endpoint" + + ports: + description: multi OF-Graph subnode + type: object + properties: + prefix: + $ref: "simple-card.yaml#/definitions/prefix" + convert-rate: + $ref: "simple-card.yaml#/definitions/convert-rate" + convert-channels: + $ref: "simple-card.yaml#/definitions/convert-channels" + patternProperties: + "^port(@[0-9a-f]+)$": + $ref: "#/definitions/port" + additionalProperties: false + +properties: + compatible: + contains: + enum: + - audio-graph-card + - audio-graph-scu-card + + dais: + $ref: /schemas/types.yaml#/definitions/phandle-array + label: + maxItems: 1 + prefix: + $ref: "simple-card.yaml#/definitions/prefix" + routing: + $ref: "simple-card.yaml#/definitions/routing" + widgets: + $ref: "simple-card.yaml#/definitions/widgets" + convert-rate: + $ref: "simple-card.yaml#/definitions/convert-rate" + convert-channels: + $ref: "simple-card.yaml#/definitions/convert-channels" + pa-gpios: + maxItems: 1 + hp-det-gpio: + maxItems: 1 + mic-det-gpio: + maxItems: 1 + +required: + - compatible + +additionalProperties: false + +examples: +#----------------------- +# Single DAI (_s) +#----------------------- + - | + sound { + compatible = "audio-graph-card"; + + dais = <&cpu_port_a>; + }; + + dai-controller { + /* + * dai-controller own settings + */ + + cpu_port_s: port { + cpu_endpoint_s: endpoint { + remote-endpoint = <&codec_endpoint_s>; + dai-format = "left_j"; + /* ... */ + }; + }; + }; + + codec { + /* + * codec own settings + */ + + port { + codec_endpoint_s: endpoint { + remote-endpoint = <&cpu_endpoint_s>; + }; + }; + }; + +#----------------------- +# Multi DAI (_m) +#----------------------- + - | + sound { + compatible = "audio-graph-card"; + + label = "sound-card"; + + dais = <&cpu_port0 + &cpu_port1 + &cpu_port2>; + }; + + audio-codec@0 { + /* + * codec0 own settings + */ + + port { + codec_endpoint_m0: endpoint { + remote-endpoint = <&cpu_endpoint_m0>; + }; + }; + }; + + audio-codec@1 { + /* + * codec1 own settings + */ + + port { + codec_endpoint_m1: endpoint { + remote-endpoint = <&cpu_endpoint_m1>; + }; + }; + }; + + audio-codec@2 { + /* + * codec2 own settings + */ + + port { + codec_endpoint_m2: endpoint { + remote-endpoint = <&cpu_endpoint_m2>; + }; + }; + }; + + dai-controller { + /* + * dai-controller own settings + */ + + ports { + cpu_port_m0: port@0 { + cpu_endpoint_m0: endpoint { + remote-endpoint = <&codec_endpoint_m0>; + dai-format = "left_j"; + }; + }; + + cpu_port_m1: port@1 { + cpu_endpoint_m1: endpoint { + remote-endpoint = <&codec_endpoint_m1>; + dai-format = "i2s"; + }; + }; + + cpu_port_m2: port@2 { + cpu_endpoint_m2: endpoint { + remote-endpoint = <&codec_endpoint_m2>; + dai-format = "i2s"; + }; + }; + }; + }; + +#----------------------- +# Sampling Rate Conversion (_r) +#----------------------- + + - | + sound { + compatible = "audio-graph-card"; + + label = "sound-card"; + prefix = "codec"; + routing = "codec Playback", "DAI0 Playback", + "DAI0 Capture", "codec Capture"; + convert-rate = <48000>; + + dais = <&cpu_port_x>; + }; + + codec { + /* + * codec own settings + */ + + port { + codec_endpoint_r: endpoint { + remote-endpoint = <&cpu_endpoint_r>; + }; + }; + }; + + dai-controller { + /* + * dai-controller own settings + */ + + cpu_port_r: port { + cpu_endpoint_r: endpoint { + remote-endpoint = <&codec_endpoint_r>; + dai-format = "left_j"; + }; + }; + }; + +#----------------------- +# 2 CPU 1 Codec (Mixing) (_x) +#----------------------- + + - | + sound { + compatible = "audio-graph-card"; + + label = "sound-card"; + routing = "codec Playback", "DAI0 Playback", + "codec Playback", "DAI1 Playback", + "DAI0 Capture", "codec Capture"; + + dais = <&cpu_port_y>; + }; + + codec { + /* + * codec own settings + */ + + audio-graph-card,prefix = "codec"; + audio-graph-card,convert-rate = <48000>; + + port { + codec_endpoint_x0: endpoint@0 { + remote-endpoint = <&cpu_endpoint_x0>; + }; + codec_endpoint_x1: endpoint@1 { + remote-endpoint = <&cpu_endpoint_x1>; + }; + }; + }; + + dai-controller { + /* + * dai-controller own settings + */ + + port { + cpu_endpoint_x0: endpoint@0 { + remote-endpoint = <&codec_endpoint_x0>; + dai-format = "left_j"; + }; + cpu_endpoint_x1: endpoint@1 { + remote-endpoint = <&codec_endpoint_x1>; + dai-format = "left_j"; + }; + }; + }; + +#----------------------- +# Multi DAI with DPCM +# +# CPU0 ------ ak4613 +# CPU1 ------ HDMI +# CPU2 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU3 --/ /* DPCM 3ch/4ch */ +# CPU4 --/ /* DPCM 5ch/6ch */ +# CPU5 --/ /* DPCM 7ch/8ch */ +# CPU6 ------ PCM3168A-c +#----------------------- + - | + + sound { + compatible = "audio-graph-card"; + label = "sound-card"; + routing = "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback", + "pcm3168a Playback", "DAI5 Playback"; + + dais = <&snd_port0 /* ak4613 */ + &snd_port1 /* HDMI0 */ + &snd_port2 /* pcm3168a playback */ + &snd_port3>; /* pcm3168a capture */ + }; + + ak4613: codec { + /* + * ak4613 own settings + */ + + port { + ak4613_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint0>; + }; + }; + }; + + pcm3168a: audio-codec@44 { + /* + * pcm3168a own settings + */ + + audio-graph-card,prefix = "pcm3168a"; + audio-graph-card,convert-channels = <8>; /* TDM Split */ + ports { + #address-cells = <1>; + #size-cells = <0>; + port@0 { + reg = <0>; + pcm3168a_endpoint_p1: endpoint@1 { + remote-endpoint = <&rsnd_endpoint2>; + }; + pcm3168a_endpoint_p2: endpoint@2 { + remote-endpoint = <&rsnd_endpoint3>; + }; + pcm3168a_endpoint_p3: endpoint@3 { + remote-endpoint = <&rsnd_endpoint4>; + }; + pcm3168a_endpoint_p4: endpoint@4 { + remote-endpoint = <&rsnd_endpoint5>; + }; + }; + port@1 { + reg = <1>; + pcm3168a_endpoint_c: endpoint { + remote-endpoint = <&rsnd_endpoint6>; + }; + }; + }; + }; + + dai-controller { + /* + * dai-controller own settings + */ + + ports { + snd_port0: port@0 { + rsnd_endpoint0: endpoint { + remote-endpoint = <&ak4613_endpoint>; + }; + }; + snd_port1: port@1 { + rsnd_endpoint1: endpoint { + remote-endpoint = <&dw_hdmi0_snd_in>; + }; + }; + snd_port2: port@2 { + rsnd_endpoint2: endpoint@2 { + remote-endpoint = <&pcm3168a_endpoint_p1>; + }; + rsnd_endpoint3: endpoint@3 { + remote-endpoint = <&pcm3168a_endpoint_p2>; + }; + rsnd_endpoint4: endpoint@4 { + remote-endpoint = <&pcm3168a_endpoint_p3>; + }; + rsnd_endpoint5: endpoint@5 { + remote-endpoint = <&pcm3168a_endpoint_p4>; + }; + }; + snd_port3: port@6 { + rsnd_endpoint6: endpoint { + remote-endpoint = <&pcm3168a_endpoint_c>; + }; + }; + }; + }; -- 2.17.1

