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[RFT] ALSA control service programs for Digidesign Digi 002/003 family and Tascam FireWire series
by Takashi Sakamoto 15 Jul '20

15 Jul '20

Hi, This is request-for-test (RFT) to the ALSA control service programs for Digidesign Digi 002/003 family and Tascam FireWire series. The programs are available in 'topic/services-for-dg00x-tascam' of below repository, and the RFT is corresponding to Pull Request #1: * https://github.com/alsa-project/snd-firewire-ctl-services/ Background ---------- The models in above family and series are already supported by drivers in ALSA firewire stack for their isochronous packet stream functionality. Although the drivers allow userspace applications to process PCM frames and MIDI messages in the packet streaming, they doesn't support the other functionalities such as volume control since it can be achieved by any usespace applications of Linux firewire subsystem. The service programs added by this patchset are for the supplemental functionality to control the models by asynchronous packet transaction or to process a part of data transferred by the isochronous packet stream. In detail, please read README.rst in the repository[1]. You can see some illustrations. How to build ------------- The project depends on several libraries in below repositories: * glib[2] * alsa-gobject v0.1.0[3] * libhinawa v2.0.0[4] Before building it, please install them with gobject-introspection support. The project is written by Rust programming language[5] and packaged by cargo[6]. To build, just run below commands in the environment to connect to internet: ``` $ git clone https://github.com/alsa-project/snd-firewire-ctl-services.git -b topic/services-for-dg00x-tascam $ cd snd-firewire-ctl-services $ cargo build ``` All of required crates are downloaded automatically, then start build. Executables ----------- The added executables are listed below: * snd-firewire-digi00x-ctl-service * snd-firewire-tascam-ctl-service When running, the programs add and maintain control elements, then enter event loop to dispatch events. Thes elements are visible and manipulable for any userspace applications. The program dispatch any manipulation event for transactions to control actual device. For console model and FE-8 in Tascam FireWire series, the program adds and maintains port in ALSA Sequencer. They have command line options. For all models of Digi 002/003 family, the executable has an option for the numerical ID of sound card. ``` Usage: snd-firewire-digi00x-ctl-service CARD_ID where: CARD_ID: The numerical ID of sound card. ``` For models of Tascam FireWire series except for FE-8, the executable should run with 'snd' as its first option and the numerical ID of sound card as second option. Tascam FE-8 is supported by the service program but not supported by ALSA firewire stack since it doesn't handle isochronous packet streaming. For FE-8, the executable should run with 'fw' as its first option and the numerical ID of firewire character device. ``` Usage: snd-firewire-tascam-ctl-service SUBSYSTEM ID where: SUBSYSTEM: The name of subsystem; 'snd' or 'fw' ID: The numerical ID of sound card or fw character device ``` It's easy to run the executables by cargo, like: ``` $ cargo run --bin snd-firewire-digi00x-ctl-service 2 ``` Receiving SIGTERM signal terminates the event loop, then the program finishes. Access permissions for relevant character devices -------------------------------------------------- The executables manipulate below character devices to dispatch, convert, and emit event from end to end. The '%u' is the numerical number in each subsystem: * ALSA control character device (/dev/snd/controlC%u) * ALSA hwdep character device (/dev/snd/hwC%uD%u) * Linux firewire character device (/dev/fw%u) Additionally, the service program for Tascam FireWire series manipulates ALSA sequencer character device (/dev/snd/seq) to handle events of control surface for console model. Users should pay enough attention to access permission for the above character devices when running the service program. Feedback -------- Any feedback is welcome. For questions, please use mailing list in alsa-devel or alsa-users[7]. For issues, please use service of github.com[8]. Issues ------ The name of control elements are not fixed yet since the convention of name for element set is not clear yet for recording equipment. Users may feel inconvenience to which channel corresponds to which physical port. Furthermore, in the case that element set includes several elements, it's unclear that which element corresponds to which physical port as well. ALSA control core has no feature for the above issues at present. [1] https://github.com/alsa-project/snd-firewire-ctl-services [2] https://gitlab.gnome.org/GNOME/glib [3] https://github.com/alsa-project/alsa-gobject/ [4] https://github.com/alsa-project/libhinawa [5] https://www.rust-lang.org/ [6] https://doc.rust-lang.org/cargo/ [7] https://www.alsa-project.org/wiki/Mailing-lists [8] https://github.com/alsa-project/snd-firewire-ctl-services/issues Regards Takashi Sakamoto

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[PATCH v2] dt-bindings: ASoC: Convert UniPhier EVEA codec to json-schema
by Masahiro Yamada 14 Jul '20

14 Jul '20

Convert the UniPhier EVEA sound codec binding to DT schema format. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- Changes in v2: - Add schema for subnode 'port' .../sound/socionext,uniphier-evea.yaml | 70 +++++++++++++++++++ .../bindings/sound/uniphier,evea.txt | 26 ------- 2 files changed, 70 insertions(+), 26 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml delete mode 100644 Documentation/devicetree/bindings/sound/uniphier,evea.txt diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml new file mode 100644 index 000000000000..228168f685cf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-evea.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier EVEA SoC-internal sound codec + +maintainers: + - <alsa-devel(a)alsa-project.org> + +properties: + compatible: + const: socionext,uniphier-evea + + reg: + maxItems: 1 + + clock-names: + items: + - const: evea + - const: exiv + + clocks: + minItems: 2 + maxItems: 2 + + reset-names: + items: + - const: evea + - const: exiv + - const: adamv + + resets: + minItems: 3 + maxItems: 3 + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + type: object + properties: + endpoint: true + required: + - endpoint + +additionalProperties: false + +required: + - compatible + - reg + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + codec@57900000 { + compatible = "socionext,uniphier-evea"; + reg = <0x57900000 0x1000>; + clock-names = "evea", "exiv"; + clocks = <&sys_clk 41>, <&sys_clk 42>; + reset-names = "evea", "exiv", "adamv"; + resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; + #sound-dai-cells = <1>; + }; diff --git a/Documentation/devicetree/bindings/sound/uniphier,evea.txt b/Documentation/devicetree/bindings/sound/uniphier,evea.txt deleted file mode 100644 index 3f31b235f18b..000000000000 --- a/Documentation/devicetree/bindings/sound/uniphier,evea.txt +++ /dev/null @@ -1,26 +0,0 @@ -Socionext EVEA - UniPhier SoC internal codec driver - -Required properties: -- compatible : should be "socionext,uniphier-evea". -- reg : offset and length of the register set for the device. -- clock-names : should include following entries: - "evea", "exiv" -- clocks : a list of phandle, should contain an entry for each - entries in clock-names. -- reset-names : should include following entries: - "evea", "exiv", "adamv" -- resets : a list of phandle, should contain reset entries of - reset-names. -- #sound-dai-cells: should be 1. - -Example: - - codec { - compatible = "socionext,uniphier-evea"; - reg = <0x57900000 0x1000>; - clock-names = "evea", "exiv"; - clocks = <&sys_clk 41>, <&sys_clk 42>; - reset-names = "evea", "exiv", "adamv"; - resets = <&sys_rst 41>, <&sys_rst 42>, <&adamv_rst 0>; - #sound-dai-cells = <1>; - }; -- 2.25.1

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[PATCH v2 0/5] ASoC: sh: remove discriminatory terms
by Kuninori Morimoto 14 Jul '20

14 Jul '20

Renesas SH drivers are using discriminatory terms. This patch-set removes or changes it as much as possible. But, because DMA related API function name, it still exists. I hope all these are removed someday. v1 -> v2 - use "secondary" instead of "follower" - care siu/ssi drivers - tidyup git-log Link: https://lore.kernel.org/r/87r1tg3swv.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (5): ASoC: rsnd: don't use discriminatory terms for function names ASoC: rsnd: don't use discriminatory terms for comment ASoC: fsi: don't use discriminatory terms for comment ASoC: siu: don't use discriminatory terms for parameter ASoC: ssi: don't use discriminatory terms for debug log sound/soc/sh/fsi.c | 4 ++-- sound/soc/sh/rcar/core.c | 4 ++-- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/ssi.c | 28 ++++++++++++++-------------- sound/soc/sh/rcar/ssiu.c | 6 +++--- sound/soc/sh/siu_pcm.c | 6 +++--- sound/soc/sh/ssi.c | 2 +- 7 files changed, 26 insertions(+), 26 deletions(-) -- 2.25.1

2 6

[PATCH 0/3] ASoC: sh: remove discriminatory terms
by Kuninori Morimoto 14 Jul '20

14 Jul '20

rsnd / fsi are using discriminatory terms. This patch-set removes it as much as possible. But, because DMA related API function name, it still exists. I hope all these are removed someday. Kuninori Morimoto (3): ASoC: rsnd: don't use discriminatory terms for function names ASoC: rsnd: don't use discriminatory terms for comment ASoC: fsi: don't use discriminatory terms for comment sound/soc/sh/fsi.c | 4 ++-- sound/soc/sh/rcar/core.c | 4 ++-- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/ssi.c | 28 ++++++++++++++-------------- sound/soc/sh/rcar/ssiu.c | 6 +++--- 5 files changed, 22 insertions(+), 22 deletions(-) -- 2.25.1

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[PATCH RESEND] ASoC: rt5682: Enable Vref2 under using PLL2
by derek.fang＠realtek.com 14 Jul '20

14 Jul '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 1

[PATCH v3] dt-bindings: sound: convert rk3328 codec binding to yaml
by Katsuhiro Suzuki 14 Jul '20

14 Jul '20

This patch converts Rockchip rk3328 audio codec binding to DT schema. And adds description about "mclk" clock and fixes some errors in original example. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> Reviewed-by: Rob Herring <robh(a)kernel.org> --- .../bindings/sound/rockchip,rk3328-codec.txt | 28 -------- .../bindings/sound/rockchip,rk3328-codec.yaml | 69 +++++++++++++++++++ 2 files changed, 69 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt deleted file mode 100644 index 1ecd75d2032a..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ /dev/null @@ -1,28 +0,0 @@ -* Rockchip Rk3328 internal codec - -Required properties: - -- compatible: "rockchip,rk3328-codec" -- reg: physical base address of the controller and length of memory mapped - region. -- rockchip,grf: the phandle of the syscon node for GRF register. -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should be "pclk". -- spk-depop-time-ms: speak depop time msec. - -Optional properties: - -- mute-gpios: GPIO specifier for external line driver control (typically the - dedicated GPIO_MUTE pin) - -Example for rk3328 internal codec: - -codec: codec@ff410000 { - compatible = "rockchip,rk3328-codec"; - reg = <0x0 0xff410000 0x0 0x1000>; - rockchip,grf = <&grf>; - clocks = <&cru PCLK_ACODEC>; - clock-names = "pclk"; - mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; - spk-depop-time-ms = 100; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml new file mode 100644 index 000000000000..5b85ad5e4834 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + const: rockchip,rk3328-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + - description: clock for I2S master clock + + clock-names: + items: + - const: pclk + - const: mclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + spk-depop-time-ms: + default: 200 + description: + Speaker depop time in msec. + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + - "#sound-dai-cells" + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/clock/rk3328-cru.h> + codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0xff410000 0x1000>; + clocks = <&cru PCLK_ACODECPHY>, <&cru SCLK_I2S1>; + clock-names = "pclk", "mclk"; + rockchip,grf = <&grf>; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; + spk-depop-time-ms = <100>; + #sound-dai-cells = <0>; + }; -- 2.27.0

2 1

[PATCH v2] dt-bindings: ASoC: Convert UniPhier AIO audio system to json-schema
by Masahiro Yamada 14 Jul '20

14 Jul '20

Convert the UniPhier AIO audio system binding to DT schema format. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- Changes in v2: - Add schema for subnode 'port' .../sound/socionext,uniphier-aio.yaml | 81 +++++++++++++++++++ .../bindings/sound/uniphier,aio.txt | 45 ----------- 2 files changed, 81 insertions(+), 45 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml delete mode 100644 Documentation/devicetree/bindings/sound/uniphier,aio.txt diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml new file mode 100644 index 000000000000..4987eb91f2ab --- /dev/null +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml @@ -0,0 +1,81 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/socionext,uniphier-aio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: UniPhier AIO audio system + +maintainers: + - <alsa-devel(a)alsa-project.org> + +properties: + compatible: + enum: + - socionext,uniphier-ld11-aio + - socionext,uniphier-ld20-aio + - socionext,uniphier-pxs2-aio + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clock-names: + const: aio + + clocks: + maxItems: 1 + + reset-names: + const: aio + + resets: + maxItems: 1 + + socionext,syscon: + description: | + Specifies a phandle to soc-glue, which is used for changing mode of S/PDIF + signal pin to output from Hi-Z. This property is optional if you use I2S + signal pins only. + $ref: "/schemas/types.yaml#/definitions/phandle" + + "#sound-dai-cells": + const: 1 + +patternProperties: + "^port@[0-9]$": + type: object + properties: + endpoint: true + required: + - endpoint + +additionalProperties: false + +required: + - compatible + - reg + - interrupts + - clock-names + - clocks + - reset-names + - resets + - "#sound-dai-cells" + +examples: + - | + audio@56000000 { + compatible = "socionext,uniphier-ld20-aio"; + reg = <0x56000000 0x80000>; + interrupts = <0 144 4>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_aout>; + clock-names = "aio"; + clocks = <&sys_clk 40>; + reset-names = "aio"; + resets = <&sys_rst 40>; + #sound-dai-cells = <1>; + socionext,syscon = <&soc_glue>; + }; diff --git a/Documentation/devicetree/bindings/sound/uniphier,aio.txt b/Documentation/devicetree/bindings/sound/uniphier,aio.txt deleted file mode 100644 index 4ce68ed6f2f2..000000000000 --- a/Documentation/devicetree/bindings/sound/uniphier,aio.txt +++ /dev/null @@ -1,45 +0,0 @@ -Socionext UniPhier SoC audio driver - -The Socionext UniPhier audio subsystem consists of I2S and S/PDIF blocks in -the same register space. - -Required properties: -- compatible : should be one of the following: - "socionext,uniphier-ld11-aio" - "socionext,uniphier-ld20-aio" - "socionext,uniphier-pxs2-aio" -- reg : offset and length of the register set for the device. -- interrupts : should contain I2S or S/PDIF interrupt. -- pinctrl-names : should be "default". -- pinctrl-0 : defined I2S signal pins for an external codec chip. -- clock-names : should include following entries: - "aio" -- clocks : a list of phandle, should contain an entry for each - entry in clock-names. -- reset-names : should include following entries: - "aio" -- resets : a list of phandle, should contain an entry for each - entry in reset-names. -- #sound-dai-cells: should be 1. - -Optional properties: -- socionext,syscon: a phandle, should contain soc-glue. - The soc-glue is used for changing mode of S/PDIF signal pin - to Output from Hi-Z. This property is optional if you use - I2S signal pins only. - -Example: - audio { - compatible = "socionext,uniphier-ld20-aio"; - reg = <0x56000000 0x80000>; - interrupts = <0 144 4>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_aout>; - clock-names = "aio"; - clocks = <&sys_clk 40>; - reset-names = "aio"; - resets = <&sys_rst 40>; - #sound-dai-cells = <1>; - - socionext,syscon = <&sg>; - }; -- 2.25.1

3 2

ASoC: codecs: wm9712: Mux no paths errors
by Primoz Fiser 14 Jul '20

14 Jul '20

Hi all, I have a sound setup based on fsl_ssi + wm9712 codec. On every drivers load I get the following console error: >> [ 5666.776770] wm9712-codec wm9712-codec: ASoC: mux Capture Phone Mux has no paths >> [ 5666.784281] wm9712-codec wm9712-codec: ASoC: mux Differential Source has no paths What is the proper way to get rid of this errors? As a temporary solution I have ifdef-ed the following muxes from the driver like so: > @@ -364,8 +365,10 @@ SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, > &wm9712_out3_mux_controls), > SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0, > &wm9712_spk_mux_controls), > +#if 0 > SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0, > &wm9712_capture_phone_mux_controls), > +#endif > SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, > &wm9712_capture_selectl_controls), > SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, > @@ -374,8 +377,10 @@ SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0, > &wm9712_mic_src_controls), > SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0, > &wm9712_mic_src_controls), > +#ifdef 0 > SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, > &wm9712_diff_sel_controls), > +#endif > SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), > SND_SOC_DAPM_MIXER("Left HP Mixer", AC97_INT_PAGING, 9, 1, > &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls)), which works but feels like a dirty workaround, right? How can this be properly fixed in the wm9712 driver or somewhere else? BR, Primoz

2 1

[PATCH] ALSA: hda/tegra: Disable sync-write operation
by Jon Hunter 14 Jul '20

14 Jul '20

Commit f34a4c9dd4ca ("ALSA: hda: Enable sync-write operation as default for all controllers") enabled sync-write for all controllers and this is causing audio playback on the Tegra186 HDA device to fail. For now, disable sync-write support for Tegra to fix this. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 0cc5fad1af8a..5637f0129932 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -443,6 +443,7 @@ static int hda_tegra_create(struct snd_card *card, if (err < 0) return err; + chip->bus.core.sync_write = 0; chip->bus.core.needs_damn_long_delay = 1; chip->bus.core.aligned_mmio = 1; -- 2.17.1

2 1

[snd-usb-6fire] Unmark struct snd_pcm_hardware const
by René Herman 14 Jul '20

14 Jul '20

struct snd_pcm_hardware pcm_hw is in fact changed in usb6fire_pcm_open() Signed-off-by: René Herman <rene.herman(a)gmail.com>

1 0

[snd-usb-6fire] Pulseaudio needs snd_pcm_hardware.channels_min > 1
by René Herman 14 Jul '20

14 Jul '20

Pulseaudio with snd_pcm_hardware.channels_min=1 creates a mono device only. Signed-off-by: René Herman <rene.herman(a)gmail.com>

1 0

[snd-usb-6fire] Move DMA-buffer off of the stack
by René Herman 14 Jul '20

14 Jul '20

snd-usb-fire currently fails its firmware load with "transfer buffer not dma capable". Move said buffer off of the stack (has been coordinated with the original driver author Torsten Schenk) Signed-off-by: René Herman <rene.herman(a)gmail.com>

1 0

[PATCH] When snd_soc_card_jack_new is not called or the call fails, calling this function causes a null pointer access
by xingxing qiao 14 Jul '20

14 Jul '20

Signed-off-by: mnlife <mnlife(a)foxmail.com> ---  sound/soc/soc-jack.c | 2 +-  1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0f1820f36b4d..51b799ee98b7 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -44,7 +44,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)  unsigned int sync = 0;  int enable;   -if (!jack) +if (!jack || !jack->jack)  return;  trace_snd_soc_jack_report(jack, mask, status);   -- 2.17.1

2 1

[PATCH AUTOSEL 4.9 07/10] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 14 Jul '20

14 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2def4ad579ccf..f7bd91b713e83 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1655,33 +1655,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

1 0

[PATCH AUTOSEL 4.14 09/12] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 14 Jul '20

14 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9e8cfc409b4b3..04420a3bf5eb0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1805,33 +1805,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 09/13] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 14 Jul '20

14 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 419d099b5582b..1fc12e680f311 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1804,33 +1804,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 11/18] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 14 Jul '20

14 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e78c4367b6c86..820f534a67b1c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1817,33 +1817,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 12/19] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Sasha Levin 14 Jul '20

14 Jul '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ] When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 137d655fed8f8..e821c9df81070 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1804,33 +1804,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.25.1

1 0

[PATCH v2 00/10] topology: decode: Various fixes
by Piotr Maziarz 14 Jul '20

14 Jul '20

This series fixes various problems with topology decoding mechanism. Some of the problems were critical like improper memory management or infinite loops that were causing undefined behaviour or program crashes, while other resulted in losing some data during conversion. Bugs found while testing with Intel SST topologies. Changelog: v2: -Divide into more patches, critical fixes are in separate patches now -More specific descriptions -Fix a typo UML to UCM -Add error reporting in topology: decode: fix channel map memory allocation -Remove goto again in topology: Make buffer for saving dynamic size for better readability Piotr Maziarz (10): topology: decode: Fix channel map memory allocation topology: decode: Fix infinite loop in decoding enum control topology: decode: Remove decoding values for enum control topology: decode: Add enum control texts as separate element topology: decode: Fix printing texts section topology: decode: Change declaration of enum decoding function topology: decode: Fix decoding PCM formats and rates topology: decode: Print sig_bits field in PCM capabilities section topology: decode: Add DAI name printing topology: Make buffer for saving dynamic size src/topology/ctl.c | 51 ++++++++++++++++++++++------------------------- src/topology/dapm.c | 3 +-- src/topology/pcm.c | 11 +++++++--- src/topology/save.c | 34 ++++++++++++++++++++++++++----- src/topology/text.c | 2 +- src/topology/tplg_local.h | 2 +- 6 files changed, 64 insertions(+), 39 deletions(-) -- 2.7.4

2 14

Remove cset value quotation
by GitHub pull_request - opened 14 Jul '20

14 Jul '20

alsa-project/alsa-ucm-conf pull request #40 was opened from libinyang: This patchset removes the quotation marks of the cset values in the ucm2 config files. In ucm2 files, the cset values should not use quotation marks. Otherwise, the cset won't work. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/40 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/40.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] iec958 plugin enhancements for HDMI
by Matthias Reichl 14 Jul '20

14 Jul '20

Testing with vc4-hdmi / Raspberry Pi 4 showed that the iec958 plugin was missing two features required for HDMI compatibility: HD audio bitstreams like DTS HD MA or Dolby True HD are usually packed into 8-channel 192kHz 16-bit frames but are in fact a single IEC 61937 stream at 768kHz frame rate. So the data in the 8-channel frame has to be formatted as 4 contiguous IEC 60958 frames. If channel status bits weren't set, iec958 indicated a sample rate of 48kHz in the channel status bits which the HDMI analyzer complained about and caused issues with some HDMI devices if it didn't match the real rate. That was a long outstanding FIXME since the iec958 plugin was added. With this series HD audio passthrough and PCM audio works fine with the downstream 5.4 RPi kernel. Matthias Reichl (2): pcm: iec958: implement HDMI HBR audio formatting pcm: iec958: set channel status bits according to rate and format src/pcm/pcm_iec958.c | 117 ++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 109 insertions(+), 8 deletions(-) -- 2.20.1

2 3

Re: [PATCH] soc jack: When snd_soc_card_jack_new is not called or the call fails, calling this function causes a null pointer access
by Mark Brown 14 Jul '20

14 Jul '20

On Fri, Jul 10, 2020 at 02:56:03PM +0000, qiao mnlife wrote: > ________________________________ > From: mnlife <mnlife(a)foxmail.com> > Sent: Friday, July 10, 2020 10:45:42 PM > To: mnlife.qiao(a)gmail.com <mnlife.qiao(a)gmail.com> > Cc: mnlife <mnlife(a)foxmail.com> > Subject: [PATCH] When snd_soc_card_jack_new is not called or the call fails, calling this function causes a null pointer access > > Signed-off-by: mnlife <mnlife(a)foxmail.com> This looks good but it's a bit mangled in submission - can you please try to resend using something closer to the format in submitting-patches.rst? The changelog should be in the body of the mail and there should be a patch subject in a similar format to to the other patches in git.

1 0

[PATCH] ALSA: hda: Enable sync-write operation as default for all controllers
by Takashi Iwai 14 Jul '20

14 Jul '20

In the end we already enabled the sync-write mode for most of HD-audio controllers including Intel, and it's no big merit to keep the async write mode for the rest. Let's make it as default and drop the superfluous AZX_DCAPS_SYNC_WRITE bit flag. Also, avoid to set the allow_bus_reset flag, which is a quite unstable and hackish behavior that was needed only for some early platforms (decades ago). The straight fallback to the single cmd mode is more robust. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_controller.c | 11 ++--------- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 16 +++++++--------- 3 files changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 9765652a73d7..80016b7b6849 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1202,15 +1202,8 @@ int azx_bus_init(struct azx *chip, const char *model) if (chip->driver_caps & AZX_DCAPS_4K_BDLE_BOUNDARY) bus->core.align_bdle_4k = true; - /* AMD chipsets often cause the communication stalls upon certain - * sequence like the pin-detection. It seems that forcing the synced - * access works around the stall. Grrr... - */ - if (chip->driver_caps & AZX_DCAPS_SYNC_WRITE) { - dev_dbg(chip->card->dev, "Enable sync_write for stable communication\n"); - bus->core.sync_write = 1; - bus->allow_bus_reset = 1; - } + /* enable sync_write flag for stable communication as default */ + bus->core.sync_write = 1; return 0; } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 82e26442724b..fe171685492d 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -33,7 +33,7 @@ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ #define AZX_DCAPS_AMD_WORKAROUND (1 << 17) /* AMD-specific workaround */ #define AZX_DCAPS_NO_64BIT (1 << 18) /* No 64bit address */ -#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ +/* 19 unused */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ #define AZX_DCAPS_NO_ALIGN_BUFSIZE (1 << 21) /* no buffer size alignment */ /* 22 unused */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3565e2ab0965..b2bd21f5a52e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -283,13 +283,12 @@ enum { /* quirks for old Intel chipsets */ #define AZX_DCAPS_INTEL_ICH \ - (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE |\ - AZX_DCAPS_SYNC_WRITE) + (AZX_DCAPS_OLD_SSYNC | AZX_DCAPS_NO_ALIGN_BUFSIZE) /* quirks for Intel PCH */ #define AZX_DCAPS_INTEL_PCH_BASE \ (AZX_DCAPS_NO_ALIGN_BUFSIZE | AZX_DCAPS_COUNT_LPIB_DELAY |\ - AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) + AZX_DCAPS_SNOOP_TYPE(SCH)) /* PCH up to IVB; no runtime PM; bind with i915 gfx */ #define AZX_DCAPS_INTEL_PCH_NOPM \ @@ -304,13 +303,13 @@ enum { #define AZX_DCAPS_INTEL_HASWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_COUNT_LPIB_DELAY |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) + AZX_DCAPS_SNOOP_TYPE(SCH)) /* Broadwell HDMI can't use position buffer reliably, force to use LPIB */ #define AZX_DCAPS_INTEL_BROADWELL \ (/*AZX_DCAPS_ALIGN_BUFSIZE |*/ AZX_DCAPS_POSFIX_LPIB |\ AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_COMPONENT |\ - AZX_DCAPS_SNOOP_TYPE(SCH) | AZX_DCAPS_SYNC_WRITE) + AZX_DCAPS_SNOOP_TYPE(SCH)) #define AZX_DCAPS_INTEL_BAYTRAIL \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_I915_COMPONENT) @@ -321,19 +320,18 @@ enum { #define AZX_DCAPS_INTEL_SKYLAKE \ (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ - AZX_DCAPS_SYNC_WRITE |\ AZX_DCAPS_SEPARATE_STREAM_TAG | AZX_DCAPS_I915_COMPONENT) #define AZX_DCAPS_INTEL_BROXTON AZX_DCAPS_INTEL_SKYLAKE /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ - (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB |\ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB |\ AZX_DCAPS_SNOOP_TYPE(ATI)) /* quirks for ATI/AMD HDMI */ #define AZX_DCAPS_PRESET_ATI_HDMI \ - (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_POSFIX_LPIB|\ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_POSFIX_LPIB|\ AZX_DCAPS_NO_MSI64) /* quirks for ATI HDMI with snoop off */ @@ -342,7 +340,7 @@ enum { /* quirks for AMD SB */ #define AZX_DCAPS_PRESET_AMD_SB \ - (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_SYNC_WRITE | AZX_DCAPS_AMD_WORKAROUND |\ + (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_AMD_WORKAROUND |\ AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) /* quirks for Nvidia */ -- 2.25.0

2 6

[PATCH v2] dt-bindings: sound: convert rk3328 codec binding to yaml
by Katsuhiro Suzuki 14 Jul '20

14 Jul '20

This patch converts Rockchip rk3328 audio codec binding to DT schema. And adds description about "mclk" clock and fixes some errors in original example. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/rockchip,rk3328-codec.txt | 28 -------- .../bindings/sound/rockchip,rk3328-codec.yaml | 70 +++++++++++++++++++ 2 files changed, 70 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt deleted file mode 100644 index 1ecd75d2032a..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ /dev/null @@ -1,28 +0,0 @@ -* Rockchip Rk3328 internal codec - -Required properties: - -- compatible: "rockchip,rk3328-codec" -- reg: physical base address of the controller and length of memory mapped - region. -- rockchip,grf: the phandle of the syscon node for GRF register. -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should be "pclk". -- spk-depop-time-ms: speak depop time msec. - -Optional properties: - -- mute-gpios: GPIO specifier for external line driver control (typically the - dedicated GPIO_MUTE pin) - -Example for rk3328 internal codec: - -codec: codec@ff410000 { - compatible = "rockchip,rk3328-codec"; - reg = <0x0 0xff410000 0x0 0x1000>; - rockchip,grf = <&grf>; - clocks = <&cru PCLK_ACODEC>; - clock-names = "pclk"; - mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; - spk-depop-time-ms = 100; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml new file mode 100644 index 000000000000..525b48c2f5de --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml @@ -0,0 +1,70 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + const: rockchip,rk3328-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + - description: clock for I2S master clock + + clock-names: + items: + - const: pclk + - const: mclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + spk-depop-time-ms: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 200 + description: + Speaker depop time in msec. + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + - "#sound-dai-cells" + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/clock/rk3328-cru.h> + codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0xff410000 0x1000>; + clocks = <&cru PCLK_ACODECPHY>, <&cru SCLK_I2S1>; + clock-names = "pclk", "mclk"; + rockchip,grf = <&grf>; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; + spk-depop-time-ms = <100>; + #sound-dai-cells = <0>; + }; -- 2.27.0

2 2

[PATCH] ASoC: rockchip: spdif: Handle clk by pm runtime
by Sugar Zhang 13 Jul '20

13 Jul '20

This patch handle the clk by pm runtime mechanism to simplify the clk management. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- sound/soc/rockchip/rockchip_spdif.c | 59 +++++++++++-------------------------- 1 file changed, 17 insertions(+), 42 deletions(-) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 6635145..6748108 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -306,44 +306,22 @@ static int rk_spdif_probe(struct platform_device *pdev) return -ENOMEM; spdif->hclk = devm_clk_get(&pdev->dev, "hclk"); - if (IS_ERR(spdif->hclk)) { - dev_err(&pdev->dev, "Can't retrieve rk_spdif bus clock\n"); + if (IS_ERR(spdif->hclk)) return PTR_ERR(spdif->hclk); - } - ret = clk_prepare_enable(spdif->hclk); - if (ret) { - dev_err(spdif->dev, "hclock enable failed %d\n", ret); - return ret; - } spdif->mclk = devm_clk_get(&pdev->dev, "mclk"); - if (IS_ERR(spdif->mclk)) { - dev_err(&pdev->dev, "Can't retrieve rk_spdif master clock\n"); - ret = PTR_ERR(spdif->mclk); - goto err_disable_hclk; - } - - ret = clk_prepare_enable(spdif->mclk); - if (ret) { - dev_err(spdif->dev, "clock enable failed %d\n", ret); - goto err_disable_clocks; - } + if (IS_ERR(spdif->mclk)) + return PTR_ERR(spdif->mclk); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) { - ret = PTR_ERR(regs); - goto err_disable_clocks; - } + if (IS_ERR(regs)) + return PTR_ERR(regs); spdif->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "hclk", regs, &rk_spdif_regmap_config); - if (IS_ERR(spdif->regmap)) { - dev_err(&pdev->dev, - "Failed to initialise managed register map\n"); - ret = PTR_ERR(spdif->regmap); - goto err_disable_clocks; - } + if (IS_ERR(spdif->regmap)) + return PTR_ERR(spdif->regmap); spdif->playback_dma_data.addr = res->start + SPDIF_SMPDR; spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; @@ -352,47 +330,44 @@ static int rk_spdif_probe(struct platform_device *pdev) spdif->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, spdif); - pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_request_idle(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = rk_spdif_runtime_resume(&pdev->dev); + if (ret) + goto err_pm_runtime; + } ret = devm_snd_soc_register_component(&pdev->dev, &rk_spdif_component, &rk_spdif_dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI\n"); - goto err_pm_runtime; + goto err_pm_suspend; } ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - goto err_pm_runtime; + goto err_pm_suspend; } return 0; +err_pm_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + rk_spdif_runtime_suspend(&pdev->dev); err_pm_runtime: pm_runtime_disable(&pdev->dev); -err_disable_clocks: - clk_disable_unprepare(spdif->mclk); -err_disable_hclk: - clk_disable_unprepare(spdif->hclk); return ret; } static int rk_spdif_remove(struct platform_device *pdev) { - struct rk_spdif_dev *spdif = dev_get_drvdata(&pdev->dev); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) rk_spdif_runtime_suspend(&pdev->dev); - clk_disable_unprepare(spdif->mclk); - clk_disable_unprepare(spdif->hclk); - return 0; } -- 2.7.4

2 1

[PATCH v4] ALSA: line6: add hw monitor volume control for POD HD500
by Vasily Khoruzhick 13 Jul '20

13 Jul '20

Add hw monitor volume control for POD HD500. The same change may work for HD500X but I don't have it to test. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- v4: add NULL check for kmalloc-ed buffer v3: - use EXPORT_SYMBOL_GPL instead of EXPORT_SYMBOL - use GFP_KERNEL instead of GFP_ATOMIC for allocating a message - use lower case for float lookup table v2: clamp volume value to [0, ARRAY_SIZE() -1] in podhd_set_monitor_level() sound/usb/line6/driver.c | 3 +- sound/usb/line6/driver.h | 4 ++ sound/usb/line6/podhd.c | 127 ++++++++++++++++++++++++++++++++++++++- 3 files changed, 132 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 7629116f570e..3e07251b80e3 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -97,7 +97,7 @@ static void line6_stop_listen(struct usb_line6 *line6) /* Send raw message in pieces of wMaxPacketSize bytes. */ -static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, +int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, int size) { int i, done = 0; @@ -132,6 +132,7 @@ static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, return done; } +EXPORT_SYMBOL_GPL(line6_send_raw_message); /* Notification of completion of asynchronous request transmission. diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 1a4e3700c80c..62c686bed0ca 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -108,6 +108,8 @@ enum { LINE6_CAP_CONTROL_MIDI = 1 << 4, /* device provides low-level information */ LINE6_CAP_CONTROL_INFO = 1 << 5, + /* device provides hardware monitoring volume control */ + LINE6_CAP_HWMON_CTL = 1 << 6, }; /* @@ -185,6 +187,8 @@ extern int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, unsigned datalen); extern int line6_read_serial_number(struct usb_line6 *line6, u32 *serial_number); +extern int line6_send_raw_message(struct usb_line6 *line6, + const char *buffer, int size); extern int line6_send_raw_message_async(struct usb_line6 *line6, const char *buffer, int size); extern int line6_send_sysex_message(struct usb_line6 *line6, diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index e39dc85c355a..1557483ec657 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -11,6 +11,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> +#include <sound/control.h> #include <sound/pcm.h> #include "driver.h" @@ -37,6 +38,9 @@ struct usb_line6_podhd { /* Firmware version */ int firmware_version; + + /* Monitor level */ + int monitor_level; }; #define line6_to_podhd(x) container_of(x, struct usb_line6_podhd, line6) @@ -250,6 +254,118 @@ static void podhd_disconnect(struct usb_line6 *line6) } } +static const unsigned int float_zero_to_one_lookup[] = { +0x00000000, 0x3c23d70a, 0x3ca3d70a, 0x3cf5c28f, 0x3d23d70a, 0x3d4ccccd, +0x3d75c28f, 0x3d8f5c29, 0x3da3d70a, 0x3db851ec, 0x3dcccccd, 0x3de147ae, +0x3df5c28f, 0x3e051eb8, 0x3e0f5c29, 0x3e19999a, 0x3e23d70a, 0x3e2e147b, +0x3e3851ec, 0x3e428f5c, 0x3e4ccccd, 0x3e570a3d, 0x3e6147ae, 0x3e6b851f, +0x3e75c28f, 0x3e800000, 0x3e851eb8, 0x3e8a3d71, 0x3e8f5c29, 0x3e947ae1, +0x3e99999a, 0x3e9eb852, 0x3ea3d70a, 0x3ea8f5c3, 0x3eae147b, 0x3eb33333, +0x3eb851ec, 0x3ebd70a4, 0x3ec28f5c, 0x3ec7ae14, 0x3ecccccd, 0x3ed1eb85, +0x3ed70a3d, 0x3edc28f6, 0x3ee147ae, 0x3ee66666, 0x3eeb851f, 0x3ef0a3d7, +0x3ef5c28f, 0x3efae148, 0x3f000000, 0x3f028f5c, 0x3f051eb8, 0x3f07ae14, +0x3f0a3d71, 0x3f0ccccd, 0x3f0f5c29, 0x3f11eb85, 0x3f147ae1, 0x3f170a3d, +0x3f19999a, 0x3f1c28f6, 0x3f1eb852, 0x3f2147ae, 0x3f23d70a, 0x3f266666, +0x3f28f5c3, 0x3f2b851f, 0x3f2e147b, 0x3f30a3d7, 0x3f333333, 0x3f35c28f, +0x3f3851ec, 0x3f3ae148, 0x3f3d70a4, 0x3f400000, 0x3f428f5c, 0x3f451eb8, +0x3f47ae14, 0x3f4a3d71, 0x3f4ccccd, 0x3f4f5c29, 0x3f51eb85, 0x3f547ae1, +0x3f570a3d, 0x3f59999a, 0x3f5c28f6, 0x3f5eb852, 0x3f6147ae, 0x3f63d70a, +0x3f666666, 0x3f68f5c3, 0x3f6b851f, 0x3f6e147b, 0x3f70a3d7, 0x3f733333, +0x3f75c28f, 0x3f7851ec, 0x3f7ae148, 0x3f7d70a4, 0x3f800000 +}; + +static void podhd_set_monitor_level(struct usb_line6_podhd *podhd, int value) +{ + unsigned int fl; + static const unsigned char msg[16] = { + /* Chunk is 0xc bytes (without first word) */ + 0x0c, 0x00, + /* First chunk in the message */ + 0x01, 0x00, + /* Message size is 2 4-byte words */ + 0x02, 0x00, + /* Unknown */ + 0x04, 0x41, + /* Unknown */ + 0x04, 0x00, 0x13, 0x00, + /* Volume, LE float32, 0.0 - 1.0 */ + 0x00, 0x00, 0x00, 0x00 + }; + unsigned char *buf; + + buf = kmalloc(sizeof(msg), GFP_KERNEL); + if (!buf) + return; + + memcpy(buf, msg, sizeof(msg)); + + if (value < 0) + value = 0; + + if (value >= ARRAY_SIZE(float_zero_to_one_lookup)) + value = ARRAY_SIZE(float_zero_to_one_lookup) - 1; + + fl = float_zero_to_one_lookup[value]; + + buf[12] = (fl >> 0) & 0xff; + buf[13] = (fl >> 8) & 0xff; + buf[14] = (fl >> 16) & 0xff; + buf[15] = (fl >> 24) & 0xff; + + line6_send_raw_message(&podhd->line6, buf, sizeof(msg)); + kfree(buf); + + podhd->monitor_level = value; +} + +/* control info callback */ +static int snd_podhd_control_monitor_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 = 0; + uinfo->value.integer.max = 100; + uinfo->value.integer.step = 1; + return 0; +} + +/* control get callback */ +static int snd_podhd_control_monitor_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol); + struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6); + + ucontrol->value.integer.value[0] = podhd->monitor_level; + return 0; +} + +/* control put callback */ +static int snd_podhd_control_monitor_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol); + struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6); + + if (ucontrol->value.integer.value[0] == podhd->monitor_level) + return 0; + + podhd_set_monitor_level(podhd, ucontrol->value.integer.value[0]); + return 1; +} + +/* control definition */ +static const struct snd_kcontrol_new podhd_control_monitor = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Monitor Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_podhd_control_monitor_info, + .get = snd_podhd_control_monitor_get, + .put = snd_podhd_control_monitor_put +}; + /* Try to init POD HD device. */ @@ -298,6 +414,15 @@ static int podhd_init(struct usb_line6 *line6, return err; } + if (pod->line6.properties->capabilities & LINE6_CAP_HWMON_CTL) { + podhd_set_monitor_level(pod, 100); + err = snd_ctl_add(line6->card, + snd_ctl_new1(&podhd_control_monitor, + line6->line6pcm)); + if (err < 0) + return err; + } + if (!(pod->line6.properties->capabilities & LINE6_CAP_CONTROL_INFO)) { /* register USB audio system directly */ return snd_card_register(line6->card); @@ -354,7 +479,7 @@ static const struct line6_properties podhd_properties_table[] = { .id = "PODHD500", .name = "POD HD500", .capabilities = LINE6_CAP_PCM | LINE6_CAP_CONTROL - | LINE6_CAP_HWMON, + | LINE6_CAP_HWMON | LINE6_CAP_HWMON_CTL, .altsetting = 1, .ctrl_if = 1, .ep_ctrl_r = 0x81, -- 2.27.0

2 1

[PATCH] ALSA: echoaudio: re-enable IRQs on failure path
by Dan Carpenter 13 Jul '20

13 Jul '20

This should be spin_unlock_irq() instead of spin_lock(). Fixes: 6c3312544873 ("ALSA: echoaudio: Prevent races in calls to set_audio_format()") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/pci/echoaudio/echoaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index e81f42811f45..6aeb99aa2414 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -721,7 +721,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream) spin_lock_irq(&chip->lock); if (snd_BUG_ON(!is_pipe_allocated(chip, pipe_index))) { - spin_unlock(&chip->lock); + spin_unlock_irq(&chip->lock); return -EINVAL; } -- 2.27.0

2 1

[PATCH] ASoC: rt5682: Add fmw property to get name of mclk
by Akshu Agrawal 13 Jul '20

13 Jul '20

Non-dts based systems can use ACPI DSDT to pass on the mclk. Thus add fmw property mclk-name to get the name of the system clk and link it to rt5682 mclk. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682.c | 7 ++++++- 2 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index e1f790561ac1..8717c02f89b8 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -42,6 +42,7 @@ struct rt5682_platform_data { unsigned int dmic_delay; const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; + const char *mclk_name; }; #endif diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index de40b6cd16cf..8b223bd0929e 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2849,7 +2849,10 @@ static int rt5682_probe(struct snd_soc_component *component) } else { #ifdef CONFIG_COMMON_CLK /* Check if MCLK provided */ - rt5682->mclk = devm_clk_get(component->dev, "mclk"); + if (rt5682->pdata.mclk_name) + rt5682->mclk = clk_get(NULL, rt5682->pdata.mclk_name); + if (!rt5682->mclk) + rt5682->mclk = devm_clk_get(component->dev, "mclk"); if (IS_ERR(rt5682->mclk)) { if (PTR_ERR(rt5682->mclk) != -ENOENT) { ret = PTR_ERR(rt5682->mclk); @@ -2976,6 +2979,8 @@ int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) rt5682->pdata.dai_clk_names[RT5682_DAI_WCLK_IDX], rt5682->pdata.dai_clk_names[RT5682_DAI_BCLK_IDX]); + device_property_read_string(dev, "realtek,mclk-name", &rt5682->pdata.mclk_name); + return 0; } EXPORT_SYMBOL_GPL(rt5682_parse_dt); -- 2.20.1

3 3

[PATCH] ALSA: hda/realtek: Enable headset mic of Acer TravelMate B311R-31 with ALC256
by Jian-Hong Pan 13 Jul '20

13 Jul '20

The Acer TravelMate B311R-31 laptop's audio (1025:1430) with ALC256 cannot detect the headset microphone until ALC256_FIXUP_ACER_MIC_NO_PRESENCE quirk maps the NID 0x19 as the headset mic pin. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- Re-send for wrong email address sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 194ffa8c66ce..0585ac97faf4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,6 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, + ALC256_FIXUP_ACER_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7359,6 +7360,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_ACER_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a11120 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7387,6 +7397,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), -- 2.27.0

2 1

[PATCH v3] ALSA: line6: add hw monitor volume control for POD HD500
by Vasily Khoruzhick 13 Jul '20