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[PATCH] ASoC: dt-bindings: ak4613: switch to yaml base Documentation
by Kuninori Morimoto 01 May '20

01 May '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/ak4613.txt | 27 -------- .../devicetree/bindings/sound/ak4613.yaml | 65 +++++++++++++++++++ 2 files changed, 65 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..b5929239505e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: GPL-2.0 +%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 + + # for OF-graph + port: + $ref: "audio-graph-card.yaml#definitions/port" + ports: + $ref: "audio-graph-card.yaml#definitions/ports" + +# use patternProperties to avoid naming "xxx,yyy" issue +patternProperties: + "^asahi-kasei,in1-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,in2-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out1-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out2-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out3-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out4-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out5-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out6-single-end$": + $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>; + }; + }; -- 2.17.1

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TLV320AIC31xx recovery after a SPKVDD power glitch
by Dexter Travis 01 May '20

01 May '20

I have a production agriculture display system using a TLV320AIC3120 for sound output to a speaker built into the monitor. The primary processor is an NXP IMX6 Dual Plus. The system has an internal battery backup but not all voltage rails are backed by the battery. In particular SPKVDD the 5V rail to the TLV320AIC3120 is not backed up. The rest of the voltage rails to this device are backed by battery. After a power glitch we do not emit sound. Using dynamic debug and adding some additional messages to the log It appears the driver is working the same both before and after the power glitch. We developed this a few years ago using the Freescale/NXP (fsl) eureka-tlv320 (with slight modifications for the tlv320aic31xx) and tlv320aic31xx.c drivers. As an experiment i compiled both the eukrea-tlv320 and the tlv320aic31xx drivers as modules. This worked as expected before the glitch. After a glitch if I manually remove the modules and re-insert them sound can be restored. We have a user space application that can detect the power loss and if the loss is persistent will gracefully shutdown the system. When power is restored we would like to use this to tell the driver to perform some kind of restorative action. I have tried triggering through sysfs various resets to the tlv320aic3120 without success. So far only a full module remove of both eukrea and tlv320 will restore sound. remove and re-insert of either driver by itself does not work. I have a few questions and areas where I would like advice. 0) What am i missing or doing wrong? Should I be taking a different approach altogether to this problem? 1) Should I migrate to simple-audio-card instead of eukrea-tlv320? 2) Suggestions on what needs to be reset/restored after the power rail comes back? 3) Suggestions on how best to trigger the reset/restore from either user space or the kernel? Below is the relevant sections of our devicetree. sound { compatible = "eukrea,asoc-tlv320"; eukrea,model = "pp-gen3display-tlv320aic3120"; ssi-controller = <&ssi2>; fsl,mux-int-port = <2>; fsl,mux-ext-port = <3>; clocks = <&clks IMX6QDL_CLK_CKO>; }; &i2c1 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c1>; status = "okay"; tlv320aic3120: tlv320aic3120@18 { reg = <0x18>; }; ... &ssi2 { codec-handle = <&tlv320aic3120>; status = "okay"; }; &audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; }; pinctrl_audmux: audmuxgrp { fsl,pins = < MX6QDL_PAD_CSI0_DAT7__AUD3_RXD 0x130b0 MX6QDL_PAD_CSI0_DAT4__AUD3_TXC 0x130b0 MX6QDL_PAD_CSI0_DAT5__AUD3_TXD 0x110b0 MX6QDL_PAD_CSI0_DAT6__AUD3_TXFS 0x130b0 MX6QDL_PAD_GPIO_2__GPIO1_IO02 0x17059 MX6QDL_PAD_CSI0_MCLK__CCM_CLKO1 0x130b0 >; }; Attached is our changes to the eukrea driver to support the TLV320AIC31xx in our system. Thank you, Dexter Travis

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[PATCH] ASoC: SOF: Update correct LED status at the first time usage of update_mute_led()
by Kai-Heng Feng 01 May '20

01 May '20

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> --- 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; -- 2.17.1

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[PATCH] Add control message quirk delay for Kingston HyperX headset
by Jesus Ramos 01 May '20

01 May '20

Signed-off-by: Jesus Ramos <jesus-ramos(a)live.com> --- sound/usb/quirks.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 351ba214a9d3..0af54728552c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1636,13 +1636,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) msleep(20); - /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here, - * otherwise requests like get/set frequency return as failed despite - * actually succeeding. + /* Zoom R16/24, Logitech H650e, Jabra 550a, Kingston HyperX needs a tiny + * delay here, otherwise requests like get/set frequency return as + * failed despite actually succeeding. */ if ((chip->usb_id == USB_ID(0x1686, 0x00dd) || chip->usb_id == USB_ID(0x046d, 0x0a46) || - chip->usb_id == USB_ID(0x0b0e, 0x0349)) && + chip->usb_id == USB_ID(0x0b0e, 0x0349) || + chip->usb_id == USB_ID(0x0951, 0x16ad)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) usleep_range(1000, 2000); } -- 2.24.0