13 Jul '20

Add hw monitor volume control for POD HD500. The same change may work for HD500X but I don't have it to test. Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- v3: - use EXPORT_SYMBOL_GPL instead of EXPORT_SYMBOL - use GFP_KERNEL instead of GFP_ATOMIC for allocating a message - use lower case for float lookup table v2: clamp volume value to [0, ARRAY_SIZE() -1] in podhd_set_monitor_level() sound/usb/line6/driver.c | 3 +- sound/usb/line6/driver.h | 4 ++ sound/usb/line6/podhd.c | 124 ++++++++++++++++++++++++++++++++++++++- 3 files changed, 129 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 7629116f570e..3e07251b80e3 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -97,7 +97,7 @@ static void line6_stop_listen(struct usb_line6 *line6) /* Send raw message in pieces of wMaxPacketSize bytes. */ -static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, +int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, int size) { int i, done = 0; @@ -132,6 +132,7 @@ static int line6_send_raw_message(struct usb_line6 *line6, const char *buffer, return done; } +EXPORT_SYMBOL_GPL(line6_send_raw_message); /* Notification of completion of asynchronous request transmission. diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 1a4e3700c80c..62c686bed0ca 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -108,6 +108,8 @@ enum { LINE6_CAP_CONTROL_MIDI = 1 << 4, /* device provides low-level information */ LINE6_CAP_CONTROL_INFO = 1 << 5, + /* device provides hardware monitoring volume control */ + LINE6_CAP_HWMON_CTL = 1 << 6, }; /* @@ -185,6 +187,8 @@ extern int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, unsigned datalen); extern int line6_read_serial_number(struct usb_line6 *line6, u32 *serial_number); +extern int line6_send_raw_message(struct usb_line6 *line6, + const char *buffer, int size); extern int line6_send_raw_message_async(struct usb_line6 *line6, const char *buffer, int size); extern int line6_send_sysex_message(struct usb_line6 *line6, diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index e39dc85c355a..0410d256aa50 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -11,6 +11,7 @@ #include <linux/slab.h> #include <linux/module.h> #include <sound/core.h> +#include <sound/control.h> #include <sound/pcm.h> #include "driver.h" @@ -37,6 +38,9 @@ struct usb_line6_podhd { /* Firmware version */ int firmware_version; + + /* Monitor level */ + int monitor_level; }; #define line6_to_podhd(x) container_of(x, struct usb_line6_podhd, line6) @@ -250,6 +254,115 @@ static void podhd_disconnect(struct usb_line6 *line6) } } +static const unsigned int float_zero_to_one_lookup[] = { +0x00000000, 0x3c23d70a, 0x3ca3d70a, 0x3cf5c28f, 0x3d23d70a, 0x3d4ccccd, +0x3d75c28f, 0x3d8f5c29, 0x3da3d70a, 0x3db851ec, 0x3dcccccd, 0x3de147ae, +0x3df5c28f, 0x3e051eb8, 0x3e0f5c29, 0x3e19999a, 0x3e23d70a, 0x3e2e147b, +0x3e3851ec, 0x3e428f5c, 0x3e4ccccd, 0x3e570a3d, 0x3e6147ae, 0x3e6b851f, +0x3e75c28f, 0x3e800000, 0x3e851eb8, 0x3e8a3d71, 0x3e8f5c29, 0x3e947ae1, +0x3e99999a, 0x3e9eb852, 0x3ea3d70a, 0x3ea8f5c3, 0x3eae147b, 0x3eb33333, +0x3eb851ec, 0x3ebd70a4, 0x3ec28f5c, 0x3ec7ae14, 0x3ecccccd, 0x3ed1eb85, +0x3ed70a3d, 0x3edc28f6, 0x3ee147ae, 0x3ee66666, 0x3eeb851f, 0x3ef0a3d7, +0x3ef5c28f, 0x3efae148, 0x3f000000, 0x3f028f5c, 0x3f051eb8, 0x3f07ae14, +0x3f0a3d71, 0x3f0ccccd, 0x3f0f5c29, 0x3f11eb85, 0x3f147ae1, 0x3f170a3d, +0x3f19999a, 0x3f1c28f6, 0x3f1eb852, 0x3f2147ae, 0x3f23d70a, 0x3f266666, +0x3f28f5c3, 0x3f2b851f, 0x3f2e147b, 0x3f30a3d7, 0x3f333333, 0x3f35c28f, +0x3f3851ec, 0x3f3ae148, 0x3f3d70a4, 0x3f400000, 0x3f428f5c, 0x3f451eb8, +0x3f47ae14, 0x3f4a3d71, 0x3f4ccccd, 0x3f4f5c29, 0x3f51eb85, 0x3f547ae1, +0x3f570a3d, 0x3f59999a, 0x3f5c28f6, 0x3f5eb852, 0x3f6147ae, 0x3f63d70a, +0x3f666666, 0x3f68f5c3, 0x3f6b851f, 0x3f6e147b, 0x3f70a3d7, 0x3f733333, +0x3f75c28f, 0x3f7851ec, 0x3f7ae148, 0x3f7d70a4, 0x3f800000 +}; + +static void podhd_set_monitor_level(struct usb_line6_podhd *podhd, int value) +{ + unsigned int fl; + static const unsigned char msg[16] = { + /* Chunk is 0xc bytes (without first word) */ + 0x0c, 0x00, + /* First chunk in the message */ + 0x01, 0x00, + /* Message size is 2 4-byte words */ + 0x02, 0x00, + /* Unknown */ + 0x04, 0x41, + /* Unknown */ + 0x04, 0x00, 0x13, 0x00, + /* Volume, LE float32, 0.0 - 1.0 */ + 0x00, 0x00, 0x00, 0x00 + }; + unsigned char *buf; + + buf = kmalloc(sizeof(msg), GFP_KERNEL); + memcpy(buf, msg, sizeof(msg)); + + if (value < 0) + value = 0; + + if (value >= ARRAY_SIZE(float_zero_to_one_lookup)) + value = ARRAY_SIZE(float_zero_to_one_lookup) - 1; + + fl = float_zero_to_one_lookup[value]; + + buf[12] = (fl >> 0) & 0xff; + buf[13] = (fl >> 8) & 0xff; + buf[14] = (fl >> 16) & 0xff; + buf[15] = (fl >> 24) & 0xff; + + line6_send_raw_message(&podhd->line6, buf, sizeof(msg)); + kfree(buf); + + podhd->monitor_level = value; +} + +/* control info callback */ +static int snd_podhd_control_monitor_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 = 0; + uinfo->value.integer.max = 100; + uinfo->value.integer.step = 1; + return 0; +} + +/* control get callback */ +static int snd_podhd_control_monitor_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol); + struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6); + + ucontrol->value.integer.value[0] = podhd->monitor_level; + return 0; +} + +/* control put callback */ +static int snd_podhd_control_monitor_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_line6_pcm *line6pcm = snd_kcontrol_chip(kcontrol); + struct usb_line6_podhd *podhd = line6_to_podhd(line6pcm->line6); + + if (ucontrol->value.integer.value[0] == podhd->monitor_level) + return 0; + + podhd_set_monitor_level(podhd, ucontrol->value.integer.value[0]); + return 1; +} + +/* control definition */ +static const struct snd_kcontrol_new podhd_control_monitor = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Monitor Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_podhd_control_monitor_info, + .get = snd_podhd_control_monitor_get, + .put = snd_podhd_control_monitor_put +}; + /* Try to init POD HD device. */ @@ -298,6 +411,15 @@ static int podhd_init(struct usb_line6 *line6, return err; } + if (pod->line6.properties->capabilities & LINE6_CAP_HWMON_CTL) { + podhd_set_monitor_level(pod, 100); + err = snd_ctl_add(line6->card, + snd_ctl_new1(&podhd_control_monitor, + line6->line6pcm)); + if (err < 0) + return err; + } + if (!(pod->line6.properties->capabilities & LINE6_CAP_CONTROL_INFO)) { /* register USB audio system directly */ return snd_card_register(line6->card); @@ -354,7 +476,7 @@ static const struct line6_properties podhd_properties_table[] = { .id = "PODHD500", .name = "POD HD500", .capabilities = LINE6_CAP_PCM | LINE6_CAP_CONTROL - | LINE6_CAP_HWMON, + | LINE6_CAP_HWMON | LINE6_CAP_HWMON_CTL, .altsetting = 1, .ctrl_if = 1, .ep_ctrl_r = 0x81, -- 2.27.0

2 1

[PATCH] ALSA: hda/realtek: Enable headset mic of Acer TravelMate B311R-31 with ALC256
by Jian-Hong Pan 13 Jul '20

13 Jul '20

The Acer TravelMate B311R-31 laptop's audio (1025:1430) with ALC256 cannot detect the headset microphone until ALC256_FIXUP_ACER_MIC_NO_PRESENCE quirk maps the NID 0x19 as the headset mic pin. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 194ffa8c66ce..0585ac97faf4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,6 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, + ALC256_FIXUP_ACER_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -7359,6 +7360,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC256_FIXUP_ACER_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a11120 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7387,6 +7397,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), -- 2.27.0

1 0

Beginning 5.8rc1 kernel USB headsets (ASUS ROG Delta and HyperX Cloud Orbit S) play sound as if in slow-motion.
by Mikhail Gavrilov 13 Jul '20

13 Jul '20

Beginning 5.8rc1 (git 69119673bd50) kernel USB headsets (ASUS ROG Delta and HyperX Cloud Orbit S) play sound as if in slow-motion. And in 5.8rc4 (git dcde237b9b0e) this still not fixed yet. The bisecting is problematic because rc1 also has another issue https://lkml.org/lkml/2020/6/22/21 which completely breaks the sound subsystem. If anyone said how to fix https://lkml.org/lkml/2020/6/22/21 I can bisect this issue. -- Best Regards, Mike Gavrilov.

2 2

alsactl do not save all the parameters
by GitHub issues - opened 12 Jul '20

12 Jul '20

alsa-project/alsa-utils issue #46 was opened from alemirandap: Hello everyone, my laptop need to mute the speaker for the full sound experience, when I configure alsamixer with all the levels and mute the speaker I execute "sudo alsactl store" but after rebooted all the levels and configuration are saved except for muting the speaker. I made some tests muting the master or the headphones and the executed "sudo alsactl store" and these are saved. I have to manually in every boot mute the speaker, I 'm under arch linux, with others distros doesn't happen, can you give me a hand with this?, thank you [alsa-info.txt.](https://github.com/alsa-project/alsa-utils/files/4908789/al… Issue URL : https://github.com/alsa-project/alsa-utils/issues/46 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: trident: Fix a memory leak in snd_trident_create
by Navid Emamdoost 11 Jul '20

11 Jul '20

In the implementation of snd_trident_create(), the allocated trident is leaked if snd_trident_mixer() fails. Release via snd_trident_free(). Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/trident/trident_main.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index 6e50376163a2..e98c692f6aa9 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -3582,8 +3582,11 @@ int snd_trident_create(struct snd_card *card, return err; } - if ((err = snd_trident_mixer(trident, pcm_spdif_device)) < 0) + err = snd_trident_mixer(trident, pcm_spdif_device); + if (err < 0) { + snd_trident_free(trident); return err; + } /* initialise synth voices */ for (i = 0; i < 64; i++) { -- 2.17.1

2 2

[PATCH v3] ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289
by Armas Spann 11 Jul '20

11 Jul '20

Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- Description: This patch adds support for headset mic to the ASUS ROG Zephyrus G14(GA401) notebook series by adding the corresponding vendor/pci_device id, as well as adding a new fixup for the used realtek ALC289. The fixup stets the correct pin to get the headset mic correctly recognized on audio-jack. Changelog: v1: initial v2: added missing vendor and device(pci) id v3: added description sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 194ffa8c66ce..ff4f3eeedfd0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,6 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, + ALC289_FIXUP_ASUS_G401, }; static const struct hda_fixup alc269_fixups[] = { @@ -7359,6 +7360,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC289_FIXUP_ASUS_G401] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7539,6 +7547,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_G401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.27.0

2 2

[PATCH v2] ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289
by Armas Spann 11 Jul '20

11 Jul '20

Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 194ffa8c66ce..ff4f3eeedfd0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,6 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, + ALC289_FIXUP_ASUS_G401, }; static const struct hda_fixup alc269_fixups[] = { @@ -7359,6 +7360,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC289_FIXUP_ASUS_G401] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7539,6 +7547,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_G401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.27.0

2 4

[PATCH v1] ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289
by Armas Spann 11 Jul '20

11 Jul '20

Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289 sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 194ffa8c66ce..2e7d4ebf8ce7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,6 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, + ALC289_FIXUP_ASUS_G401, }; static const struct hda_fixup alc269_fixups[] = { @@ -7359,6 +7360,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, + [ALC289_FIXUP_ASUS_G401] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { -- 2.27.0

1 0

Change model for ASUS platform
by Kailang 11 Jul '20

11 Jul '20

Hi Takashi, ASUS platform couldn't need to use Headset Mode model. Attach patch will change it. BR, Kailang

2 1

[PATCH] ALSA: usb-audio: Fix race against the error recovery URB submission
by Takashi Iwai 10 Jul '20

10 Jul '20

USB MIDI driver has an error recovery mechanism to resubmit the URB in the delayed timer handler, and this may race with the standard start / stop operations. Although both start and stop operations themselves don't race with each other due to the umidi->mutex protection, but this isn't applied to the timer handler. For fixing this potential race, the following changes are applied: - Since the timer handler can't use the mutex, we apply the umidi->disc_lock protection at each input stream URB submission; this also needs to change the GFP flag to GFP_ATOMIC - Add a check of the URB refcount and skip if already submitted - Move the timer cancel call at disconnection to the beginning of the procedure; this assures the in-flight timer handler is gone properly before killing all pending URBs Reported-by: syzbot+0f4ecfe6a2c322c81728(a)syzkaller.appspotmail.com Reported-by: syzbot+5f1d24c49c1d2c427497(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 047b90595d65..354f57692938 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1499,6 +1499,8 @@ void snd_usbmidi_disconnect(struct list_head *p) spin_unlock_irq(&umidi->disc_lock); up_write(&umidi->disc_rwsem); + del_timer_sync(&umidi->error_timer); + for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i]; if (ep->out) @@ -1525,7 +1527,6 @@ void snd_usbmidi_disconnect(struct list_head *p) ep->in = NULL; } } - del_timer_sync(&umidi->error_timer); } EXPORT_SYMBOL(snd_usbmidi_disconnect); @@ -2301,16 +2302,22 @@ void snd_usbmidi_input_stop(struct list_head *p) } EXPORT_SYMBOL(snd_usbmidi_input_stop); -static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint *ep) +static void snd_usbmidi_input_start_ep(struct snd_usb_midi *umidi, + struct snd_usb_midi_in_endpoint *ep) { unsigned int i; + unsigned long flags; if (!ep) return; for (i = 0; i < INPUT_URBS; ++i) { struct urb *urb = ep->urbs[i]; - urb->dev = ep->umidi->dev; - snd_usbmidi_submit_urb(urb, GFP_KERNEL); + spin_lock_irqsave(&umidi->disc_lock, flags); + if (!atomic_read(&urb->use_count)) { + urb->dev = ep->umidi->dev; + snd_usbmidi_submit_urb(urb, GFP_ATOMIC); + } + spin_unlock_irqrestore(&umidi->disc_lock, flags); } } @@ -2326,7 +2333,7 @@ void snd_usbmidi_input_start(struct list_head *p) if (umidi->input_running || !umidi->opened[1]) return; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) - snd_usbmidi_input_start_ep(umidi->endpoints[i].in); + snd_usbmidi_input_start_ep(umidi, umidi->endpoints[i].in); umidi->input_running = 1; } EXPORT_SYMBOL(snd_usbmidi_input_start); -- 2.16.4

1 0

[PATCH 0/6] ASoC: mediatek: mt8183: support DP audio
by Tzung-Bi Shih 10 Jul '20

10 Jul '20

This series is a follow up for a long time ago series (https://patchwork.kernel.org/cover/11204303/). The old series bound too much on the patches of DRM bridge and ASoC machine driver. And unluckily, the dependencies (https://lore.kernel.org/patchwork/patch/1126819/) have not applied. Revewing the ASoC patches in the old series, I found that they could be decoupled from the DRM bridge patches. And they are harmless as it is an optional attribute ("hdmi-codec") in DTS. This series arranges and rebases the harmless ASoC patches for mt8183-mt6358-ts3a227-max98357 and mt8183-da7219-max98357. The 1st and 4th patch add an optional DT property. The 1st patch was acked long time ago (https://patchwork.kernel.org/patch/11204321/). The 2nd and 5th patch add DAI link for using hdmi-codec. The 3rd and 6th patch support the HDMI jack reporting. Tzung-Bi Shih (6): ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183: use hdmi-codec ASoC: mediatek: mt8183: support HDMI jack reporting ASoC: dt-bindings: mt8183-da7219: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183-da7219: use hdmi-codec ASoC: mediatek: mt8183-da7219: support HDMI jack reporting .../bindings/sound/mt8183-da7219-max98357.txt | 4 +++ .../sound/mt8183-mt6358-ts3a227-max98357.txt | 2 ++ sound/soc/mediatek/Kconfig | 2 ++ .../mediatek/mt8183/mt8183-da7219-max98357.c | 29 +++++++++++++++++-- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 29 +++++++++++++++++-- 5 files changed, 60 insertions(+), 6 deletions(-) -- 2.27.0.383.g050319c2ae-goog

2 7

[PATCH 0/2] ASoC: fsl_spdif: Clear the validity bit for TX
by Shengjiu Wang 10 Jul '20

10 Jul '20

Clear the validity bit for TX Add kctl for configuring TX validity bit Shengjiu Wang (2): ASoC: fsl_spdif: Clear the validity bit for TX ASoC: fsl_spdif: Add kctl for configuring TX validity bit sound/soc/fsl/fsl_spdif.c | 51 ++++++++++++++++++++++++++++++++++++--- 1 file changed, 47 insertions(+), 4 deletions(-) -- 2.21.0

2 3

[PATCH] ASoC: Intel: cht_bsw_rt5672: Improve dai-set-fmt comment in cht_codec_fixup()
by Hans de Goede 10 Jul '20

10 Jul '20

As Pierre-Louis Bossart pointed out, saying that the default mode for the SSP is TDM 4 slot is not entirely accurate. There really are 2 default modes: The default mode for the SSP configuration is TDM 4 slot for the cpu-dai (hard-coded in DSP firmware), The default mode for the SSP configuration is I2S for the codec-dai (hard-coded in the 'SSP2-Codec" .dai_fmt masks, so far unused). This commit updates the comment in cht_codec_fixup() to properly reflect this. Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_rt5672.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 1470c3de7895..1812b786d33b 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -253,13 +253,17 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); /* - * Default mode for SSP configuration is TDM 4 slot. One board/design, - * the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The - * second piggy-backed, output-only codec is inside the keyboard-dock - * (which has extra speakers). Unlike the main rt5672 codec, we cannot - * configure this codec, it is hard coded to use 2 channel 24 bit I2S. - * Since we only support 2 channels anyways, there is no need for TDM - * on any cht-bsw-rt5672 designs. So we simply use I2S 2ch everywhere. + * The default mode for the cpu-dai is TDM 4 slot. The default mode + * for the codec-dai is I2S. So we need to either set the cpu-dai to + * I2S mode to match the codec-dai, or set the codec-dai to TDM 4 slot + * (or program both to yet another mode). + * One board, the Lenovo Miix 2 10, uses not 1 but 2 codecs connected + * to SSP2. The second piggy-backed, output-only codec is inside the + * keyboard-dock (which has extra speakers). Unlike the main rt5672 + * codec, we cannot configure this codec, it is hard coded to use + * 2 channel 24 bit I2S. For this to work we must use I2S mode on this + * board. Since we only support 2 channels anyways, there is no need + * for TDM on any cht-bsw-rt5672 designs. So we use I2S 2ch everywhere. */ ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), SND_SOC_DAIFMT_I2S | -- 2.26.2

2 1

[PATCH v4 00/10] ASoC: Clean-up W=1 build warnings - part2
by Pierre-Louis Bossart 10 Jul '20

10 Jul '20

Both Lee Jones and I submitted separate series, this is the second part of the merged result, for which no feedback was provided. I picked Lee's patches for rt5659 and ak4458 and added the pxa and ux500 that I didn't fix. The rest is largely identical between our respective series, with the exception of the sunxi which I documented and Lee removed. I don't have any specific preference and will go with the flow on this. Changes since v3: Improved commit subjects from 'fix kernel-doc' as suggested by Lee Jones. In a couple of cases I just reverted to Lee's patches when the code was identical. Added a couple of CC: tags from Lee's patches. Added Arnaud Pouliquen's Acked-by tag in first patch. Lee Jones (6): ASoC: sunxi: sun4i-spdif: Fix misspelling of 'reg_dac_txdata' in kernel-doc ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and 'reg_val_TSTDR' ASoC: codecs: rt5659: Remove many unused const variables ASoC: codecs: tlv320aic26: Demote seemingly unintentional kerneldoc header ASoC: codecs: ak4458: Remove set but never checked variable 'ret' Pierre-Louis Bossart (4): ASoC: sti: uniperif: fix 'defined by not used' warning ASoC: qcom: qdsp6: q6asm: Provide documentation for 'codec_profile' ASoC: sunxi: sun4i-i2s: add missing clock and format arguments in kernel-doc ASoC: codecs: rt5631: fix spurious kernel-doc start and missing arguments sound/soc/codecs/ak4458.c | 6 +++--- sound/soc/codecs/rt5631.c | 8 +++++-- sound/soc/codecs/rt5659.c | 37 --------------------------------- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/qcom/qdsp6/q6asm.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/sunxi/sun4i-i2s.c | 10 ++++++++- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 8 +++---- 10 files changed, 27 insertions(+), 52 deletions(-) base-commit: 6940701c715e193282cf8b31f970b2ea6eb26341 -- 2.25.1

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[PATCH v3 00/10] ASoC: Clean-up W=1 build warnings - part2
by Pierre-Louis Bossart 10 Jul '20

10 Jul '20

Both Lee Jones and I submitted separate series, this is the second part of the merged result, for which no feedback was provided. I picked Lee's patches for rt5659 and ak4458 and added the pxa and ux500 that I didn't fix. The rest is largely identical between our respective series, with the exception of the sunxi which I documented and Lee removed. I don't have any specific preference and will go with the flow on this. Lee Jones (4): ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and 'reg_val_TSTDR' ASoC: codecs: rt5659: Remove many unused const variables ASoC: codecs: ak4458: Remove set but never checked variable 'ret' Pierre-Louis Bossart (6): ASoC: qcom: q6asm: fix kernel-doc ASoC: sunxi: sun4i-i2s: fix kernel-doc ASoC: sunxi: sun4i-spdif: fix kernel-doc ASoC: sti: uniperif: fix 'defined by not used' warning ASoC: codecs: rt5631: fix kernel-doc ASoC: codecs: tlv320aic26: fix kernel-doc warning sound/soc/codecs/ak4458.c | 6 +++--- sound/soc/codecs/rt5631.c | 8 +++++-- sound/soc/codecs/rt5659.c | 37 --------------------------------- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/qcom/qdsp6/q6asm.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/sunxi/sun4i-i2s.c | 10 ++++++++- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 8 +++---- 10 files changed, 27 insertions(+), 52 deletions(-) -- 2.25.1

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Weird jack detection on ASRock B450M Pro4
by Arun Raghavan 10 Jul '20

10 Jul '20

Hi folks, I ran into some weird behaviour on this motherboard. First, alsa-info is here: http://alsa-project.org/db/?f=f48105f391d848b34b3779f27369130ca024f417 The weird behaviour is that the HDA card is picked up fine, but jack detection is broken on boot. However, this gets "fixed" after you run alsa-info. The before/after codec info is at: https://arunraghavan.net/temp/alsa-codecs-before.txt https://arunraghavan.net/temp/alsa-codecs-after.txt The key delta is: ---- @@ -133,24 +133,24 @@ Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC892 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 - Amp-Out vals: [0x34 0x34] - Converter: stream=0, channel=0 + Amp-Out vals: [0x29 0x29] + Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 - Amp-Out vals: [0x34 0x34] - Converter: stream=0, channel=0 + Amp-Out vals: [0x29 0x29] + Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS @@ -159,12 +159,12 @@ Control: name="Center Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=2, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 - Amp-Out vals: [0x34 0x34] - Converter: stream=0, channel=0 + Amp-Out vals: [0x29 0x29] + Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS @@ -172,11 +172,11 @@ Node 0x05 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] - Converter: stream=0, channel=0 + Converter: stream=5, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS ---- Any thoughts on what's changing that, and how this should be fixed? Cheers, Arun

1 1

ASoC: rt5670/cht_bsw_rt5672: Some bugfixes and cleanups
by Hans de Goede 10 Jul '20

10 Jul '20

Hi All, This series is mainly the result of me working on fixing the extra set of speakers in the Lenovo Miix 2 10 kbd dock not working with Linux (patches 1 and 3) while working on this I noticed some other bugs (patches 2 and 4) and I also ended up doing some generic cleanups (patches 5 and 6). For patches 1-3 or 1-4 it would be nice it they can go to a future 5.8-rc# 5-6 are mostly -next material. Regards, Hans

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Re: KASAN: use-after-free Read in line6_submit_audio_in_all_urbs
by Alan Stern 10 Jul '20

10 Jul '20

On Fri, Jun 26, 2020 at 10:18:12AM -0700, syzbot wrote: > Hello, > > syzbot found the following crash on: > > HEAD commit: fb574682 usbip: tools: fix module name in man page > git tree: https://git.kernel.org/pub/scm/linux/kernel/git/gregkh/usb.git usb-testing > console output: https://syzkaller.appspot.com/x/log.txt?x=156560b1100000 > kernel config: https://syzkaller.appspot.com/x/.config?x=63b40b2ae167bad6 > dashboard link: https://syzkaller.appspot.com/bug?extid=145012a46658ac00fc9e > compiler: gcc (GCC) 10.1.0-syz 20200507 > > Unfortunately, I don't have any reproducer for this crash yet. > > IMPORTANT: if you fix the bug, please add the following tag to the commit: > Reported-by: syzbot+145012a46658ac00fc9e(a)syzkaller.appspotmail.com > > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > snd_usb_toneport 5-1:0.0: URB in #0 submission failed (-19) > ================================================================== > BUG: KASAN: use-after-free in line6_submit_audio_in_all_urbs+0x10b/0x120 sound/usb/line6/capture.c:72 > Read of size 8 at addr ffff8881cffb1800 by task kworker/1:5/3257 > > CPU: 1 PID: 3257 Comm: kworker/1:5 Not tainted 5.8.0-rc1-syzkaller #0 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 > Workqueue: events line6_startup_work > Call Trace: > __dump_stack lib/dump_stack.c:77 [inline] > dump_stack+0xf6/0x16e lib/dump_stack.c:118 > print_address_description.constprop.0+0x1a/0x210 mm/kasan/report.c:383 > __kasan_report mm/kasan/report.c:513 [inline] > kasan_report.cold+0x37/0x7c mm/kasan/report.c:530 > line6_submit_audio_in_all_urbs+0x10b/0x120 sound/usb/line6/capture.c:72 > line6_stream_start+0x207/0x230 sound/usb/line6/pcm.c:197 > line6_pcm_acquire+0x161/0x210 sound/usb/line6/pcm.c:318 > line6_startup_work+0x42/0x50 sound/usb/line6/driver.c:734 > process_one_work+0x94c/0x15f0 kernel/workqueue.c:2269 > worker_thread+0x64c/0x1120 kernel/workqueue.c:2415 > kthread+0x392/0x470 kernel/kthread.c:291 > ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:293 > > Allocated by task 76: > save_stack+0x1b/0x40 mm/kasan/common.c:48 > set_track mm/kasan/common.c:56 [inline] > __kasan_kmalloc.constprop.0+0xc2/0xd0 mm/kasan/common.c:494 > kmalloc include/linux/slab.h:555 [inline] > kzalloc include/linux/slab.h:669 [inline] > line6_init_pcm+0x2a7/0x9e0 sound/usb/line6/pcm.c:533 > toneport_init+0xdd/0x6b0 sound/usb/line6/toneport.c:419 > line6_probe+0xaa0/0x1330 sound/usb/line6/driver.c:809 It look like the cancel_delayed_work() in line6_disconnect() needs to be cancel_delayed_work_sync(). Unfortunately we can't test this until syzbot is able to reproduce the bug. Alan Stern

2 1

Re: 回复: WARNING in submit_audio_out_urb/usb_submit_urb
by Takashi Iwai 10 Jul '20

10 Jul '20

On Fri, 10 Jul 2020 10:12:43 +0200, Zhang, Qiang wrote: > > > > ________________________________________ > 发件人: linux-kernel-owner(a)vger.kernel.org <linux-kernel-owner(a)vger.kernel.org> 代表 syzbot <syzbot+c190f6858a04ea7fbc52(a)syzkaller.appspotmail.com> > 发送时间: 2020年7月9日 21:34 > 收件人: andreyknvl(a)google.com; gregkh(a)linuxfoundation.org; ingrassia(a)epigenesys.com; linux-kernel(a)vger.kernel.org; linux-usb(a)vger.kernel.org; syzkaller-bugs(a)googlegroups.com > 主题: WARNING in submit_audio_out_urb/usb_submit_urb > > Hello, > > syzbot found the following crash on: > > HEAD commit: 768a0741 usb: dwc2: gadget: Remove assigned but never used.. > git tree: https://git.kernel.org/pub/scm/linux/kernel/git/gregkh/usb.git usb-testing > console output: https://syzkaller.appspot.com/x/log.txt?x=1568d11f100000 > kernel config: https://syzkaller.appspot.com/x/.config?x=999be4eb2478ffa5 > dashboard link: https://syzkaller.appspot.com/bug?extid=c190f6858a04ea7fbc52 > compiler: gcc (GCC) 10.1.0-syz 20200507 > syz repro: https://syzkaller.appspot.com/x/repro.syz?x=123aa2fb100000 > > IMPORTANT: if you fix the bug, please add the following tag to the commit: > Reported-by: syzbot+c190f6858a04ea7fbc52(a)syzkaller.appspotmail.com > > usb 1-1: send failed (error -32) > snd_usb_toneport 1-1:0.0: Line 6 TonePort GX now attached > ------------[ cut here ]------------ > usb 1-1: BOGUS urb xfer, pipe 0 != type 3 > WARNING: CPU: 0 PID: 12 at drivers/usb/core/urb.c:478 usb_submit_urb+0xa17/0x13e0 drivers/usb/core/urb.c:478 > Kernel panic - not syncing: panic_on_warn set ... > CPU: 0 PID: 12 Comm: kworker/0:1 Not tainted 5.8.0-rc3-syzkaller #0 > Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 > Workqueue: events line6_startup_work > Call Trace: > __dump_stack lib/dump_stack.c:77 [inline] > dump_stack+0xf6/0x16e lib/dump_stack.c:118 > panic+0x2aa/0x6e1 kernel/panic.c:231 > __warn.cold+0x20/0x50 kernel/panic.c:600 > report_bug+0x1bd/0x210 lib/bug.c:198 > handle_bug+0x41/0x80 arch/x86/kernel/traps.c:235 > exc_invalid_op+0x13/0x40 arch/x86/kernel/traps.c:255 > asm_exc_invalid_op+0x12/0x20 arch/x86/include/asm/idtentry.h:563 > RIP: 0010:usb_submit_urb+0xa17/0x13e0 drivers/usb/core/urb.c:478 > Code: 84 e7 04 00 00 e8 a9 10 ca fd 4c 89 ef e8 41 79 12 ff 41 89 d8 44 89 e1 4c 89 f2 48 89 c6 48 c7 c7 80 a0 5d 86 e8 db 77 9e fd <0f> 0b e8 82 10 ca fd 0f b6 6c 24 08 48 c7 c6 e0 a1 5d 86 48 89 ef > RSP: 0018:ffff8881da227b10 EFLAGS: 00010086 > RAX: 0000000000000000 RBX: 0000000000000003 RCX: 0000000000000000 > RDX: ffff8881da211900 RSI: ffffffff8129b4e3 RDI: ffffed103b444f54 > RBP: 0000000000000030 R08: 0000000000000001 R09: ffff8881db21fe8b > R10: 0000000000000000 R11: 0000000000000004 R12: 0000000000000000 > R13: ffff8881d6ecd0a0 R14: ffff8881d3d8c690 R15: ffff8881d54c4000 > submit_audio_out_urb+0x6d6/0x1a00 sound/usb/line6/playback.c:271 > line6_submit_audio_out_all_urbs+0xc9/0x120 sound/usb/line6/playback.c:291 > line6_stream_start+0x187/0x230 sound/usb/line6/pcm.c:195 > line6_pcm_acquire+0x137/0x210 sound/usb/line6/pcm.c:318 > line6_startup_work+0x42/0x50 sound/usb/line6/driver.c:734 > process_one_work+0x94c/0x15f0 kernel/workqueue.c:2269 > worker_thread+0x64c/0x1120 kernel/workqueue.c:2415 > kthread+0x392/0x470 kernel/kthread.c:291 > ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:293 > > It's like Alan Stern's reply to the email titled "KASAN: use-after-free Read in line6_submit_audio_in_all_urbs." It's also like a problem with asynchronous operations. can replace "cancel_delayed_work" with" cancel_delayed_work_sync" > in "line6_disconnect" func？ No, this looks a different problem. It's just a matter of the missing URB type check that leads to a USB warning. A typical issue with the USB fuzzing. I'll check it later. thanks, Takashi

1 1

Feature request: volume parameter for aplay
by GitHub issues - opened 10 Jul '20

10 Jul '20

alsa-project/alsa-utils issue #45 was opened from flatsiedatsie: I am currently in a situation where I'm trying to play a notification sound when a privacy friendly voice control system hears its wake word. Specifically, it's the [Voco addon](https://github.com/createcandle/voco) for the [Mozilla WebThings Gateway](https://webthingsgateway.com/), an open source smart home controller. Up till now I used `aplay` for this. But I need/want to support a ReSpeaker USB microphone, and that device doesn't have a simple volume control. Unfortunately addons are not allowed to make changes to asound.conf (which could add softvol) or install something like Pulse audio to mitigate this situation, as those changes could affect other addons of the system. My current solution has been to switch to a different available audio player, `ffplay`. It has a `-volume` parameter. Having a `-volume` option actually works out quite nicely, as it allows sounds to be played at relative volume to each other. For example, I can play a music stream at 40% volume, while setting the audio feedback from the voice control to be louder so it can always be heard over the music. The downside of ffplay is that it's slow to start up. It can take up to a second or two. Aplay's responses were pretty much instant, which I now realise was really valuable and important. I'd prefer to switch back to aplay, but I'd need to somehow get aplay to output audio at a relative volume. Is this possible somehow? And if not, would it be an idea to add this feature to aplay? Issue URL : https://github.com/alsa-project/alsa-utils/issues/45 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH AUTOSEL 5.7 15/53] ASoC: SOF: Intel: add PCI IDs for ICL-H and TGL-H
by Sasha Levin 10 Jul '20

10 Jul '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit c8d2e2bfaeffa0f914330e8b4e45b986c8d30b58 ] Usually the DSP is not traditionally enabled on H skews but this might be used moving forward. 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> Link: https://lore.kernel.org/r/20200617164755.18104-4-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/sof-pci-dev.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 7b1846aeadd59..3a71f813fb563 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -411,8 +411,11 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&cfl_desc}, #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) - { PCI_DEVICE(0x8086, 0x34C8), + { PCI_DEVICE(0x8086, 0x34C8), /* ICL-LP */ .driver_data = (unsigned long)&icl_desc}, + { PCI_DEVICE(0x8086, 0x3dc8), /* ICL-H */ + .driver_data = (unsigned long)&icl_desc}, + #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_JASPERLAKE) { PCI_DEVICE(0x8086, 0x38c8), @@ -431,8 +434,11 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&cml_desc}, #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) - { PCI_DEVICE(0x8086, 0xa0c8), + { PCI_DEVICE(0x8086, 0xa0c8), /* TGL-LP */ .driver_data = (unsigned long)&tgl_desc}, + { PCI_DEVICE(0x8086, 0x43c8), /* TGL-H */ + .driver_data = (unsigned long)&tgl_desc}, + #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) { PCI_DEVICE(0x8086, 0x4b55), -- 2.25.1

3 4

[PATCH] ASoC: atmel-pdmic: remove codec component
by Codrin Ciubotariu 10 Jul '20

10 Jul '20

The CPU and the codec both are represented now as components, so for PDMIC we are registering two componenets with the same name. Since there is no actual codec, we will merge the codec component into the CPU one and use a dummy codec instead, for the DAI link. As a bonus, debugfs will no longer report an error when will try to create entries for both componenets with the same name. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/atmel-pdmic.c | 110 +++++++++++++--------------------- 1 file changed, 41 insertions(+), 69 deletions(-) diff --git a/sound/soc/atmel/atmel-pdmic.c b/sound/soc/atmel/atmel-pdmic.c index 5245826cd99d..c2b639928c69 100644 --- a/sound/soc/atmel/atmel-pdmic.c +++ b/sound/soc/atmel/atmel-pdmic.c @@ -147,32 +147,26 @@ static int atmel_pdmic_cpu_dai_prepare(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = cpu_dai->component; u32 val; + int ret; /* Clean the PDMIC Converted Data Register */ - return regmap_read(dd->regmap, PDMIC_CDR, &val); -} - -static const struct snd_soc_dai_ops atmel_pdmic_cpu_dai_ops = { - .startup = atmel_pdmic_cpu_dai_startup, - .shutdown = atmel_pdmic_cpu_dai_shutdown, - .prepare = atmel_pdmic_cpu_dai_prepare, -}; + ret = regmap_read(dd->regmap, PDMIC_CDR, &val); + if (ret < 0) + return 0; -#define ATMEL_PDMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + ret = snd_soc_component_update_bits(component, PDMIC_CR, + PDMIC_CR_ENPDM_MASK, + PDMIC_CR_ENPDM_DIS << + PDMIC_CR_ENPDM_SHIFT); + if (ret < 0) + return ret; -static struct snd_soc_dai_driver atmel_pdmic_cpu_dai = { - .capture = { - .channels_min = 1, - .channels_max = 1, - .rates = SNDRV_PCM_RATE_KNOT, - .formats = ATMEL_PDMIC_FORMATS,}, - .ops = &atmel_pdmic_cpu_dai_ops, -}; + return 0; +} -static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = { - .name = "atmel-pdmic", -}; +#define ATMEL_PDMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) /* platform */ #define ATMEL_PDMIC_MAX_BUF_SIZE (64 * 1024) @@ -355,27 +349,16 @@ static int atmel_pdmic_component_probe(struct snd_soc_component *component) return 0; } -static struct snd_soc_component_driver soc_component_dev_pdmic = { - .probe = atmel_pdmic_component_probe, - .controls = atmel_pdmic_snd_controls, - .num_controls = ARRAY_SIZE(atmel_pdmic_snd_controls), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -/* codec dai component */ #define PDMIC_MR_PRESCAL_MAX_VAL 127 static int -atmel_pdmic_codec_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *codec_dai) +atmel_pdmic_cpu_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_pdmic *dd = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; unsigned int rate_min = substream->runtime->hw.rate_min; unsigned int rate_max = substream->runtime->hw.rate_max; int fs = params_rate(params); @@ -445,21 +428,10 @@ atmel_pdmic_codec_dai_hw_params(struct snd_pcm_substream *substream, return 0; } -static int atmel_pdmic_codec_dai_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) -{ - struct snd_soc_component *component = codec_dai->component; - - snd_soc_component_update_bits(component, PDMIC_CR, PDMIC_CR_ENPDM_MASK, - PDMIC_CR_ENPDM_DIS << PDMIC_CR_ENPDM_SHIFT); - - return 0; -} - -static int atmel_pdmic_codec_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) +static int atmel_pdmic_cpu_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 val; switch (cmd) { @@ -482,16 +454,16 @@ static int atmel_pdmic_codec_dai_trigger(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops atmel_pdmic_codec_dai_ops = { - .hw_params = atmel_pdmic_codec_dai_hw_params, - .prepare = atmel_pdmic_codec_dai_prepare, - .trigger = atmel_pdmic_codec_dai_trigger, +static const struct snd_soc_dai_ops atmel_pdmic_cpu_dai_ops = { + .startup = atmel_pdmic_cpu_dai_startup, + .shutdown = atmel_pdmic_cpu_dai_shutdown, + .prepare = atmel_pdmic_cpu_dai_prepare, + .hw_params = atmel_pdmic_cpu_dai_hw_params, + .trigger = atmel_pdmic_cpu_dai_trigger, }; -#define ATMEL_PDMIC_CODEC_DAI_NAME "atmel-pdmic-hifi" -static struct snd_soc_dai_driver atmel_pdmic_codec_dai = { - .name = ATMEL_PDMIC_CODEC_DAI_NAME, +static struct snd_soc_dai_driver atmel_pdmic_cpu_dai = { .capture = { .stream_name = "Capture", .channels_min = 1, @@ -499,7 +471,17 @@ static struct snd_soc_dai_driver atmel_pdmic_codec_dai = { .rates = SNDRV_PCM_RATE_KNOT, .formats = ATMEL_PDMIC_FORMATS, }, - .ops = &atmel_pdmic_codec_dai_ops, + .ops = &atmel_pdmic_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_pdmic_cpu_dai_component = { + .name = "atmel-pdmic", + .probe = atmel_pdmic_component_probe, + .controls = atmel_pdmic_snd_controls, + .num_controls = ARRAY_SIZE(atmel_pdmic_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, }; /* ASoC sound card */ @@ -528,9 +510,9 @@ static int atmel_pdmic_asoc_card_init(struct device *dev, dai_link->name = "PDMIC"; dai_link->stream_name = "PDMIC PCM"; - dai_link->codecs->dai_name = ATMEL_PDMIC_CODEC_DAI_NAME; + dai_link->codecs->dai_name = "snd-soc-dummy-dai"; dai_link->cpus->dai_name = dev_name(dev); - dai_link->codecs->name = dev_name(dev); + dai_link->codecs->name = "snd-soc-dummy"; dai_link->platforms->name = dev_name(dev); card->dai_link = dai_link; @@ -684,16 +666,6 @@ static int atmel_pdmic_probe(struct platform_device *pdev) return ret; } - /* register codec and codec dai */ - atmel_pdmic_codec_dai.capture.rate_min = rate_min; - atmel_pdmic_codec_dai.capture.rate_max = rate_max; - ret = devm_snd_soc_register_component(dev, &soc_component_dev_pdmic, - &atmel_pdmic_codec_dai, 1); - if (ret) { - dev_err(dev, "could not register component: %d\n", ret); - return ret; - } - /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) { -- 2.25.1

2 1

[PATCH v3] ASoC: atmel-classd: remove codec component
by Codrin Ciubotariu 10 Jul '20

10 Jul '20

The CPU and the codec both are represented now as components, so for CLASS-D we are registering two componenets with the same name. Since there is no actual codec, we will merge the codec component into the CPU one and use a dummy codec instead, for the DAI link. As a bonus, debugfs will no longer report an error when will try to create entries for both componenets with the same name. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Changes in v3: - fixed some newlines modified by no reason; Changes in v2: - removed no longer used ATMEL_CLASSD_CODEC_DAI_NAME macro; sound/soc/atmel/atmel-classd.c | 132 ++++++++++++--------------------- 1 file changed, 47 insertions(+), 85 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index e98601eccfa3..2d35b08f0565 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -120,39 +120,21 @@ static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + int err; regmap_write(dd->regmap, CLASSD_THR, 0x0); - return clk_prepare_enable(dd->pclk); -} - -static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - clk_disable_unprepare(dd->pclk); + err = clk_prepare_enable(dd->pclk); + if (err) + return err; + err = clk_prepare_enable(dd->gclk); + if (err) { + clk_disable_unprepare(dd->pclk); + return err; + } + return 0; } -static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { - .startup = atmel_classd_cpu_dai_startup, - .shutdown = atmel_classd_cpu_dai_shutdown, -}; - -static struct snd_soc_dai_driver atmel_classd_cpu_dai = { - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = ATMEL_CLASSD_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &atmel_classd_cpu_dai_ops, -}; - -static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { - .name = "atmel-classd", -}; - /* platform */ static int atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream, @@ -306,31 +288,10 @@ static int atmel_classd_component_resume(struct snd_soc_component *component) return regcache_sync(dd->regmap); } -static struct snd_soc_component_driver soc_component_dev_classd = { - .probe = atmel_classd_component_probe, - .resume = atmel_classd_component_resume, - .controls = atmel_classd_snd_controls, - .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -/* codec dai component */ -static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - return clk_prepare_enable(dd->gclk); -} - -static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai, - int mute) +static int atmel_classd_cpu_dai_digital_mute(struct snd_soc_dai *cpu_dai, + int mute) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK; @@ -373,13 +334,13 @@ static struct { }; static int -atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; int fs; int i, best, best_val, cur_val, ret; u32 mask, val; @@ -417,8 +378,8 @@ atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, } static void -atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); @@ -426,10 +387,10 @@ atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(dd->gclk); } -static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; snd_soc_component_update_bits(component, CLASSD_MR, CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK, @@ -439,10 +400,10 @@ static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, return 0; } -static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK; @@ -468,19 +429,16 @@ static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = { - .digital_mute = atmel_classd_codec_dai_digital_mute, - .startup = atmel_classd_codec_dai_startup, - .shutdown = atmel_classd_codec_dai_shutdown, - .hw_params = atmel_classd_codec_dai_hw_params, - .prepare = atmel_classd_codec_dai_prepare, - .trigger = atmel_classd_codec_dai_trigger, +static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { + .startup = atmel_classd_cpu_dai_startup, + .shutdown = atmel_classd_cpu_dai_shutdown, + .digital_mute = atmel_classd_cpu_dai_digital_mute, + .hw_params = atmel_classd_cpu_dai_hw_params, + .prepare = atmel_classd_cpu_dai_prepare, + .trigger = atmel_classd_cpu_dai_trigger, }; -#define ATMEL_CLASSD_CODEC_DAI_NAME "atmel-classd-hifi" - -static struct snd_soc_dai_driver atmel_classd_codec_dai = { - .name = ATMEL_CLASSD_CODEC_DAI_NAME, +static struct snd_soc_dai_driver atmel_classd_cpu_dai = { .playback = { .stream_name = "Playback", .channels_min = 1, @@ -488,7 +446,18 @@ static struct snd_soc_dai_driver atmel_classd_codec_dai = { .rates = ATMEL_CLASSD_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &atmel_classd_codec_dai_ops, + .ops = &atmel_classd_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { + .name = "atmel-classd", + .probe = atmel_classd_component_probe, + .resume = atmel_classd_component_resume, + .controls = atmel_classd_snd_controls, + .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, }; /* ASoC sound card */ @@ -517,9 +486,9 @@ static int atmel_classd_asoc_card_init(struct device *dev, dai_link->name = "CLASSD"; dai_link->stream_name = "CLASSD PCM"; - dai_link->codecs->dai_name = ATMEL_CLASSD_CODEC_DAI_NAME; + dai_link->codecs->dai_name = "snd-soc-dummy-dai"; dai_link->cpus->dai_name = dev_name(dev); - dai_link->codecs->name = dev_name(dev); + dai_link->codecs->name = "snd-soc-dummy"; dai_link->platforms->name = dev_name(dev); card->dai_link = dai_link; @@ -620,13 +589,6 @@ static int atmel_classd_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(dev, &soc_component_dev_classd, - &atmel_classd_codec_dai, 1); - if (ret) { - dev_err(dev, "could not register component: %d\n", ret); - return ret; - } - /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) { -- 2.25.1

2 1

[PATCH v3 00/13] ASoC: Clean-up W=1 build warnings - part1
by Pierre-Louis Bossart 10 Jul '20

10 Jul '20

Both Lee Jones and I submitted separate series, this is the first part of the merged result, which includes all previously reviewed patches or suggested changes along with trivial ones for CONFIG_ACPI. Lee Jones (2): ASoC: codecs: jz4770: Remove defined but never used variable 'mic_boost_tlv' ASoC: ti: omap-mcbsp-st: Remove set, but unused variable 'w' Pierre-Louis Bossart (11): ASoC: atmel: fix kernel-doc ASoC: samsung: spdif: fix kernel-doc ASoC: samsung: pcm: fix kernel-doc ASoC: tegra: tegra20_das: remove always-true comparison ASoC: uniphier: aio-core: fix kernel-doc ASoC: codecs: da7210: fix kernel-doc ASoC: codecs: da7219: fix 'defined but not used' warning ASoC: codecs: cros_ec_codec: fix 'defined but not used' warning ASoC: codecs: es8316: fix 'defined but not used' warning ASoC: codecs: max98390: fix 'defined but not used' warning ASoC: codecs: rt*: fix 'defined but not used' warning sound/soc/atmel/atmel-pcm-dma.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 1 + sound/soc/codecs/cros_ec_codec.c | 2 ++ sound/soc/codecs/da7210.c | 12 +++++++----- sound/soc/codecs/da7219.c | 2 ++ sound/soc/codecs/es8316.c | 2 ++ sound/soc/codecs/jz4770.c | 1 - sound/soc/codecs/max98390.c | 8 -------- sound/soc/codecs/rt274.c | 2 ++ sound/soc/codecs/rt286.c | 2 ++ sound/soc/codecs/rt298.c | 2 ++ sound/soc/codecs/rt5660.c | 2 ++ sound/soc/codecs/rt5677-spi.c | 2 ++ sound/soc/samsung/pcm.c | 5 +++++ sound/soc/samsung/spdif.c | 6 +++--- sound/soc/tegra/tegra20_das.c | 3 +-- sound/soc/ti/omap-mcbsp-st.c | 3 --- sound/soc/uniphier/aio-core.c | 7 +++---- 18 files changed, 37 insertions(+), 27 deletions(-) -- 2.25.1

3 15

[PATCH 1/2] ASoC: tlv320adcx140: Add ASI enable for channel 5-8
by Dan Murphy 10 Jul '20