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[PATCH AUTOSEL 5.6 12/38] ALSA: hda: Skip controller resume if not needed
by Sasha Levin 01 May '20

01 May '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c4c8dd6ef807663e42a5f04ea77cd62029eb99fa ] The HD-audio controller does system-suspend and resume operations by directly calling its helpers __azx_runtime_suspend() and __azx_runtime_resume(). However, in general, we don't have to resume always the device fully at the system resume; typically, if a device has been runtime-suspended, we can leave it to runtime resume. Usually for achieving this, the driver would call pm_runtime_force_suspend() and pm_runtime_force_resume() pairs in the system suspend and resume ops. Unfortunately, this doesn't work for the resume path in our case. For handling the jack detection at the system resume, a child codec device may need the (literally) forcibly resume even if it's been runtime-suspended, and for that, the controller device must be also resumed even if it's been suspended. This patch is an attempt to improve the situation. It replaces the direct __azx_runtime_suspend()/_resume() calls with with pm_runtime_force_suspend() and pm_runtime_force_resume() with a slight trick as we've done for the codec side. More exactly: - azx_has_pm_runtime() check is dropped from azx_runtime_suspend() and azx_runtime_resume(), so that it can be properly executed from the system-suspend/resume path - The WAKEEN handling depends on the card's power state now; it's set and cleared only for the runtime-suspend - azx_resume() checks whether any codec may need the forcible resume beforehand. If the forcible resume is required, it does temporary PM refcount up/down for actually triggering the runtime resume. - A new helper function, hda_codec_need_resume(), is introduced for checking whether the codec needs a forcible runtime-resume, and the existing code is rewritten with that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-6-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/hda_codec.h | 5 +++++ sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_intel.c | 38 +++++++++++++++++++++++++++----------- 3 files changed, 33 insertions(+), 12 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 3ee8036f5436d..225154a4f2ed0 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -494,6 +494,11 @@ void snd_hda_update_power_acct(struct hda_codec *codec); static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} #endif +static inline bool hda_codec_need_resume(struct hda_codec *codec) +{ + return !codec->relaxed_resume && codec->jacktbl.used; +} + #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 53e7732ef7520..aed1f8188e662 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2951,7 +2951,7 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used; + bool forced_resume = hda_codec_need_resume(codec); int ret; /* The get/put pair below enforces the runtime resume even if the diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index aa0be85614b6c..02c6308502b1e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1027,7 +1027,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - __azx_runtime_suspend(chip); + pm_runtime_force_suspend(dev); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1044,7 +1044,9 @@ static int azx_suspend(struct device *dev) static int azx_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); + struct hda_codec *codec; struct azx *chip; + bool forced_resume = false; if (!azx_is_pm_ready(card)) return 0; @@ -1055,7 +1057,20 @@ static int azx_resume(struct device *dev) chip->msi = 0; if (azx_acquire_irq(chip, 1) < 0) return -EIO; - __azx_runtime_resume(chip, false); + + /* check for the forced resume */ + list_for_each_codec(codec, &chip->bus) { + if (hda_codec_need_resume(codec)) { + forced_resume = true; + break; + } + } + + if (forced_resume) + pm_runtime_get_noresume(dev); + pm_runtime_force_resume(dev); + if (forced_resume) + pm_runtime_put(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); @@ -1102,12 +1117,12 @@ static int azx_runtime_suspend(struct device *dev) if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; /* enable controller wake up event */ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | - STATESTS_INT_MASK); + if (snd_power_get_state(card) == SNDRV_CTL_POWER_D0) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + } __azx_runtime_suspend(chip); trace_azx_runtime_suspend(chip); @@ -1118,17 +1133,18 @@ static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; + bool from_rt = snd_power_get_state(card) == SNDRV_CTL_POWER_D0; if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; - __azx_runtime_resume(chip, true); + __azx_runtime_resume(chip, from_rt); /* disable controller Wake Up event*/ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & - ~STATESTS_INT_MASK); + if (from_rt) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); + } trace_azx_runtime_resume(chip); return 0; -- 2.20.1

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[PATCH AUTOSEL 4.9 03/13] ALSA: hda: Keep the controller initialization even if no codecs found
by Sasha Levin 01 May '20