10 Jul '20

Add the ALSA controls to enable the ASI for channels 5-8 Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 35fe8ee5bce9..f884abdb576e 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -313,6 +313,14 @@ static const struct snd_kcontrol_new adcx140_dapm_ch3_en_switch = SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 5, 1, 0); static const struct snd_kcontrol_new adcx140_dapm_ch4_en_switch = SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 4, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch5_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 3, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch6_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 2, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch7_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 1, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch8_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 0, 1, 0); static const struct snd_kcontrol_new adcx140_dapm_ch1_dre_en_switch = SOC_DAPM_SINGLE("Switch", ADCX140_CH1_CFG0, 0, 1, 0); @@ -406,6 +414,15 @@ static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("CH4_ASI_EN", SND_SOC_NOPM, 0, 0, &adcx140_dapm_ch4_en_switch), + SND_SOC_DAPM_SWITCH("CH5_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch5_en_switch), + SND_SOC_DAPM_SWITCH("CH6_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch6_en_switch), + SND_SOC_DAPM_SWITCH("CH7_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch7_en_switch), + SND_SOC_DAPM_SWITCH("CH8_ASI_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch8_en_switch), + SND_SOC_DAPM_SWITCH("DRE_ENABLE", SND_SOC_NOPM, 0, 0, &adcx140_dapm_dre_en_switch), @@ -446,6 +463,11 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"CH3_ASI_EN", "Switch", "CH3_ADC"}, {"CH4_ASI_EN", "Switch", "CH4_ADC"}, + {"CH5_ASI_EN", "Switch", "CH5_OUT"}, + {"CH6_ASI_EN", "Switch", "CH6_OUT"}, + {"CH7_ASI_EN", "Switch", "CH7_OUT"}, + {"CH8_ASI_EN", "Switch", "CH8_OUT"}, + {"Decimation Filter", "Linear Phase", "DRE_ENABLE"}, {"Decimation Filter", "Low Latency", "DRE_ENABLE"}, {"Decimation Filter", "Ultra-low Latency", "DRE_ENABLE"}, -- 2.27.0

2 2

[PATCH 0/5] ASoC: mediatek: mt8183-da7219: support machine driver for rt1015
by Tzung-Bi Shih 10 Jul '20

10 Jul '20

This series tries to reuse mt8183-da7219-max98357.c for supporting machine driver with rt1015 speaker amplifier. The first 3 patches refactor the code for easier to change for subsequent patches. The 4th patch adds document for the new proposed compatible string. The 5th patch changes the machine driver to support either "MAX98357A" or "RT1015" codecs. Tzung-Bi Shih (5): ASoC: mediatek: mt8183-da7219: sort header inclusions in alphabetical ASoC: mediatek: mt8183-da7219: remove forward declaration of headset_init ASoC: mediatek: mt8183-da7219: extract codec and DAI names ASoC: mediatek: mt8183-da7219: add compatible string for using rt1015 ASoC: mediatek: mt8183-da7219: support machine driver with rt1015 .../bindings/sound/mt8183-da7219-max98357.txt | 5 +- sound/soc/mediatek/Kconfig | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 244 ++++++++++++++---- 3 files changed, 197 insertions(+), 57 deletions(-) -- 2.27.0.383.g050319c2ae-goog

2 6

[PATCH v3 0/4] ASoC: codecs: add MAX98373 Soundwire driver
by Pierre-Louis Bossart 10 Jul '20

10 Jul '20

V3: Rebased on top of two fixes already merged from the v2 patchset - no code changes Added explicit commit reference in last commit message V2 with a number of cleanups: split between I2C and SoundWire modes, as done for rt5682, and updated Kconfigs. removed useless initializations common to both modes removed idle_bias on fixed register classified as volatile in error fixed SPDX comments Pierre-Louis Bossart (2): ASoC: codecs: max98373: split I2C and common parts ASoC: Intel: sof-sdw: add MAX98373 I2C dependencies Ryan Lee (1): ASoC: codecs: max98373: add SoundWire support randerwang (1): ASoC: Intel: sdw_max98373: add card_late_probe support sound/soc/codecs/Kconfig | 20 +- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/max98373-i2c.c | 612 +++++++++++++++ sound/soc/codecs/max98373-sdw.c | 887 ++++++++++++++++++++++ sound/soc/codecs/max98373-sdw.h | 72 ++ sound/soc/codecs/max98373.c | 611 +-------------- sound/soc/codecs/max98373.h | 17 +- sound/soc/intel/boards/Kconfig | 7 +- sound/soc/intel/boards/sof_sdw.c | 19 +- sound/soc/intel/boards/sof_sdw_common.h | 6 + sound/soc/intel/boards/sof_sdw_max98373.c | 12 + 11 files changed, 1668 insertions(+), 599 deletions(-) create mode 100644 sound/soc/codecs/max98373-i2c.c create mode 100644 sound/soc/codecs/max98373-sdw.c create mode 100644 sound/soc/codecs/max98373-sdw.h base-commit: 6940701c715e193282cf8b31f970b2ea6eb26341 -- 2.25.1

2 5

[PATCH] ASoC: rt286: fix unexpected interrupt happens
by shumingf＠realtek.com 10 Jul '20

10 Jul '20

From: Shuming Fan <shumingf(a)realtek.com> The HV/VREF should not turn off if the headphone jack plug-in. This patch could solve the unexpected interrupt issue in some devices. Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt286.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 9593a9a27bf8..e8d14eefc41b 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -272,13 +272,13 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf); *mic = buf & 0x80000000; } - if (!*mic) { + + if (!*hp) { snd_soc_dapm_disable_pin(dapm, "HV"); snd_soc_dapm_disable_pin(dapm, "VREF"); - } - if (!*hp) snd_soc_dapm_disable_pin(dapm, "LDO1"); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync(dapm); + } return 0; } -- 2.27.0

3 2

[PATCH][next] ASoC: codecs: Use fallthrough pseudo-keyword
by Gustavo A. R. Silva 10 Jul '20

10 Jul '20

Replace the existing /* fall through */ comments and its variants with the new pseudo-keyword macro fallthrough[1]. [1] https://www.kernel.org/doc/html/latest/process/deprecated.html?highlight=fa… Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/codecs/ab8500-codec.c | 2 +- sound/soc/codecs/adau1761.c | 4 ++-- sound/soc/codecs/adau17x1.c | 4 ++-- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/ak4613.c | 6 +++--- sound/soc/codecs/es8328.c | 4 ++-- sound/soc/codecs/max9860.c | 2 +- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/rt274.c | 4 ++-- sound/soc/codecs/rt5640.c | 4 ++-- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/sta350.c | 2 +- sound/soc/codecs/tas2552.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 3 ++- sound/soc/codecs/tpa6130a2.c | 2 +- sound/soc/codecs/wm8753.c | 6 ++++-- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8993.c | 4 ++-- sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- 29 files changed, 43 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index ea92007d1ef5..31a8c4162d20 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2126,7 +2126,7 @@ static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) dev_err(dai->component->dev, "%s: ERROR: The device is either a master or a slave.\n", __func__); - /* fall through */ + fallthrough; default: dev_err(dai->component->dev, "%s: ERROR: Unsupporter master mask 0x%x\n", diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 5ca9b744b7d8..fb006fc81653 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -642,7 +642,7 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component) ARRAY_SIZE(adau1761_jack_detect_controls)); if (ret) return ret; - /* fall through */ + fallthrough; case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes, ARRAY_SIZE(adau1761_no_dmic_routes)); @@ -693,7 +693,7 @@ static int adau1761_setup_headphone_mode(struct snd_soc_component *component) ADAU1761_PLAY_MONO_OUTPUT_VOL_UNMUTE, ADAU1761_PLAY_MONO_OUTPUT_VOL_MODE_HP | ADAU1761_PLAY_MONO_OUTPUT_VOL_UNMUTE); - /* fallthrough */ + fallthrough; case ADAU1761_OUTPUT_MODE_HEADPHONE: regmap_update_bits(adau->regmap, ADAU1761_PLAY_HP_RIGHT_VOL, ADAU1761_PLAY_HP_RIGHT_VOL_MODE_HP, diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index b6352de077b5..30e072c80ac1 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -385,7 +385,7 @@ static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, case ADAU17X1_CLK_SRC_PLL_AUTO: if (!adau->mclk) return -EINVAL; - /* Fall-through */ + fallthrough; case ADAU17X1_CLK_SRC_PLL: is_pll = true; break; @@ -469,7 +469,7 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, ret = adau17x1_auto_pll(dai, params); if (ret) return ret; - /* Fall-through */ + fallthrough; case ADAU17X1_CLK_SRC_PLL: freq = adau->pll_freq; break; diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index c4b9722c3d8f..4fd99280d7db 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -647,7 +647,7 @@ static int adav80x_set_pll(struct snd_soc_component *component, int pll_id, pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; break; } - /* fall through */ + fallthrough; default: return -EINVAL; } diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index d4d2f0d9231a..8d663e8d64c4 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -451,13 +451,13 @@ static int ak4613_set_bias_level(struct snd_soc_component *component, switch (level) { case SND_SOC_BIAS_ON: mgmt1 |= RSTN; - /* fall through */ + fallthrough; case SND_SOC_BIAS_PREPARE: mgmt1 |= PMADC | PMDAC; - /* fall through */ + fallthrough; case SND_SOC_BIAS_STANDBY: mgmt1 |= PMVR; - /* fall through */ + fallthrough; case SND_SOC_BIAS_OFF: default: break; diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index fdf64c29f563..757e740459fb 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -562,14 +562,14 @@ static int es8328_set_sysclk(struct snd_soc_dai *codec_dai, break; case 22579200: mclkdiv2 = 1; - /* fall through */ + fallthrough; case 11289600: es8328->sysclk_constraints = &constraints_11289; es8328->mclk_ratios = ratios_11289; break; case 24576000: mclkdiv2 = 1; - /* fall through */ + fallthrough; case 12288000: es8328->sysclk_constraints = &constraints_12288; es8328->mclk_ratios = ratios_12288; diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index 8be636fe6552..d5925c42b4b5 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -334,7 +334,7 @@ static int max9860_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } ifc1a ^= MAX9860_WCI; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_IB_NF: ifc1a ^= MAX9860_DBCI; ifc1b ^= MAX9860_ABCI; diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 30da00a3e789..4428c62e25cf 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -608,7 +608,7 @@ static int pm8916_wcd_analog_enable_adc(struct snd_soc_dapm_widget *w, case CDC_A_TX_2_EN: snd_soc_component_update_bits(component, CDC_A_MICB_1_CTL, MICB_1_CTL_CFILT_REF_SEL_MASK, 0); - /* fall through */ + fallthrough; case CDC_A_TX_3_EN: snd_soc_component_update_bits(component, CDC_D_CDC_CONN_TX2_CTL, CONN_TX2_SERIAL_TX2_MUX, diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c index cbb5e176d11a..923b8f919189 100644 --- a/sound/soc/codecs/rt274.c +++ b/sound/soc/codecs/rt274.c @@ -760,7 +760,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, break; default: dev_warn(component->dev, "invalid pll source, use BCLK\n"); - /* fall through */ + fallthrough; case RT274_PLL2_S_BCLK: snd_soc_component_update_bits(component, RT274_PLL2_CTRL, RT274_PLL2_SRC_MASK, RT274_PLL2_SRC_BCLK); @@ -788,7 +788,7 @@ static int rt274_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, break; default: dev_warn(component->dev, "invalid freq_in, assume 4.8M\n"); - /* fall through */ + fallthrough; case 100: snd_soc_component_write(component, 0x7a, 0xaab6); snd_soc_component_write(component, 0x7b, 0x0301); diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 3b2bb62a2136..1414ad15d01c 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1662,7 +1662,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_113: ret |= RT5640_U_IF1; - /* fall through */ + fallthrough; case RT5640_IF_312: case RT5640_IF_213: ret |= RT5640_U_IF2; @@ -1678,7 +1678,7 @@ static int get_sdp_info(struct snd_soc_component *component, int dai_id) break; case RT5640_IF_223: ret |= RT5640_U_IF1; - /* fall through */ + fallthrough; case RT5640_IF_123: case RT5640_IF_321: ret |= RT5640_U_IF2; diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index e9a051a50ab2..9e449d35fc28 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4609,7 +4609,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, break; case 25: slot_width_25 = 0x8080; - /* fall through */ + fallthrough; case 24: val |= (2 << 8); break; diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index e8d2ca4b4603..86528b930de8 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -697,7 +697,7 @@ static int sta32x_hw_params(struct snd_pcm_substream *substream, switch (params_width(params)) { case 24: dev_dbg(component->dev, "24bit\n"); - /* fall through */ + fallthrough; case 32: dev_dbg(component->dev, "24bit or 32bit\n"); switch (sta32x->format) { diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c index ccb7100b6644..75d3b0618ab5 100644 --- a/sound/soc/codecs/sta350.c +++ b/sound/soc/codecs/sta350.c @@ -726,7 +726,7 @@ static int sta350_hw_params(struct snd_pcm_substream *substream, switch (params_width(params)) { case 24: dev_dbg(component->dev, "24bit\n"); - /* fall through */ + fallthrough; case 32: dev_dbg(component->dev, "24bit or 32bit\n"); switch (sta350->format) { diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 529c0fb93f9b..d9d239d4256e 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -407,7 +407,7 @@ static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, clk_id = TAS2552_PLL_CLKIN_BCLK; freq = 0; } - /* fall through */ + fallthrough; case TAS2552_PLL_CLKIN_BCLK: case TAS2552_PLL_CLKIN_1_8_FIXED: mask = TAS2552_PLL_SRC_MASK; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index d22f75e8fb6a..7d5b6dbf6273 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -449,7 +449,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, break; case SND_SOC_DAIFMT_DSP_A: iface_reg |= TLV320AIC23_LRP_ON; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_B: iface_reg |= TLV320AIC23_FOR_DSP; break; diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 31daa60695bd..6694e56cfe1f 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1080,7 +1080,8 @@ static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_I2S: break; case SND_SOC_DAIFMT_DSP_A: - dsp_a_val = 0x1; /* fall through */ + dsp_a_val = 0x1; + fallthrough; case SND_SOC_DAIFMT_DSP_B: /* * NOTE: This CODEC samples on the falling edge of BCLK in diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 0b1f1a5e2a2d..e2d7ae615c52 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -261,7 +261,7 @@ static int tpa6130a2_probe(struct i2c_client *client, default: dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", data->id); - /* fall through */ + fallthrough; case TPA6130A2: regulator = "Vdd"; break; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a1b6765c8f23..f3c31121d100 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -966,7 +966,8 @@ static int wm8753_pcm_set_dai_fmt(struct snd_soc_component *component, case SND_SOC_DAIFMT_CBS_CFS: break; case SND_SOC_DAIFMT_CBM_CFM: - ioctl |= 0x2; /* fall through */ + ioctl |= 0x2; + fallthrough; case SND_SOC_DAIFMT_CBM_CFS: voice |= 0x0040; break; @@ -1091,7 +1092,8 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_component *component, case SND_SOC_DAIFMT_CBS_CFS: break; case SND_SOC_DAIFMT_CBM_CFM: - ioctl |= 0x1; /* fall through */ + ioctl |= 0x1; + fallthrough; case SND_SOC_DAIFMT_CBM_CFS: hifi |= 0x0040; break; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 5de663d61ba6..a52cb8fee82f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1927,7 +1927,7 @@ static int wm8903_set_pdata_irq_trigger(struct i2c_client *i2c, * We assume the controller imposes no restrictions, * so we are able to select active-high */ - /* Fall-through */ + fallthrough; case IRQ_TYPE_LEVEL_HIGH: pdata->irq_active_low = false; break; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3f0e49c51fd5..d54257097d56 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1436,7 +1436,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; @@ -1824,7 +1824,7 @@ static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id, break; } clk_id = WM8904_CLK_MCLK; - /* fallthrough */ + fallthrough; case WM8904_CLK_MCLK: priv->sysclk_src = clk_id; diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 73c192f58382..0630dcb66c6f 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -683,7 +683,7 @@ static int wm8955_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8955_LRP; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif |= 0x3; break; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9dca6e28032a..e1ab2be51ee7 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -836,7 +836,7 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, iface |= 0x000c; break; } - /* fall through */ + fallthrough; default: dev_err(component->dev, "unsupported width %d\n", params_width(params)); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index d11a38a0b283..e62a0a8ac297 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -650,7 +650,7 @@ static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_B: aif |= WM8961_LRP; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif |= 3; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 6ef022295f55..df8cdc71357d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2645,7 +2645,7 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif0 |= WM8962_LRCLK_INV | 3; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif0 |= 3; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 207c0211caa9..8c9f82efcceb 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1073,7 +1073,7 @@ static int wm8993_set_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case WM8993_SYSCLK_MCLK: wm8993->mclk_rate = freq; - /* fall through */ + fallthrough; case WM8993_SYSCLK_FLL: wm8993->sysclk_source = clk_id; break; @@ -1121,7 +1121,7 @@ static int wm8993_set_dai_fmt(struct snd_soc_dai *dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8993_AIF_LRCLK_INV; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 75242ec47406..903f8e81cd89 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -853,7 +853,7 @@ static void vmid_reference(struct snd_soc_component *component) switch (wm8994->vmid_mode) { default: WARN_ON(NULL == "Invalid VMID mode"); - /* fall through */ + fallthrough; case WM8994_VMID_NORMAL: /* Startup bias, VMID ramp & buffer */ snd_soc_component_update_bits(component, WM8994_ANTIPOP_2, @@ -2776,7 +2776,7 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8994_AIF1_LRCLK_INV; lrclk |= WM8958_AIF1_LRCLK_INV; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 276ffa84cc31..ec752819cb2c 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -1462,7 +1462,7 @@ static int wm8995_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif |= WM8995_AIF1_LRCLK_INV; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif |= (0x3 << WM8995_AIF1_FMT_SHIFT); break; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1d3b3f4e66b3..d303ef7571e9 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1854,7 +1854,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, case 24576000: ratediv = WM8996_SYSCLK_DIV; wm8996->sysclk /= 2; - /* fall through */ + fallthrough; case 11289600: case 12288000: snd_soc_component_update_bits(component, WM8996_AIF_RATE, diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index be5c9c2b0162..b5465e486fb5 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -929,7 +929,7 @@ static int wm9081_set_dai_fmt(struct snd_soc_dai *dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif2 |= WM9081_AIF_LRCLK_INV; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: aif2 |= 0x3; break; -- 2.27.0

2 1

[PATCH 0/3] ALSA: Convert remaining fall through comments
by Takashi Iwai 09 Jul '20

09 Jul '20

Hi, here is a simple patch set to replace or remove the remaining "fall through" comments in sound tree (modulo sound/soc/*). I suppose ASoC parts are coverted by another patch. Takashi === Takashi Iwai (3): ALSA: atmel: Remove invalid "fall through" comments ALSA: xen: Remove superfluous fall through comments ALSA: Replace with fallthrough pseudo keyword in the remaining places sound/atmel/ac97c.c | 4 ++-- sound/isa/es18xx.c | 2 +- sound/pci/au88x0/au88x0_core.c | 12 ++++++------ sound/pci/oxygen/oxygen_pcm.c | 2 +- sound/usb/caiaq/audio.c | 2 +- sound/usb/caiaq/device.c | 2 +- sound/usb/midi.c | 2 +- sound/xen/xen_snd_front.c | 4 ---- sound/xen/xen_snd_front_evtchnl.c | 4 ---- 9 files changed, 13 insertions(+), 21 deletions(-) -- 2.16.4

2 6

[PATCH v2] ALSA: Use fallthrough pseudo-keyword
by Gustavo A. R. Silva 09 Jul '20

09 Jul '20

Replace the existing /* fall through */ comments and its variants with the new pseudo-keyword macro fallthrough[1]. Also, remove unnecessary fall-through markings when it is the case. [1] https://www.kernel.org/doc/html/latest/process/deprecated.html?highlight=fa… Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- Changes in v2: - Remove unnecessary fall-through comment. sound/atmel/ac97c.c | 16 ++++++++-------- sound/core/memalloc.c | 2 +- sound/core/oss/pcm_oss.c | 2 +- sound/core/oss/pcm_plugin.c | 2 +- sound/core/pcm_native.c | 8 ++++---- sound/core/seq/oss/seq_oss_timer.c | 2 +- sound/core/seq/seq_midi_emul.c | 2 +- sound/drivers/opl3/opl3_midi.c | 4 ++-- sound/isa/galaxy/galaxy.c | 6 +++--- sound/isa/msnd/msnd_pinnacle_mixer.c | 4 +--- sound/isa/opti9xx/miro.c | 10 +++++----- sound/isa/opti9xx/opti92x-ad1848.c | 12 +++++++----- sound/isa/sb/sb8_main.c | 10 +++++----- sound/oss/dmasound/dmasound_atari.c | 2 +- sound/oss/dmasound/dmasound_core.c | 4 ++-- sound/pci/ac97/ac97_codec.c | 4 ++-- sound/pci/atiixp.c | 6 +++--- sound/pci/azt3328.c | 2 +- sound/pci/echoaudio/echoaudio.c | 4 ++-- sound/pci/emu10k1/emupcm.c | 5 ++--- sound/pci/hda/hda_beep.c | 2 +- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_realtek.c | 4 ++-- sound/pci/ice1712/delta.c | 2 +- sound/pci/intel8x0.c | 8 ++++---- sound/pci/mixart/mixart.c | 2 +- sound/pci/mixart/mixart_core.c | 2 +- sound/pci/nm256/nm256.c | 6 +++--- sound/pci/ymfpci/ymfpci_main.c | 2 +- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 4 ++-- sound/sparc/dbri.c | 8 ++++---- sound/usb/card.c | 2 +- sound/usb/clock.c | 2 +- sound/usb/pcm.c | 2 +- sound/usb/stream.c | 3 +-- 35 files changed, 78 insertions(+), 80 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index a1dce9725b98..a62f1cdad68f 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -356,14 +356,14 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) camr = ac97c_readl(chip, CAMR); switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ - case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: ptcr = ATMEL_PDC_TXTEN; camr |= AC97C_CMR_CENA | AC97C_CSR_ENDTX; break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ - case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: ptcr |= ATMEL_PDC_TXTDIS; if (chip->opened <= 1) @@ -388,14 +388,14 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) ptcr = readl(chip->regs + ATMEL_PDC_PTSR); switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ - case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: ptcr = ATMEL_PDC_RXTEN; camr |= AC97C_CMR_CENA | AC97C_CSR_ENDRX; break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ - case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: ptcr |= ATMEL_PDC_RXTDIS; if (chip->opened <= 1) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index d8f28d030985..ad74ea9cbff5 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -158,7 +158,7 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, */ dmab->dev.type = SNDRV_DMA_TYPE_DEV; #endif /* CONFIG_GENERIC_ALLOCATOR */ - /* fall through */ + fallthrough; case SNDRV_DMA_TYPE_DEV: case SNDRV_DMA_TYPE_DEV_UC: snd_malloc_dev_pages(dmab, size); diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 68630244b00f..327ec42a36b0 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2851,7 +2851,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; if (substream) break; - /* Fall through */ + fallthrough; case VM_READ: substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE]; break; diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 1545f8fdb4db..d5ca161d588c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -357,7 +357,7 @@ snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format, if (snd_mask_test(format_mask, (__force int)format1)) return format1; } - /* fall through */ + fallthrough; default: return (__force snd_pcm_format_t)-EINVAL; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1e51c849d3d4..9e0b2d73faf6 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1903,7 +1903,7 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, switch (substream->runtime->status->state) { case SNDRV_PCM_STATE_PAUSED: snd_pcm_pause(substream, false); - /* fallthru */ + fallthrough; case SNDRV_PCM_STATE_SUSPENDED: snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); break; @@ -2811,7 +2811,7 @@ static int do_pcm_hwsync(struct snd_pcm_substream *substream) case SNDRV_PCM_STATE_DRAINING: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) return -EBADFD; - /* Fall through */ + fallthrough; case SNDRV_PCM_STATE_RUNNING: return snd_pcm_update_hw_ptr(substream); case SNDRV_PCM_STATE_PREPARED: @@ -3814,7 +3814,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) case SNDRV_PCM_MMAP_OFFSET_STATUS_OLD: if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT)) return -ENXIO; - /* fallthrough */ + fallthrough; case SNDRV_PCM_MMAP_OFFSET_STATUS_NEW: if (!pcm_status_mmap_allowed(pcm_file)) return -ENXIO; @@ -3822,7 +3822,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) case SNDRV_PCM_MMAP_OFFSET_CONTROL_OLD: if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT)) return -ENXIO; - /* fallthrough */ + fallthrough; case SNDRV_PCM_MMAP_OFFSET_CONTROL_NEW: if (!pcm_control_mmap_allowed(pcm_file)) return -ENXIO; diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c index a35d429e4c27..f9f57232a83f 100644 --- a/sound/core/seq/oss/seq_oss_timer.c +++ b/sound/core/seq/oss/seq_oss_timer.c @@ -79,7 +79,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev) case TMR_WAIT_REL: parm += rec->cur_tick; rec->realtime = 0; - /* fall through */ + fallthrough; case TMR_WAIT_ABS: if (parm == 0) { rec->realtime = 1; diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 198f285594e3..81d2ef5e5811 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -309,7 +309,7 @@ do_control(const struct snd_midi_op *ops, void *drv, break; case MIDI_CTL_MSB_DATA_ENTRY: chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0; - /* fall through */ + fallthrough; case MIDI_CTL_LSB_DATA_ENTRY: if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED) rpn(ops, drv, chan, chset); diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 2f6e8023e05c..eb23c55323ae 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -354,7 +354,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) instr_4op = 1; break; } - /* fall through */ + fallthrough; default: spin_unlock_irqrestore(&opl3->voice_lock, flags); return; @@ -443,7 +443,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) switch (connection) { case 0x03: snd_opl3_calc_volume(&vol_op[2], vel, chan); - /* fallthru */ + fallthrough; case 0x02: snd_opl3_calc_volume(&vol_op[0], vel, chan); break; diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index ce409e75ae51..65f9f46c9f58 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -247,7 +247,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) break; case 2: irq[n] = 9; - /* Fall through */ + fallthrough; case 9: wss_config[n] |= WSS_CONFIG_IRQ_9; break; @@ -292,7 +292,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) case 1: if (dma1[n] == 0) break; - /* Fall through */ + fallthrough; default: dev_err(dev, "invalid capture DMA %d\n", dma2[n]); return 0; @@ -322,7 +322,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) break; case 2: mpu_irq[n] = 9; - /* Fall through */ + fallthrough; case 9: config[n] |= GALAXY_CONFIG_MPUIRQ_2; break; diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c index 02c566fca9e5..63633bd41e5b 100644 --- a/sound/isa/msnd/msnd_pinnacle_mixer.c +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -219,11 +219,9 @@ static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right) case MSND_MIXER_VOLUME: /* master volume */ writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft); writew(wRight, dev->SMA + SMA_wCurrMastVolRight); - /* fall through */ - + fallthrough; case MSND_MIXER_AUX: /* aux pot control */ /* scaled by master volume */ - /* fall through */ /* digital controls */ case MSND_MIXER_SYNTH: /* synth vol (dsp mix) */ diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b039429e6871..44ed1b65f6ce 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -163,13 +163,13 @@ static int aci_busy_wait(struct snd_miro_aci *aci) switch (timeout-ACI_MINTIME) { case 0 ... 9: out /= 10; - /* fall through */ + fallthrough; case 10 ... 19: out /= 10; - /* fall through */ + fallthrough; case 20 ... 30: out /= 10; - /* fall through */ + fallthrough; default: set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(out); @@ -824,7 +824,7 @@ static unsigned char snd_miro_read(struct snd_miro *chip, retval = inb(chip->mc_base + 9); break; } - /* fall through */ + fallthrough; case OPTi9XX_HW_82C929: retval = inb(chip->mc_base + reg); @@ -854,7 +854,7 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } - /* fall through */ + fallthrough; case OPTi9XX_HW_82C929: outb(value, chip->mc_base + reg); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 0e6d20e49158..881d3b5711d2 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -249,7 +249,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, retval = inb(chip->mc_base + 9); break; } - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -292,7 +292,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -343,7 +343,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc); /* enable wave audio */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C925: /* enable WSS mode */ @@ -380,7 +380,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: /* disable 3D sound (set GPIO1 as output, low) */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c); - /* fall through */ + fallthrough; + case OPTi9XX_HW_82C933: /* * The BTC 1817DW has QS1000 wavetable which is connected @@ -392,7 +393,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, * or digital input signal. */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); - /* fall through */ + fallthrough; + case OPTi9XX_HW_82C930: snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index e33dfe165276..86d0d2fdf48a 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -116,13 +116,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_20: chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; break; @@ -261,7 +261,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_20: chip->capture_format = SB_DSP_LO_INPUT_AUTO; break; @@ -361,7 +361,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ if (chip->hardware != SB_HW_JAZZ16) break; - /* fall through */ + fallthrough; case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; if (chip->playback_format == SB_DSP_OUTPUT) @@ -371,7 +371,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_CAPTURE_16: if (chip->hardware != SB_HW_JAZZ16) break; - /* fall through */ + fallthrough; case SB_MODE_CAPTURE_8: substream = chip->capture_substream; if (chip->capture_format == SB_DSP_INPUT) diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 823ccfa089b2..81c6a9830727 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -1449,7 +1449,7 @@ static int FalconMixerIoctl(u_int cmd, u_long arg) tt_dmasnd.input_gain = RECLEVEL_VOXWARE_TO_GAIN(data & 0xff) << 4 | RECLEVEL_VOXWARE_TO_GAIN(data >> 8 & 0xff); - /* fall through - return set value */ + fallthrough; /* return set value */ case SOUND_MIXER_READ_MIC: return IOCTL_OUT(arg, RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain >> 4 & 0xf) | diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index f802ea331e24..38f25e97538f 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -1478,13 +1478,13 @@ static int dmasound_setup(char *str) printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius); else catchRadius = ints[3]; - /* fall through */ + fallthrough; case 2: if (ints[1] < MIN_BUFFERS) printk("dmasound_setup: invalid number of buffers, using default = %d\n", numWriteBufs); else numWriteBufs = ints[1]; - /* fall through */ + fallthrough; case 1: if ((size = ints[2]) < 256) /* check for small buffer specs */ size <<= 10 ; diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6758c072000e..012a7ee849e8 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -218,11 +218,11 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg) case AC97_ID_ST_AC97_ID4: if (reg == 0x08) return 0; - /* fall through */ + fallthrough; case AC97_ID_ST7597: if (reg == 0x22 || reg == 0x7a) return 1; - /* fall through */ + fallthrough; case AC97_ID_AK4540: case AC97_ID_AK4542: if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 85d3b4e95489..a25d75455802 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -896,15 +896,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream) case 8: data |= ATI_REG_OUT_DMA_SLOT_BIT(10) | ATI_REG_OUT_DMA_SLOT_BIT(11); - /* fall through */ + fallthrough; case 6: data |= ATI_REG_OUT_DMA_SLOT_BIT(7) | ATI_REG_OUT_DMA_SLOT_BIT(8); - /* fall through */ + fallthrough; case 4: data |= ATI_REG_OUT_DMA_SLOT_BIT(6) | ATI_REG_OUT_DMA_SLOT_BIT(9); - /* fall through */ + fallthrough; default: data |= ATI_REG_OUT_DMA_SLOT_BIT(3) | ATI_REG_OUT_DMA_SLOT_BIT(4); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 58167d8469e1..77c7030ebbfa 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1232,7 +1232,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec, case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break; default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); - /* fall-through */ + fallthrough; case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break; case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break; case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index bee3271e29da..b07d3c5be9aa 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -690,7 +690,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream) break; case SNDRV_PCM_FORMAT_S32_BE: format.data_are_bigendian = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_FORMAT_S32_LE: format.bits_per_sample = 32; break; @@ -742,7 +742,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) pipe->last_counter = 0; pipe->position = 0; *pipe->dma_counter = 0; - /* fall through */ + fallthrough; case PIPE_STATE_PAUSED: pipe->state = PIPE_STATE_STARTED; break; diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index b934c6ac52dd..b2ddabb99438 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -753,7 +753,7 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); /* do we need this? */ snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[0]); - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) @@ -902,8 +902,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream, snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[i]); } snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); - - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL); diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 841523f6b88d..53a2b89f8983 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -102,7 +102,7 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, case SND_BELL: if (hz) hz = 1000; - /* fallthru */ + fallthrough; case SND_TONE: if (beep->linear_tone) beep->tone = beep_linear_tone(beep, hz); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d796963b80d7..be5000dd1585 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1018,7 +1018,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; case 0x14f150f2: codec->power_save_node = 1; - /* Fall through */ + fallthrough; default: codec->pin_amp_workaround = 1; snd_hda_pick_fixup(codec, cxt5066_fixup_models, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ad43812d63b5..605c76bcc52a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -382,7 +382,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0295: case 0x10ec0299: alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); - /* fallthrough */ + fallthrough; case 0x10ec0215: case 0x10ec0233: case 0x10ec0235: @@ -4698,7 +4698,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, break; case 0x10ec0867: alc_update_coefex_idx(codec, 0x57, 0x5, 0, 1<<14); - /* fallthru */ + fallthrough; case 0x10ec0221: case 0x10ec0662: snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 81929063b2fa..1d2a0287284b 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -691,7 +691,7 @@ static int snd_ice1712_delta_init(struct snd_ice1712 *ice) break; case ICE1712_SUBDEVICE_DELTADIO2496: ice->gpio.set_pro_rate = delta_1010_set_rate_val; - /* fall thru */ + fallthrough; case ICE1712_SUBDEVICE_DELTA66: ice->spdif.ops.open = delta_open_spdif; ice->spdif.ops.setup_rate = delta_setup_spdif; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 1781a1c081c3..675bb0da00a0 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -810,7 +810,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: val = ICH_IOCE | ICH_STARTBM; @@ -818,7 +818,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: val = 0; break; @@ -852,7 +852,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -869,7 +869,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* pause */ diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 7ba487443c7f..efff220b26ea 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -169,7 +169,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr, case PIPE_RUNNING: if(rate != 0) break; - /* fall through */ + fallthrough; default: if(rate == 0) return 0; /* nothing to do */ diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 048a2660d18d..0bdd33b0af65 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -527,7 +527,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) dev_err(&mgr->pci->dev, "canceled notification %x !\n", msg); } - /* fall through */ + fallthrough; case MSG_TYPE_ANSWER: /* answer or notification to a message we are waiting for*/ mutex_lock(&mgr->msg_lock); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index ebce4d259e06..240195e85e3d 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -560,7 +560,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: s->suspended = 0; - /* fallthru */ + fallthrough; case SNDRV_PCM_TRIGGER_START: if (! s->running) { snd_nm256_playback_start(chip, s, substream); @@ -569,7 +569,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: s->suspended = 1; - /* fallthru */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: if (s->running) { snd_nm256_playback_stop(chip); @@ -1664,7 +1664,7 @@ static int snd_nm256_probe(struct pci_dev *pci, return -ENODEV; case NM_RESET_WORKAROUND_2: reset_workaround_2 = 1; - /* Fall-through */ + fallthrough; case NM_RESET_WORKAROUND: reset_workaround = 1; break; diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index db7d76a3cfeb..cacc6a9d14c8 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -400,7 +400,7 @@ static int snd_ymfpci_playback_trigger(struct snd_pcm_substream *substream, kctl = chip->pcm_mixer[substream->number].ctl; kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; } - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: chip->ctrl_playback[ypcm->voices[0]->number + 1] = 0; diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 6cbe5cb34358..dfc4295b69c4 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -45,7 +45,7 @@ static int pdacf_pcm_trigger(struct snd_pcm_substream *subs, int cmd) case SNDRV_PCM_TRIGGER_START: chip->pcm_hwptr = 0; chip->pcm_tdone = 0; - /* fall thru */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: mask = 0; @@ -132,7 +132,7 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs) case SNDRV_PCM_FORMAT_S24_3LE: case SNDRV_PCM_FORMAT_S24_3BE: chip->pcm_sample = 3; - /* fall through */ + fallthrough; default: /* 24-bit */ aval = AK4117_DIF_24R; chip->pcm_frame = 3; diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index cf7049999261..17b79d45d17f 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -580,16 +580,16 @@ static __u32 reverse_bytes(__u32 b, int len) switch (len) { case 32: b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16); - /* fall through */ + fallthrough; case 16: b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8); - /* fall through */ + fallthrough; case 8: b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4); - /* fall through */ + fallthrough; case 4: b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2); - /* fall through */ + fallthrough; case 2: b = ((b & 0xaaaaaaaa) >> 1) | ((b & 0x55555555) << 1); case 1: diff --git a/sound/usb/card.c b/sound/usb/card.c index 162bdd6eb4d4..696e788c5d31 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -230,7 +230,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) dev_warn(&dev->dev, "unknown interface protocol %#02x, assuming v1\n", protocol); - /* fall through */ + fallthrough; case UAC_VERSION_1: { struct uac1_ac_header_descriptor *h1; diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b118cf97607f..f3ca59005d91 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -670,7 +670,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, else return 0; } - /* fall through */ + fallthrough; case UAC_VERSION_2: return set_sample_rate_v2v3(chip, iface, alts, fmt, rate); } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 4987b6dab84c..deb977e9c35a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1743,7 +1743,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea switch (cmd) { case SNDRV_PCM_TRIGGER_START: subs->trigger_tstamp_pending_update = true; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: subs->data_endpoint->prepare_data_urb = prepare_playback_urb; subs->data_endpoint->retire_data_urb = retire_playback_urb; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 15296f2c902c..4d1e6579e54d 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1146,9 +1146,8 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, dev_dbg(&dev->dev, "%u:%d: unknown interface protocol %#02x, assuming v1\n", iface_no, altno, protocol); protocol = UAC_VERSION_1; - /* fall through */ + fallthrough; case UAC_VERSION_1: - /* fall through */ case UAC_VERSION_2: { int bm_quirk = 0; -- 2.27.0

2 1

echoaudio: Fix some long standing bugs
by Mark Hills 09 Jul '20

09 Jul '20

Here follows some patches for the echoaudio PCM handling. Tested on the Layla3G on x86-64. I do have a Darla somewhere, but it's not practical for me to test with that right now. -- Mark

4 42

[PATCH 0/3] ASoC: mediatek: mt8183: support machine driver for rt1015
by Tzung-Bi Shih 09 Jul '20

09 Jul '20

This series tries to reuse mt8183-mt6358-ts3a227-max98357.c for supporting machine driver with rt1015 speaker amplifier. The 1st patch is straightforward: re-order the header inclusions. The 2nd patch adds document for the new proposed compatible string. The 3rd patch changes the machine driver to support either "MAX98357A" or "RT1015" codecs. Tzung-Bi Shih (3): ASoC: mediatek: mt8183: sort header inclusions in alphabetical dt-bindings: mt8183: add compatible string for using rt1015 ASoC: mediatek: mt8183: support machine driver with rt1015 .../sound/mt8183-mt6358-ts3a227-max98357.txt | 5 +- sound/soc/mediatek/Kconfig | 5 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 171 +++++++++++++++--- 3 files changed, 153 insertions(+), 28 deletions(-) -- 2.27.0.383.g050319c2ae-goog

2 6

Fixing sound on Asus UX534F / some UX533 models
by Hans de Goede 09 Jul '20

09 Jul '20

Hi Kai-Heng Feng, Takashi, I see that you are on the notification list for this bug: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1850439 So you may have already seen this. Some owners of affected laptops (which have non working speakers / headphone output atm), have done some heroic debugging work and come up with a set of 2 hda-verb commands which fix this. I'm not all that familiar with writing hda quirks, so I was hoping that one of you 2 can come up with a patch to fix this at the kernel level. This would also resolve these 2 bugs, which I believe are the same bug really: https://bugzilla.kernel.org/show_bug.cgi?id=206289 https://bugzilla.redhat.com/show_bug.cgi?id=1834751 Regards, Hans

3 10

ucm2: Add config for Rockchip/rk3399-gru-sound
by GitHub pull_request - edited 08 Jul '20

08 Jul '20

alsa-project/alsa-ucm-conf pull request #39 was edited from alpernebbi: This is the internal card for a number of ChromeOS devices based on the Gru board. Config is taken from ChromeOS sources [1] and modified to mostly look like Rockchip/max98090. Tested on a Samsung Chromebook Plus (rk3399-gru-kevin) running Linux 5.7+. [1] https://chromium.googlesource.com/chromiumos/overlays/board-overlays/+/75cf… Signed-off-by: Alper Nebi Yasak \<alpernebiyasak(a)gmail.com\> --- I've tested this on a Debian installation (with alsa-lib version 1.2.2) meaning I had to put the two files to /usr/share/alsa/ucm2/rk3399-gru-sound/ instead of .../Rockchip/rk3399-gru-sound, so I hope the latter is a correct path for them with later versions. All four devices work, but I couldn't get the Speaker/Headphones to play different things or the Mic/Headset to record simultaneously, so I marked those pairs as "ConflictingDevice"s. Without that, e.g. I see all devices in pulseaudio, but when I play two different audio files to Speaker/Headphones sinks I hear an alternating mix of both files on both devices; and both Mic/Headset sources end up recording only from the headset microphone in an alternating way. Pulseaudio doesn't recognize the headset jack, but acpi_listen does report "jack/headphone HEADPHONE plug" events. I'll look into this more, my best guess now is that the kernel doesn't expose the jack in a way it should. I also uploaded the alsa-info.sh output to cardinfo database: http://alsa-project.org/db/?f=9b4102975fa9c0f109af6ad687c856c620d76909 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/39 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Add config for Rockchip/rk3399-gru-sound
by GitHub pull_request - opened 08 Jul '20

08 Jul '20

alsa-project/alsa-ucm-conf pull request #39 was opened from alpernebbi: This is the internal card for a number of ChromeOS devices based on the Gru board. Config is taken from ChromeOS sources [1] and modified to mostly look like Rockchip/max98090. Tested on a Samsung Chromebook Plus (rk3399-gru-kevin) running Linux 5.7+. [1] https://chromium.googlesource.com/chromiumos/overlays/board-overlays/+/75cf… Signed-off-by: Alper Nebi Yasak <alpernebiyasak(a)gmail.com> --- I've tested this on a Debian installation (with alsa-lib version 1.2.2) meaning I had to put the two files to /usr/share/alsa/ucm2/rk3399-gru-sound/ instead of .../Rockchip/rk3399-gru-sound, so I hope the latter is a correct path for them with later versions. All four devices work, but I couldn't get the Speaker/Headphones to play different things or the Mic/Headset to record simultaneously, so I marked those pairs as "ConflictingDevice"s. Without that, e.g. I see all devices in pulseaudio, but when I play two different audio files to Speaker/Headphones sinks I hear an alternating mix of both files on both devices; and both Mic/Headset sources end up recording only from the headset microphone in an alternating way. Pulseaudio doesn't recognize the headset jack, but acpi_listen does report "jack/headphone HEADPHONE plug" events. I'll look into this more, my best guess now is that the kernel doesn't expose the jack in a way it should. I also uploaded the alsa-info.sh output to cardinfo database: http://alsa-project.org/db/?f=9b4102975fa9c0f109af6ad687c856c620d76909 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/39 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/realtek: Add quirk for Samsung Notebook Pen S
by Junar Wohn 08 Jul '20

08 Jul '20

Fixed no headphone sound bug on laptop Samsung Notebook Pen S (950SBE-951SBE), by using existing patch in Linus' tree, commit 14425f1f521f (ALSA: hda/realtek: Add quirk for Samsung Notebook). This laptop uses the same ALC298 but different subsystem id 0x144dc812. I added SND_PCI_QUIRK at sound/pci/hda/patch_realtek.c Signed-off-by: Joonho Wohn <doomsheart(a)gmail.com> --- 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 737ef82a75fd..ebedc6093642 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7528,6 +7528,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), + SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), --

1 0

[PATCH] ALSA: hda/realtek: Add quirk for Samsung Notebook Pen S
by Junar Wohn 08 Jul '20

08 Jul '20

Signed-off-by: Joonho Wohn <doomsheart(a)gmail.com> --- 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 737ef82a75fd..ebedc6093642 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7528,6 +7528,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), + SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), -- 2.25.1

2 1

aplay: Fix typo in error message
by GitHub pull_request - opened 08 Jul '20

08 Jul '20

alsa-project/alsa-utils pull request #44 was opened from trebmuh: replace #43 Request URL : https://github.com/alsa-project/alsa-utils/pull/44 Patch URL : https://github.com/alsa-project/alsa-utils/pull/44.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] regmap: add missing dependency on SoundWire
by Pierre-Louis Bossart 08 Jul '20

08 Jul '20

CONFIG_REGMAP is not selected when no other serial bus is supported. It's largely academic since CONFIG_I2C is usually selected e.g. by DRM, but still this can break randconfig so let's be explicit. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/base/regmap/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/base/regmap/Kconfig b/drivers/base/regmap/Kconfig index 0fd6f97ee523..1d1d26b0d279 100644 --- a/drivers/base/regmap/Kconfig +++ b/drivers/base/regmap/Kconfig @@ -4,7 +4,7 @@ # subsystems should select the appropriate symbols. config REGMAP - default y if (REGMAP_I2C || REGMAP_SPI || REGMAP_SPMI || REGMAP_W1 || REGMAP_AC97 || REGMAP_MMIO || REGMAP_IRQ || REGMAP_SCCB || REGMAP_I3C) + default y if (REGMAP_I2C || REGMAP_SPI || REGMAP_SPMI || REGMAP_W1 || REGMAP_AC97 || REGMAP_MMIO || REGMAP_IRQ || REGMAP_SOUNDWIRE || REGMAP_SCCB || REGMAP_I3C) select IRQ_DOMAIN if REGMAP_IRQ bool -- 2.25.1

3 3

[PATCH] ASoC: amd: fixed kernel warnings
by Vijendar Mukunda 08 Jul '20

08 Jul '20

This patch will fix unused variables kernel warnings when CONFIG_ACPI is disabled. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 165c93f..b943e59 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -168,8 +168,10 @@ static int snd_rn_acp_probe(struct pci_dev *pci, { struct acp_dev_data *adata; struct platform_device_info pdevinfo[ACP_DEVS]; +#if defined(CONFIG_ACPI) acpi_handle handle; acpi_integer dmic_status; +#endif unsigned int irqflags; int ret, index; u32 addr; -- 2.7.4

3 2

[PATCH] ASoC: SOF: core: fix null-ptr-deref bug during device removal
by Pierre-Louis Bossart 08 Jul '20

08 Jul '20

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> The DSP should be notified for device removal only if the probe was successful. Fixes the following KASAN bug: BUG: KASAN: null-ptr-deref in sof_ipc_tx_message+0x80/0x160 [snd_sof] Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 339c4930b0c0..adc7c37145d6 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -345,15 +345,15 @@ int snd_sof_device_remove(struct device *dev) struct snd_sof_pdata *pdata = sdev->pdata; int ret; - ret = snd_sof_dsp_power_down_notify(sdev); - if (ret < 0) - dev_warn(dev, "error: %d failed to prepare DSP for device removal", - ret); - if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) { + ret = snd_sof_dsp_power_down_notify(sdev); + if (ret < 0) + dev_warn(dev, "error: %d failed to prepare DSP for device removal", + ret); + snd_sof_fw_unload(sdev); snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); -- 2.25.1

2 1

[PATCH 0/5] ASoC: topology: fix error handling flow
by Pierre-Louis Bossart 08 Jul '20

08 Jul '20

While experimenting and introducing errors in Baytrail topology files until I got them right, I encountered multiple kernel oopses and memory leaks. This is a first batch to harden the code, but we should probably think of a tool to fuzz the topology... Pierre-Louis Bossart (5): ASoC: topology: fix kernel oops on route addition error ASoC: topology: fix tlvs in error handling for widget_dmixer ASoC: topology: use break on errors, not continue ASoC: topology: factor kfree(se) in error handling ASoC: topology: add more logs when topology load fails. sound/soc/soc-topology.c | 97 ++++++++++++++++++++++++---------------- 1 file changed, 58 insertions(+), 39 deletions(-) base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759 -- 2.25.1