01 May '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 9479e75fca370a5220784f7596bf598c4dad0b9b ] Currently, when the HD-audio controller driver doesn't detect any codecs, it tries to abort the probe. But this abort happens at the delayed probe, i.e. the primary probe call already returned success, hence the driver is never unbound until user does so explicitly. As a result, it may leave the HD-audio device in the running state without the runtime PM. More badly, if the device is a HD-audio bus that is tied with a GPU, GPU cannot reach to the full power down and consumes unnecessarily much power. This patch changes the logic after no-codec situation; it continues probing without the further codec initialization but keep the controller driver running normally. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Tested-by: Roy Spliet <nouveau(a)spliet.org> Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5a578ebca1055..8b9d1c6574e43 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1806,7 +1806,7 @@ static int azx_first_init(struct azx *chip) /* codec detection */ if (!azx_bus(chip)->codec_mask) { dev_err(card->dev, "no codecs found!\n"); - return -ENODEV; + /* keep running the rest for the runtime PM */ } if (azx_acquire_irq(chip, 0) < 0) @@ -2129,9 +2129,11 @@ static int azx_probe_continue(struct azx *chip) #endif /* create codec instances */ - err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); - if (err < 0) - goto out_free; + if (bus->codec_mask) { + err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); + if (err < 0) + goto out_free; + } #ifdef CONFIG_SND_HDA_PATCH_LOADER if (chip->fw) { @@ -2145,7 +2147,7 @@ static int azx_probe_continue(struct azx *chip) #endif } #endif - if ((probe_only[dev] & 1) == 0) { + if (bus->codec_mask && !(probe_only[dev] & 1)) { err = azx_codec_configure(chip); if (err < 0) goto out_free; -- 2.20.1

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Functionality of pcm_notify in snd-aloop?
by Pavel Hofman 30 Apr '20

30 Apr '20

Hi, Please how is the module params pcm_notify supposed to be used, to do what the documentation says: Break capture when PCM format/rate/channels changes? Breaking capture side operation when the playback side changes the params is very useful, but I cannot find a way to use this param properly. When the capture side is open, the playback side cannot use a different parameter than the one currently used by the capture side (the configuration space is limited) - how can the param be changed then? Thanks a lot for help. Best regards, Pavel.

4 9

[PATCH] ALSA: pcm: oss: Avoid plugin buffer overflow
by Takashi Iwai 30 Apr '20

30 Apr '20

Each OSS PCM plugins allocate its internal buffer per pre-calculation of the max buffer size through the chain of plugins (calling src_frames and dst_frames callbacks). This works for most plugins, but the rate plugin might behave incorrectly. The calculation in the rate plugin involves with the fractional position, i.e. it may vary depending on the input position. Since the buffer size pre-calculation is always done with the offset zero, it may return a shorter size than it might be; this may result in the out-of-bound access as spotted by fuzzer. This patch addresses those possible buffer overflow accesses by simply setting the upper limit per the given buffer size for each plugin before src_frames() and after dst_frames() calls. Reported-by: syzbot+e1fe9f44fb8ecf4fb5dd(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000b25ea005a02bcf21@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_plugin.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 240e4702c098..c9401832967c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -209,6 +209,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p if (stream == SNDRV_PCM_STREAM_PLAYBACK) { plugin = snd_pcm_plug_last(plug); while (plugin && drv_frames > 0) { + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) drv_frames = plugin->src_frames(plugin, drv_frames); @@ -220,6 +222,8 @@ snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *plug, snd_p plugin_next = plugin->next; if (plugin->dst_frames) drv_frames = plugin->dst_frames(plugin, drv_frames); + if (drv_frames > plugin->buf_frames) + drv_frames = plugin->buf_frames; plugin = plugin_next; } } else @@ -248,11 +252,15 @@ snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *plug, snd_pc if (frames < 0) return frames; } + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin = plugin_next; } } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { plugin = snd_pcm_plug_last(plug); while (plugin) { + if (frames > plugin->buf_frames) + frames = plugin->buf_frames; plugin_prev = plugin->prev; if (plugin->src_frames) { frames = plugin->src_frames(plugin, frames); -- 2.16.4

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[PATCH 1/3] ALSA: hda/realtek - Introduce polarity for micmute LED GPIO
by Kai-Heng Feng 30 Apr '20

30 Apr '20

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> --- 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 c16f63957c5a..3150a61befc5 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; @@ -4075,11 +4076,9 @@ 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 */ } @@ -4090,7 +4089,8 @@ 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 */ @@ -4099,6 +4099,7 @@ 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); } @@ -5795,7 +5796,8 @@ 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. -- 2.17.1

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