2 6

[PATCH 00/11] ASoC: qdsp6: add gapless compressed audio support
by Srinivas Kandagatla 08 Jul '20

08 Jul '20

This patchset adds gapless compressed audio support on q6asm. Gapless on q6asm is implemented using 2 streams in a single asm session. First few patches are enhacements done to q6asm interface to allow stream id per each command, gapless flags and silence meta data. Along with this there are few trivial changes which I thought are necessary! Last patch implements copy callback to allow finer control over buffer offsets, specially in partial drain cases. This patchset is tested on RB3 aka DB845c platform. Thanks, srini Srinivas Kandagatla (11): ASoC: q6asm: add command opcode to timeout error report ASoC: q6asm: rename misleading session id variable ASoC: q6asm: make commands specific to streams ASoC: q6asm: use flags directly from asm-dai ASoC: q6asm: add length to write command token ASoC: q6asm: add support to remove intial and trailing silence ASoC: q6asm: add support to gapless flag in asm open ASoC: q6asm-dai: add next track metadata support ASoC: qdsp6: use dev_err instead of pr_err ASoC: qdsp6-dai: add gapless support ASoC: q6asm-dai: add support to copy callback sound/soc/qcom/qdsp6/q6asm-dai.c | 397 +++++++++++++++++++++++-------- sound/soc/qcom/qdsp6/q6asm.c | 173 +++++++++----- sound/soc/qcom/qdsp6/q6asm.h | 48 ++-- 3 files changed, 458 insertions(+), 160 deletions(-) -- 2.21.0

3 28

[PATCH 0/3] ASoC: more fixes for dpcm checks
by Pierre-Louis Bossart 08 Jul '20

08 Jul '20

This is hopefully the last set of fixes to avoid probe errors due to stricter checks of DAI capabilities introduced late in the 5.8 cycle. Daniel Baluta (1): ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M Pierre-Louis Bossart (2): ASoC: soc-dai: set dai_link dpcm_ flags with a helper ASoC: Intel: bdw-rt5677: fix non BE conversion include/sound/soc-dai.h | 1 + sound/soc/generic/audio-graph-card.c | 4 +-- sound/soc/generic/simple-card.c | 4 +-- sound/soc/intel/boards/bdw-rt5677.c | 1 + sound/soc/soc-dai.c | 38 ++++++++++++++++++++++++++++ sound/soc/sof/imx/imx8.c | 8 ++++++ sound/soc/sof/imx/imx8m.c | 8 ++++++ 7 files changed, 60 insertions(+), 4 deletions(-) base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759 -- 2.25.1

2 8

[PATCH v2 0/6] ASoC: codecs: add MAX98373 Soundwire driver
by Pierre-Louis Bossart 08 Jul '20

08 Jul '20

V2 with a number of cleanups: split between I2C and SoundWire modes, as done for rt5682, and updated Kconfigs. removed useless initializations common to both modes removed idle_bias on fixed register classified as volatile in error fixed SPDX comments Pierre-Louis Bossart (2): ASoC: codecs: max98373: split I2C and common parts ASoC: Intel: sof-sdw: add MAX98373 I2C dependencies Ryan Lee (2): ASoC: codecs: max98373: Removed superfluous volume control from chip default ASoC: codecs: max98373: add SoundWire support randerwang (2): ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend ASoC: Intel: sdw_max98373: add card_late_probe support sound/soc/codecs/Kconfig | 20 +- sound/soc/codecs/Makefile | 4 + sound/soc/codecs/max98373-i2c.c | 612 +++++++++++++++ sound/soc/codecs/max98373-sdw.c | 887 ++++++++++++++++++++++ sound/soc/codecs/max98373-sdw.h | 72 ++ sound/soc/codecs/max98373.c | 619 +-------------- sound/soc/codecs/max98373.h | 17 +- sound/soc/intel/boards/Kconfig | 7 +- sound/soc/intel/boards/sof_sdw.c | 19 +- sound/soc/intel/boards/sof_sdw_common.h | 6 + sound/soc/intel/boards/sof_sdw_max98373.c | 12 + 11 files changed, 1668 insertions(+), 607 deletions(-) create mode 100644 sound/soc/codecs/max98373-i2c.c create mode 100644 sound/soc/codecs/max98373-sdw.c create mode 100644 sound/soc/codecs/max98373-sdw.h base-commit: a5911ac5790acaf98c929b826b3f7b4a438f9759 -- 2.25.1

2 11

Typo fix in aplay.c
by GitHub pull_request - opened 08 Jul '20

08 Jul '20

alsa-project/alsa-utils pull request #43 was opened from trebmuh: HTH Request URL : https://github.com/alsa-project/alsa-utils/pull/43 Patch URL : https://github.com/alsa-project/alsa-utils/pull/43.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Re: [PATCH] Fixed no headphone sound bug on laptop Samsung Notebook Pen S, by using exsisting patch Linus' tree as commit 14425f1f521f (ALSA: hda/realtek: Add quirk for Samsung Notebook). This laptop uses same ALC298 but different subsystem id 0x144dc812. So I add SND_PCI_QUIRK at sound/pci/hda/patch_realtek.c
by Takashi Iwai 08 Jul '20

08 Jul '20

On Thu, 02 Jul 2020 10:17:09 +0200, Joonho Wohn wrote: > > --- > 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 737ef82a75fd..ebedc6093642 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -7528,6 +7528,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), > SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), > SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), > + SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), > SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), > SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), > SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), Thanks for the patch. The code change itself looks OK, but the patch has to be provided in a formal way described in Documentation/process/submitting-patches.rst. That is, - Move the description about your change into the mail body, not in the mail subject; keep the mail subject concise - Don't forget to give your Signed-off-by line, which is a legal requirement Once after fixing those, please resubmit the patch. thanks, Takashi

2 1

French translation update
by GitHub pull_request - edited 08 Jul '20

08 Jul '20

alsa-project/alsa-utils pull request #42 was edited from trebmuh: - spaces fixes - improvements - min/MAJ - update some strings - new strings translations - typo fixes Request URL : https://github.com/alsa-project/alsa-utils/pull/42 Patch URL : https://github.com/alsa-project/alsa-utils/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

French translation update
by GitHub pull_request - edited 08 Jul '20

08 Jul '20

alsa-project/alsa-utils pull request #42 was edited from trebmuh: - fixes spaces - improvements - min/MAJ - update some strings - new strings translations - typo fixes Request URL : https://github.com/alsa-project/alsa-utils/pull/42 Patch URL : https://github.com/alsa-project/alsa-utils/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

French translation update
by GitHub pull_request - opened 08 Jul '20

08 Jul '20

alsa-project/alsa-utils pull request #42 was opened from trebmuh: - fixes spaces - improvements - min/MAJ - update - new translations - typo fixes Request URL : https://github.com/alsa-project/alsa-utils/pull/42 Patch URL : https://github.com/alsa-project/alsa-utils/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/3] ASoC: Clean-up W=1 build warnings - part3
by Pierre-Louis Bossart 08 Jul '20

08 Jul '20

This is a much smaller set of cleanups, all related to warnings thrown by the use of GENMASK() with an unsigned variable. I just made the warning go away but I wonder if there's a better fix in the definition of GENMASK() itself? Pierre-Louis Bossart (3): ASoC: Intel: Skylake: remove comparison always false warning ASoC: meson: axg-pdm: remove comparison always false warning ASoC: meson: axg-spdifin: remove comparison always false warning sound/soc/intel/skylake/skl-sst-dsp.c | 2 +- sound/soc/meson/axg-pdm.c | 2 +- sound/soc/meson/axg-spdifin.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) -- 2.25.1

4 7

[PATCH 0/4] crypto: add sha256() function
by Eric Biggers 08 Jul '20

08 Jul '20

This series adds a function sha256() to the sha256 library so that users who want to compute a hash in one step can just call sha256() instead of sha256_init() + sha256_update() + sha256_final(). Patches 2-4 then convert some users to use it. Eric Biggers (4): crypto: lib/sha256 - add sha256() function efi: use sha256() instead of open coding mptcp: use sha256() instead of open coding ASoC: cros_ec_codec: use sha256() instead of open coding drivers/firmware/efi/embedded-firmware.c | 9 +++----- include/crypto/sha.h | 1 + lib/crypto/sha256.c | 10 +++++++++ net/mptcp/crypto.c | 15 +++---------- sound/soc/codecs/cros_ec_codec.c | 27 ++---------------------- 5 files changed, 19 insertions(+), 43 deletions(-) base-commit: 57c8aa43b9f272c382c253573c82be5cb68fe22d -- 2.27.0

4 5

[RESEND][PATCH v3 7/8] ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio
by Rohit kumar 08 Jul '20

08 Jul '20

From: Ajit Pandey <ajitp(a)codeaurora.org> Add platform driver for configuring sc7180 lpass core I2S and DMA configuration to support playback & capture to external codecs connected over primary & secondary MI2S interfaces. Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- Resending to update Signed-off mail id. sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-sc7180.c | 216 ++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 223 insertions(+) create mode 100644 sound/soc/qcom/lpass-sc7180.c diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 0ea4cde..87bec7f 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -24,6 +24,11 @@ config SND_SOC_LPASS_APQ8016 select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM +config SND_SOC_LPASS_SC7180 + tristate + select SND_SOC_LPASS_CPU + select SND_SOC_LPASS_PLATFORM + config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" depends on SND_SOC_QCOM diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 41b2c7a..7972c94 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -4,11 +4,13 @@ snd-soc-lpass-cpu-objs := lpass-cpu.o snd-soc-lpass-platform-objs := lpass-platform.o snd-soc-lpass-ipq806x-objs := lpass-ipq806x.o snd-soc-lpass-apq8016-objs := lpass-apq8016.o +snd-soc-lpass-sc7180-objs := lpass-sc7180.o obj-$(CONFIG_SND_SOC_LPASS_CPU) += snd-soc-lpass-cpu.o obj-$(CONFIG_SND_SOC_LPASS_PLATFORM) += snd-soc-lpass-platform.o obj-$(CONFIG_SND_SOC_LPASS_IPQ806X) += snd-soc-lpass-ipq806x.o obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o +obj-$(CONFIG_SND_SOC_LPASS_SC7180) += snd-soc-lpass-sc7180.o # Machine snd-soc-storm-objs := storm.o diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c new file mode 100644 index 00000000..dd85a97 --- /dev/null +++ b/sound/soc/qcom/lpass-sc7180.c @@ -0,0 +1,216 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Copyright (c) 2020, The Linux Foundation. All rights reserved. + * + * lpass-sc7180.c -- ALSA SoC platform-machine driver for QTi LPASS + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/err.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <dt-bindings/sound/sc7180-lpass.h> +#include <sound/pcm.h> +#include <sound/soc.h> + +#include "lpass-lpaif-reg.h" +#include "lpass.h" + +static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { + [MI2S_PRIMARY] = { + .id = MI2S_PRIMARY, + .name = "Primary MI2S", + .playback = { + .stream_name = "Primary Playback", + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .stream_name = "Primary Capture", + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + }, + .probe = &asoc_qcom_lpass_cpu_dai_probe, + .ops = &asoc_qcom_lpass_cpu_dai_ops, + }, + + [MI2S_SECONDARY] = { + .id = MI2S_SECONDARY, + .name = "Secondary MI2S", + .playback = { + .stream_name = "Secondary Playback", + .formats = SNDRV_PCM_FMTBIT_S16, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + }, + .probe = &asoc_qcom_lpass_cpu_dai_probe, + .ops = &asoc_qcom_lpass_cpu_dai_ops, + }, +}; + +static int sc7180_lpass_alloc_dma_channel(struct lpass_data *drvdata, + int direction) +{ + struct lpass_variant *v = drvdata->variant; + int chan = 0; + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + chan = find_first_zero_bit(&drvdata->dma_ch_bit_map, + v->rdma_channels); + + if (chan >= v->rdma_channels) + return -EBUSY; + } else { + chan = find_next_zero_bit(&drvdata->dma_ch_bit_map, + v->wrdma_channel_start + + v->wrdma_channels, + v->wrdma_channel_start); + + if (chan >= v->wrdma_channel_start + v->wrdma_channels) + return -EBUSY; + } + + set_bit(chan, &drvdata->dma_ch_bit_map); + + return chan; +} + +static int sc7180_lpass_free_dma_channel(struct lpass_data *drvdata, int chan) +{ + clear_bit(chan, &drvdata->dma_ch_bit_map); + + return 0; +} + +static int sc7180_lpass_init(struct platform_device *pdev) +{ + struct lpass_data *drvdata = platform_get_drvdata(pdev); + struct lpass_variant *variant = drvdata->variant; + struct device *dev = &pdev->dev; + int ret, i; + + drvdata->clks = devm_kcalloc(dev, variant->num_clks, + sizeof(*drvdata->clks), GFP_KERNEL); + drvdata->num_clks = variant->num_clks; + + for (i = 0; i < drvdata->num_clks; i++) + drvdata->clks[i].id = variant->clk_name[i]; + + ret = devm_clk_bulk_get(dev, drvdata->num_clks, drvdata->clks); + if (ret) { + dev_err(dev, "Failed to get clocks %d\n", ret); + return ret; + } + + ret = clk_bulk_prepare_enable(drvdata->num_clks, drvdata->clks); + if (ret) { + dev_err(dev, "sc7180 clk_enable failed\n"); + return ret; + } + + return 0; +} + +static int sc7180_lpass_exit(struct platform_device *pdev) +{ + struct lpass_data *drvdata = platform_get_drvdata(pdev); + + clk_bulk_disable_unprepare(drvdata->num_clks, drvdata->clks); + + return 0; +} + +static struct lpass_variant sc7180_data = { + .i2sctrl_reg_base = 0x1000, + .i2sctrl_reg_stride = 0x1000, + .i2s_ports = 3, + .irq_reg_base = 0x9000, + .irq_reg_stride = 0x1000, + .irq_ports = 3, + .rdma_reg_base = 0xC000, + .rdma_reg_stride = 0x1000, + .rdma_channels = 5, + .dmactl_audif_start = 1, + .wrdma_reg_base = 0x18000, + .wrdma_reg_stride = 0x1000, + .wrdma_channel_start = 5, + .wrdma_channels = 4, + + .loopback = REG_FIELD_ID(0x1000, 17, 17, 3, 0x1000), + .spken = REG_FIELD_ID(0x1000, 16, 16, 3, 0x1000), + .spkmode = REG_FIELD_ID(0x1000, 11, 15, 3, 0x1000), + .spkmono = REG_FIELD_ID(0x1000, 10, 10, 3, 0x1000), + .micen = REG_FIELD_ID(0x1000, 9, 9, 3, 0x1000), + .micmode = REG_FIELD_ID(0x1000, 4, 8, 3, 0x1000), + .micmono = REG_FIELD_ID(0x1000, 3, 3, 3, 0x1000), + .wssrc = REG_FIELD_ID(0x1000, 2, 2, 3, 0x1000), + .bitwidth = REG_FIELD_ID(0x1000, 0, 0, 3, 0x1000), + + .rdma_dyncclk = REG_FIELD_ID(0xC000, 21, 21, 5, 0x1000), + .rdma_bursten = REG_FIELD_ID(0xC000, 20, 20, 5, 0x1000), + .rdma_wpscnt = REG_FIELD_ID(0xC000, 16, 19, 5, 0x1000), + .rdma_intf = REG_FIELD_ID(0xC000, 12, 15, 5, 0x1000), + .rdma_fifowm = REG_FIELD_ID(0xC000, 1, 5, 5, 0x1000), + .rdma_enable = REG_FIELD_ID(0xC000, 0, 0, 5, 0x1000), + + .wrdma_dyncclk = REG_FIELD_ID(0x18000, 22, 22, 4, 0x1000), + .wrdma_bursten = REG_FIELD_ID(0x18000, 21, 21, 4, 0x1000), + .wrdma_wpscnt = REG_FIELD_ID(0x18000, 17, 20, 4, 0x1000), + .wrdma_intf = REG_FIELD_ID(0x18000, 12, 16, 4, 0x1000), + .wrdma_fifowm = REG_FIELD_ID(0x18000, 1, 5, 4, 0x1000), + .wrdma_enable = REG_FIELD_ID(0x18000, 0, 0, 4, 0x1000), + + .clk_name = (const char*[]) { + "noc", + "audio-core", + "sysnoc_mport", + }, + .num_clks = 3, + .dai_driver = sc7180_lpass_cpu_dai_driver, + .num_dai = ARRAY_SIZE(sc7180_lpass_cpu_dai_driver), + .dai_osr_clk_names = (const char *[]) { + "mclk0", + "null", + }, + .dai_bit_clk_names = (const char *[]) { + "pri_ibit", + "sec_ibit", + }, + .init = sc7180_lpass_init, + .exit = sc7180_lpass_exit, + .alloc_dma_channel = sc7180_lpass_alloc_dma_channel, + .free_dma_channel = sc7180_lpass_free_dma_channel, +}; + +static const struct of_device_id sc7180_lpass_cpu_device_id[] = { + {.compatible = "qcom,lpass-cpu-sc7180", .data = &sc7180_data}, + {} +}; + +static struct platform_driver sc7180_lpass_cpu_platform_driver = { + .driver = { + .name = "sc7180-lpass-cpu", + .of_match_table = of_match_ptr(sc7180_lpass_cpu_device_id), + }, + .probe = asoc_qcom_lpass_cpu_platform_probe, + .remove = asoc_qcom_lpass_cpu_platform_probe, +}; + +module_platform_driver(sc7180_lpass_cpu_platform_driver); + +MODULE_DESCRIPTION("SC7180 LPASS CPU DRIVER"); +MODULE_LICENSE("GPL v2"); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH v2] ASoC: atmel-classd: remove codec component
by Codrin Ciubotariu 08 Jul '20

08 Jul '20

The CPU and the codec both are represented now as components, so for CLASS-D we are registering two componenets with the same name. Since there is no actual codec, we will merge the codec component into the CPU one and use a dummy codec instead, for the DAI link. As a bonus, debugfs will no longer report an error when will try to create entries for both componenets with the same name. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- Changes in v2: - removed no longer used ATMEL_CLASSD_CODEC_DAI_NAME macro; sound/soc/atmel/atmel-classd.c | 134 ++++++++++++--------------------- 1 file changed, 48 insertions(+), 86 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index e98601eccfa3..24bc93c468af 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -5,7 +5,6 @@ * * Author: Songjun Wu <songjun.wu(a)atmel.com> */ - #include <linux/of.h> #include <linux/clk.h> #include <linux/module.h> @@ -120,39 +119,21 @@ static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + int err; regmap_write(dd->regmap, CLASSD_THR, 0x0); - return clk_prepare_enable(dd->pclk); -} - -static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - clk_disable_unprepare(dd->pclk); + err = clk_prepare_enable(dd->pclk); + if (err) + return err; + err = clk_prepare_enable(dd->gclk); + if (err) { + clk_disable_unprepare(dd->pclk); + return err; + } + return 0; } -static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { - .startup = atmel_classd_cpu_dai_startup, - .shutdown = atmel_classd_cpu_dai_shutdown, -}; - -static struct snd_soc_dai_driver atmel_classd_cpu_dai = { - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = ATMEL_CLASSD_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &atmel_classd_cpu_dai_ops, -}; - -static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { - .name = "atmel-classd", -}; - /* platform */ static int atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream, @@ -306,31 +287,10 @@ static int atmel_classd_component_resume(struct snd_soc_component *component) return regcache_sync(dd->regmap); } -static struct snd_soc_component_driver soc_component_dev_classd = { - .probe = atmel_classd_component_probe, - .resume = atmel_classd_component_resume, - .controls = atmel_classd_snd_controls, - .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -/* codec dai component */ -static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - return clk_prepare_enable(dd->gclk); -} - -static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai, - int mute) +static int atmel_classd_cpu_dai_digital_mute(struct snd_soc_dai *cpu_dai, + int mute) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK; @@ -373,13 +333,13 @@ static struct { }; static int -atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; int fs; int i, best, best_val, cur_val, ret; u32 mask, val; @@ -417,8 +377,8 @@ atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, } static void -atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); @@ -426,10 +386,10 @@ atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(dd->gclk); } -static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; snd_soc_component_update_bits(component, CLASSD_MR, CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK, @@ -439,10 +399,10 @@ static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, return 0; } -static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK; @@ -468,19 +428,16 @@ static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = { - .digital_mute = atmel_classd_codec_dai_digital_mute, - .startup = atmel_classd_codec_dai_startup, - .shutdown = atmel_classd_codec_dai_shutdown, - .hw_params = atmel_classd_codec_dai_hw_params, - .prepare = atmel_classd_codec_dai_prepare, - .trigger = atmel_classd_codec_dai_trigger, +static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { + .startup = atmel_classd_cpu_dai_startup, + .shutdown = atmel_classd_cpu_dai_shutdown, + .digital_mute = atmel_classd_cpu_dai_digital_mute, + .hw_params = atmel_classd_cpu_dai_hw_params, + .prepare = atmel_classd_cpu_dai_prepare, + .trigger = atmel_classd_cpu_dai_trigger, }; -#define ATMEL_CLASSD_CODEC_DAI_NAME "atmel-classd-hifi" - -static struct snd_soc_dai_driver atmel_classd_codec_dai = { - .name = ATMEL_CLASSD_CODEC_DAI_NAME, +static struct snd_soc_dai_driver atmel_classd_cpu_dai = { .playback = { .stream_name = "Playback", .channels_min = 1, @@ -488,7 +445,18 @@ static struct snd_soc_dai_driver atmel_classd_codec_dai = { .rates = ATMEL_CLASSD_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &atmel_classd_codec_dai_ops, + .ops = &atmel_classd_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { + .name = "atmel-classd", + .probe = atmel_classd_component_probe, + .resume = atmel_classd_component_resume, + .controls = atmel_classd_snd_controls, + .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, }; /* ASoC sound card */ @@ -517,9 +485,10 @@ static int atmel_classd_asoc_card_init(struct device *dev, dai_link->name = "CLASSD"; dai_link->stream_name = "CLASSD PCM"; - dai_link->codecs->dai_name = ATMEL_CLASSD_CODEC_DAI_NAME; + dai_link->codecs->dai_name = "snd-soc-dummy-dai"; + dai_link->cpus->dai_name = dev_name(dev); - dai_link->codecs->name = dev_name(dev); + dai_link->codecs->name = "snd-soc-dummy"; dai_link->platforms->name = dev_name(dev); card->dai_link = dai_link; @@ -620,13 +589,6 @@ static int atmel_classd_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(dev, &soc_component_dev_classd, - &atmel_classd_codec_dai, 1); - if (ret) { - dev_err(dev, "could not register component: %d\n", ret); - return ret; - } - /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) { -- 2.25.1

1 0

[PATCH] ASoC: atmel-classd: remove codec component
by Codrin Ciubotariu 08 Jul '20

08 Jul '20

The CPU and the codec both are represented now as components, so for CLASS-D we are registering two componenets with the same name. Since there is no actual codec, we will merge the codec component into the CPU one and use a dummy codec instead, for the DAI link. As a bonus, debugfs will no longer report an error when will try to create entries for both componenets with the same name. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/atmel/atmel-classd.c | 133 ++++++++++++--------------------- 1 file changed, 48 insertions(+), 85 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index e98601eccfa3..d1ce67d30cd7 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -5,7 +5,6 @@ * * Author: Songjun Wu <songjun.wu(a)atmel.com> */ - #include <linux/of.h> #include <linux/clk.h> #include <linux/module.h> @@ -120,39 +119,21 @@ static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); + int err; regmap_write(dd->regmap, CLASSD_THR, 0x0); - return clk_prepare_enable(dd->pclk); -} - -static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - clk_disable_unprepare(dd->pclk); + err = clk_prepare_enable(dd->pclk); + if (err) + return err; + err = clk_prepare_enable(dd->gclk); + if (err) { + clk_disable_unprepare(dd->pclk); + return err; + } + return 0; } -static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { - .startup = atmel_classd_cpu_dai_startup, - .shutdown = atmel_classd_cpu_dai_shutdown, -}; - -static struct snd_soc_dai_driver atmel_classd_cpu_dai = { - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = ATMEL_CLASSD_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE,}, - .ops = &atmel_classd_cpu_dai_ops, -}; - -static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { - .name = "atmel-classd", -}; - /* platform */ static int atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream, @@ -306,31 +287,10 @@ static int atmel_classd_component_resume(struct snd_soc_component *component) return regcache_sync(dd->regmap); } -static struct snd_soc_component_driver soc_component_dev_classd = { - .probe = atmel_classd_component_probe, - .resume = atmel_classd_component_resume, - .controls = atmel_classd_snd_controls, - .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), - .idle_bias_on = 1, - .use_pmdown_time = 1, - .endianness = 1, - .non_legacy_dai_naming = 1, -}; - -/* codec dai component */ -static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - - return clk_prepare_enable(dd->gclk); -} - -static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai, - int mute) +static int atmel_classd_cpu_dai_digital_mute(struct snd_soc_dai *cpu_dai, + int mute) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK; @@ -373,13 +333,13 @@ static struct { }; static int -atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; int fs; int i, best, best_val, cur_val, ret; u32 mask, val; @@ -417,8 +377,8 @@ atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream, } static void -atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card); @@ -426,10 +386,10 @@ atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(dd->gclk); } -static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; snd_soc_component_update_bits(component, CLASSD_MR, CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK, @@ -439,10 +399,10 @@ static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream, return 0; } -static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *codec_dai) +static int atmel_classd_cpu_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *cpu_dai) { - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component = cpu_dai->component; u32 mask, val; mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK; @@ -468,19 +428,17 @@ static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream, return 0; } -static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = { - .digital_mute = atmel_classd_codec_dai_digital_mute, - .startup = atmel_classd_codec_dai_startup, - .shutdown = atmel_classd_codec_dai_shutdown, - .hw_params = atmel_classd_codec_dai_hw_params, - .prepare = atmel_classd_codec_dai_prepare, - .trigger = atmel_classd_codec_dai_trigger, +static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = { + .startup = atmel_classd_cpu_dai_startup, + .shutdown = atmel_classd_cpu_dai_shutdown, + .digital_mute = atmel_classd_cpu_dai_digital_mute, + .hw_params = atmel_classd_cpu_dai_hw_params, + .prepare = atmel_classd_cpu_dai_prepare, + .trigger = atmel_classd_cpu_dai_trigger, }; #define ATMEL_CLASSD_CODEC_DAI_NAME "atmel-classd-hifi" - -static struct snd_soc_dai_driver atmel_classd_codec_dai = { - .name = ATMEL_CLASSD_CODEC_DAI_NAME, +static struct snd_soc_dai_driver atmel_classd_cpu_dai = { .playback = { .stream_name = "Playback", .channels_min = 1, @@ -488,7 +446,18 @@ static struct snd_soc_dai_driver atmel_classd_codec_dai = { .rates = ATMEL_CLASSD_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &atmel_classd_codec_dai_ops, + .ops = &atmel_classd_cpu_dai_ops, +}; + +static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = { + .name = "atmel-classd", + .probe = atmel_classd_component_probe, + .resume = atmel_classd_component_resume, + .controls = atmel_classd_snd_controls, + .num_controls = ARRAY_SIZE(atmel_classd_snd_controls), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, }; /* ASoC sound card */ @@ -517,9 +486,10 @@ static int atmel_classd_asoc_card_init(struct device *dev, dai_link->name = "CLASSD"; dai_link->stream_name = "CLASSD PCM"; - dai_link->codecs->dai_name = ATMEL_CLASSD_CODEC_DAI_NAME; + dai_link->codecs->dai_name = "snd-soc-dummy-dai"; + dai_link->cpus->dai_name = dev_name(dev); - dai_link->codecs->name = dev_name(dev); + dai_link->codecs->name = "snd-soc-dummy"; dai_link->platforms->name = dev_name(dev); card->dai_link = dai_link; @@ -620,13 +590,6 @@ static int atmel_classd_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(dev, &soc_component_dev_classd, - &atmel_classd_codec_dai, 1); - if (ret) { - dev_err(dev, "could not register component: %d\n", ret); - return ret; - } - /* register sound card */ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); if (!card) { -- 2.25.1

1 0

[PATCH v2 00/28] Clean-up ASoC's W=1 build warnings
by Lee Jones 08 Jul '20

08 Jul '20

This set is part of a larger effort attempting to clean-up W=1 kernel builds, which are currently overwhelmingly riddled with niggly little warnings. After these patches are applied, the build system no longer complains about any W=0 nor W=1 level warnings in sound/soc, when building for Arm. Hurrah! Changelog v1 => v2: - Remove whole read from patch 21 - Suggested by Peter Lee Jones (28): ASoC: atmel: atmel-pcm-dma: Demote function header which is clearly not kerneldoc ASoC: atmel: atmel_ssc_dai: Demote function header which is clearly not kerneldoc ASoC: fsl: fsl-asoc-card: Use correct format when providing struct documentation ASoC: fsl: fsl_asrc: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_ssi: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_ssi_dbg: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_spdif: Use correct formatting when denoting struct documentation ASoC: fsl: fsl_spdif: Update 'struct fsl_spdif_priv's descriptions ASoC: fsl: fsl_esai: Fix a bunch of kerneldoc issues ASoC: fsl: fsl_esai: Add descriptions for 'tx_mask' and 'rx_mask' to 'fsl_esai' ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header ASoC: codecs: ak4458: Remove set but never checked variable 'ret' ASoC: qcom: qdsp6: q6asm: Provide documentation for WMA 'codec_profile' ASoC: samsung: pcm: Demote half-documented kerneldoc header ASoC: samsung: spdif: Fix 3 misspelled struct properties ASoC: sunxi: sun4i-i2s: Demote half-documented kerneldoc header ASoC: uniphier: aio-core: Fix incorrectly named struct property and remove another ASoC: sunxi: sun4i-spdif: Fix misspelling of 'reg_dac_txdata' ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and 'reg_val_TSTDR' ASoC: sti: uniperif: Mark 'uni_tdm_hw' as __maybe_unused ASoC: ti: omap-mcbsp-st: Remove set, but unused variable 'w' ASoC: codecs: da7210: Fix formatting issues in da7210_set_dai_pll() header ASoC: soc-ac97: Demote seemingly unintentional kerneldoc header ASoC: codecs: jz4770: Remove defined but never used variable 'mic_boost_tlv' ASoC: codecs: rt5631: Demote misuse of kerneldoc to standard comment blocks ASoC: codecs: rt5659: Remove many unused const variables ASoC: codecs: tlv320aic26: Demote seemingly unintentional kerneldoc header ASoC: codecs: wm8400: Remove a bunch of unused variables sound/soc/atmel/atmel-pcm-dma.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/ak4458.c | 6 +++--- sound/soc/codecs/da7210.c | 12 ++++++----- sound/soc/codecs/jz4770.c | 1 - sound/soc/codecs/rt5631.c | 12 +++++------ sound/soc/codecs/rt5659.c | 37 --------------------------------- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/wm8400.c | 12 ----------- sound/soc/fsl/fsl-asoc-card.c | 16 +++++++------- sound/soc/fsl/fsl_asrc.c | 28 ++++++++++++------------- sound/soc/fsl/fsl_esai.c | 19 ++++++++++------- sound/soc/fsl/fsl_spdif.c | 5 +++-- sound/soc/fsl/fsl_ssi.c | 28 ++++++++++++------------- sound/soc/fsl/fsl_ssi_dbg.c | 4 ++-- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/qcom/qdsp6/q6asm.c | 1 + sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/spdif.c | 6 +++--- sound/soc/soc-ac97.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/ti/omap-mcbsp-st.c | 3 --- sound/soc/uniphier/aio-core.c | 7 +++---- sound/soc/ux500/ux500_msp_i2s.c | 8 +++---- 26 files changed, 88 insertions(+), 135 deletions(-) -- 2.25.1

4 31

[PATCH v2] ALSA: usb/line6: remove 'defined but not used' warning
by Pierre-Louis Bossart 08 Jul '20

08 Jul '20

Fix W=1 warning. Variables are declared in a header file included from multiple C files, replace by #defines as suggested by Takashi sound/usb/line6/driver.h:70:18: warning: ‘SYSEX_EXTRA_SIZE’ defined but not used [-Wunused-const-variable=] 70 | static const int SYSEX_EXTRA_SIZE = sizeof(line6_midi_id) + 4; | ^~~~~~~~~~~~~~~~ sound/usb/line6/driver.h:69:18: warning: ‘SYSEX_DATA_OFS’ defined but not used [-Wunused-const-variable=] 69 | static const int SYSEX_DATA_OFS = sizeof(line6_midi_id) + 3; | ^~~~~~~~~~~~~~ Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/usb/line6/driver.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 1a4e3700c80c..ec20fbec6475 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -66,8 +66,8 @@ extern const unsigned char line6_midi_id[3]; -static const int SYSEX_DATA_OFS = sizeof(line6_midi_id) + 3; -static const int SYSEX_EXTRA_SIZE = sizeof(line6_midi_id) + 4; +#define SYSEX_DATA_OFS (sizeof(line6_midi_id) + 3) +#define SYSEX_EXTRA_SIZE (sizeof(line6_midi_id) + 4) /* Common properties of Line 6 devices. -- 2.25.1

2 1

[PATCH] ALSA: Use fallthrough pseudo-keyword
by Gustavo A. R. Silva 08 Jul '20

08 Jul '20

Replace the existing /* fall through */ comments and its variants with the new pseudo-keyword macro fallthrough[1]. Also, remove unnecessary fall-through markings when it is the case. [1] https://www.kernel.org/doc/html/latest/process/deprecated.html?highlight=fa… Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/atmel/ac97c.c | 16 ++++++++-------- sound/core/memalloc.c | 2 +- sound/core/oss/pcm_oss.c | 2 +- sound/core/oss/pcm_plugin.c | 2 +- sound/core/pcm_native.c | 8 ++++---- sound/core/seq/oss/seq_oss_timer.c | 2 +- sound/core/seq/seq_midi_emul.c | 2 +- sound/drivers/opl3/opl3_midi.c | 4 ++-- sound/isa/galaxy/galaxy.c | 6 +++--- sound/isa/msnd/msnd_pinnacle_mixer.c | 3 +-- sound/isa/opti9xx/miro.c | 10 +++++----- sound/isa/opti9xx/opti92x-ad1848.c | 12 +++++++----- sound/isa/sb/sb8_main.c | 10 +++++----- sound/oss/dmasound/dmasound_atari.c | 2 +- sound/oss/dmasound/dmasound_core.c | 4 ++-- sound/pci/ac97/ac97_codec.c | 4 ++-- sound/pci/atiixp.c | 6 +++--- sound/pci/azt3328.c | 2 +- sound/pci/echoaudio/echoaudio.c | 4 ++-- sound/pci/emu10k1/emupcm.c | 5 ++--- sound/pci/hda/hda_beep.c | 2 +- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_realtek.c | 4 ++-- sound/pci/ice1712/delta.c | 2 +- sound/pci/intel8x0.c | 8 ++++---- sound/pci/mixart/mixart.c | 2 +- sound/pci/mixart/mixart_core.c | 2 +- sound/pci/nm256/nm256.c | 6 +++--- sound/pci/ymfpci/ymfpci_main.c | 2 +- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 4 ++-- sound/sparc/dbri.c | 8 ++++---- sound/usb/card.c | 2 +- sound/usb/clock.c | 2 +- sound/usb/pcm.c | 2 +- sound/usb/stream.c | 3 +-- 35 files changed, 78 insertions(+), 79 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index a1dce9725b98..a62f1cdad68f 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -356,14 +356,14 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) camr = ac97c_readl(chip, CAMR); switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ - case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: ptcr = ATMEL_PDC_TXTEN; camr |= AC97C_CMR_CENA | AC97C_CSR_ENDTX; break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ - case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: ptcr |= ATMEL_PDC_TXTDIS; if (chip->opened <= 1) @@ -388,14 +388,14 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) ptcr = readl(chip->regs + ATMEL_PDC_PTSR); switch (cmd) { - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ - case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: ptcr = ATMEL_PDC_RXTEN; camr |= AC97C_CMR_CENA | AC97C_CSR_ENDRX; break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ - case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: ptcr |= ATMEL_PDC_RXTDIS; if (chip->opened <= 1) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index bea46ed157a6..9f3e14aebe30 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -157,7 +157,7 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size, */ dmab->dev.type = SNDRV_DMA_TYPE_DEV; #endif /* CONFIG_GENERIC_ALLOCATOR */ - /* fall through */ + fallthrough; case SNDRV_DMA_TYPE_DEV: case SNDRV_DMA_TYPE_DEV_UC: snd_malloc_dev_pages(dmab, size); diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 68630244b00f..327ec42a36b0 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2851,7 +2851,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; if (substream) break; - /* Fall through */ + fallthrough; case VM_READ: substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE]; break; diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 1545f8fdb4db..d5ca161d588c 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -357,7 +357,7 @@ snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format, if (snd_mask_test(format_mask, (__force int)format1)) return format1; } - /* fall through */ + fallthrough; default: return (__force snd_pcm_format_t)-EINVAL; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9630d2523948..84b140f800fd 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1903,7 +1903,7 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, switch (substream->runtime->status->state) { case SNDRV_PCM_STATE_PAUSED: snd_pcm_pause(substream, false); - /* fallthru */ + fallthrough; case SNDRV_PCM_STATE_SUSPENDED: snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); break; @@ -2811,7 +2811,7 @@ static int do_pcm_hwsync(struct snd_pcm_substream *substream) case SNDRV_PCM_STATE_DRAINING: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) return -EBADFD; - /* Fall through */ + fallthrough; case SNDRV_PCM_STATE_RUNNING: return snd_pcm_update_hw_ptr(substream); case SNDRV_PCM_STATE_PREPARED: @@ -3816,7 +3816,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) case SNDRV_PCM_MMAP_OFFSET_STATUS_OLD: if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT)) return -ENXIO; - /* fallthrough */ + fallthrough; case SNDRV_PCM_MMAP_OFFSET_STATUS_NEW: if (!pcm_status_mmap_allowed(pcm_file)) return -ENXIO; @@ -3824,7 +3824,7 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) case SNDRV_PCM_MMAP_OFFSET_CONTROL_OLD: if (pcm_file->no_compat_mmap || !IS_ENABLED(CONFIG_64BIT)) return -ENXIO; - /* fallthrough */ + fallthrough; case SNDRV_PCM_MMAP_OFFSET_CONTROL_NEW: if (!pcm_control_mmap_allowed(pcm_file)) return -ENXIO; diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c index a35d429e4c27..f9f57232a83f 100644 --- a/sound/core/seq/oss/seq_oss_timer.c +++ b/sound/core/seq/oss/seq_oss_timer.c @@ -79,7 +79,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev) case TMR_WAIT_REL: parm += rec->cur_tick; rec->realtime = 0; - /* fall through */ + fallthrough; case TMR_WAIT_ABS: if (parm == 0) { rec->realtime = 1; diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index 198f285594e3..81d2ef5e5811 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -309,7 +309,7 @@ do_control(const struct snd_midi_op *ops, void *drv, break; case MIDI_CTL_MSB_DATA_ENTRY: chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0; - /* fall through */ + fallthrough; case MIDI_CTL_LSB_DATA_ENTRY: if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED) rpn(ops, drv, chan, chset); diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 2f6e8023e05c..eb23c55323ae 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -354,7 +354,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) instr_4op = 1; break; } - /* fall through */ + fallthrough; default: spin_unlock_irqrestore(&opl3->voice_lock, flags); return; @@ -443,7 +443,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) switch (connection) { case 0x03: snd_opl3_calc_volume(&vol_op[2], vel, chan); - /* fallthru */ + fallthrough; case 0x02: snd_opl3_calc_volume(&vol_op[0], vel, chan); break; diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index ce409e75ae51..65f9f46c9f58 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -247,7 +247,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) break; case 2: irq[n] = 9; - /* Fall through */ + fallthrough; case 9: wss_config[n] |= WSS_CONFIG_IRQ_9; break; @@ -292,7 +292,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) case 1: if (dma1[n] == 0) break; - /* Fall through */ + fallthrough; default: dev_err(dev, "invalid capture DMA %d\n", dma2[n]); return 0; @@ -322,7 +322,7 @@ static int snd_galaxy_match(struct device *dev, unsigned int n) break; case 2: mpu_irq[n] = 9; - /* Fall through */ + fallthrough; case 9: config[n] |= GALAXY_CONFIG_MPUIRQ_2; break; diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c index 02c566fca9e5..287ab740dd16 100644 --- a/sound/isa/msnd/msnd_pinnacle_mixer.c +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -219,8 +219,7 @@ static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right) case MSND_MIXER_VOLUME: /* master volume */ writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft); writew(wRight, dev->SMA + SMA_wCurrMastVolRight); - /* fall through */ - + fallthrough; case MSND_MIXER_AUX: /* aux pot control */ /* scaled by master volume */ /* fall through */ diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b039429e6871..44ed1b65f6ce 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -163,13 +163,13 @@ static int aci_busy_wait(struct snd_miro_aci *aci) switch (timeout-ACI_MINTIME) { case 0 ... 9: out /= 10; - /* fall through */ + fallthrough; case 10 ... 19: out /= 10; - /* fall through */ + fallthrough; case 20 ... 30: out /= 10; - /* fall through */ + fallthrough; default: set_current_state(TASK_UNINTERRUPTIBLE); schedule_timeout(out); @@ -824,7 +824,7 @@ static unsigned char snd_miro_read(struct snd_miro *chip, retval = inb(chip->mc_base + 9); break; } - /* fall through */ + fallthrough; case OPTi9XX_HW_82C929: retval = inb(chip->mc_base + reg); @@ -854,7 +854,7 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } - /* fall through */ + fallthrough; case OPTi9XX_HW_82C929: outb(value, chip->mc_base + reg); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 0e6d20e49158..881d3b5711d2 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -249,7 +249,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, retval = inb(chip->mc_base + 9); break; } - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -292,7 +292,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, outb(value, chip->mc_base + 9); break; } - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: @@ -343,7 +343,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc); /* enable wave audio */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); - /* Fall through */ + fallthrough; case OPTi9XX_HW_82C925: /* enable WSS mode */ @@ -380,7 +380,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, case OPTi9XX_HW_82C931: /* disable 3D sound (set GPIO1 as output, low) */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c); - /* fall through */ + fallthrough; + case OPTi9XX_HW_82C933: /* * The BTC 1817DW has QS1000 wavetable which is connected @@ -392,7 +393,8 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, * or digital input signal. */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); - /* fall through */ + fallthrough; + case OPTi9XX_HW_82C930: snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x03); snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0x00, 0xff); diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index e33dfe165276..86d0d2fdf48a 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -116,13 +116,13 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_20: chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; break; @@ -261,7 +261,7 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } - /* fall through */ + fallthrough; case SB_HW_20: chip->capture_format = SB_DSP_LO_INPUT_AUTO; break; @@ -361,7 +361,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ if (chip->hardware != SB_HW_JAZZ16) break; - /* fall through */ + fallthrough; case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; if (chip->playback_format == SB_DSP_OUTPUT) @@ -371,7 +371,7 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) case SB_MODE_CAPTURE_16: if (chip->hardware != SB_HW_JAZZ16) break; - /* fall through */ + fallthrough; case SB_MODE_CAPTURE_8: substream = chip->capture_substream; if (chip->capture_format == SB_DSP_INPUT) diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 823ccfa089b2..81c6a9830727 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -1449,7 +1449,7 @@ static int FalconMixerIoctl(u_int cmd, u_long arg) tt_dmasnd.input_gain = RECLEVEL_VOXWARE_TO_GAIN(data & 0xff) << 4 | RECLEVEL_VOXWARE_TO_GAIN(data >> 8 & 0xff); - /* fall through - return set value */ + fallthrough; /* return set value */ case SOUND_MIXER_READ_MIC: return IOCTL_OUT(arg, RECLEVEL_GAIN_TO_VOXWARE(tt_dmasnd.input_gain >> 4 & 0xf) | diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index f802ea331e24..38f25e97538f 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -1478,13 +1478,13 @@ static int dmasound_setup(char *str) printk("dmasound_setup: invalid catch radius, using default = %d\n", catchRadius); else catchRadius = ints[3]; - /* fall through */ + fallthrough; case 2: if (ints[1] < MIN_BUFFERS) printk("dmasound_setup: invalid number of buffers, using default = %d\n", numWriteBufs); else numWriteBufs = ints[1]; - /* fall through */ + fallthrough; case 1: if ((size = ints[2]) < 256) /* check for small buffer specs */ size <<= 10 ; diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6758c072000e..012a7ee849e8 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -218,11 +218,11 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg) case AC97_ID_ST_AC97_ID4: if (reg == 0x08) return 0; - /* fall through */ + fallthrough; case AC97_ID_ST7597: if (reg == 0x22 || reg == 0x7a) return 1; - /* fall through */ + fallthrough; case AC97_ID_AK4540: case AC97_ID_AK4542: if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c) diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 85d3b4e95489..a25d75455802 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -896,15 +896,15 @@ static int snd_atiixp_playback_prepare(struct snd_pcm_substream *substream) case 8: data |= ATI_REG_OUT_DMA_SLOT_BIT(10) | ATI_REG_OUT_DMA_SLOT_BIT(11); - /* fall through */ + fallthrough; case 6: data |= ATI_REG_OUT_DMA_SLOT_BIT(7) | ATI_REG_OUT_DMA_SLOT_BIT(8); - /* fall through */ + fallthrough; case 4: data |= ATI_REG_OUT_DMA_SLOT_BIT(6) | ATI_REG_OUT_DMA_SLOT_BIT(9); - /* fall through */ + fallthrough; default: data |= ATI_REG_OUT_DMA_SLOT_BIT(3) | ATI_REG_OUT_DMA_SLOT_BIT(4); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 58167d8469e1..77c7030ebbfa 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1232,7 +1232,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec, case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break; default: snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); - /* fall-through */ + fallthrough; case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break; case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break; case AZF_FREQ_66200: freq = SOUNDFORMAT_FREQ_SUSPECTED_66200; break; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 0941a7a17623..6c285380b150 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -690,7 +690,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream) break; case SNDRV_PCM_FORMAT_S32_BE: format.data_are_bigendian = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_FORMAT_S32_LE: format.bits_per_sample = 32; break; @@ -742,7 +742,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) pipe->last_counter = 0; pipe->position = 0; *pipe->dma_counter = 0; - /* fall through */ + fallthrough; case PIPE_STATE_PAUSED: pipe->state = PIPE_STATE_STARTED; break; diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index b934c6ac52dd..b2ddabb99438 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -753,7 +753,7 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); /* do we need this? */ snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[0]); - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) @@ -902,8 +902,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream, snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[i]); } snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); - - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL); diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 841523f6b88d..53a2b89f8983 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -102,7 +102,7 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, case SND_BELL: if (hz) hz = 1000; - /* fallthru */ + fallthrough; case SND_TONE: if (beep->linear_tone) beep->tone = beep_linear_tone(beep, hz); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 396b5503038a..9002888670f9 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1014,7 +1014,7 @@ static int patch_conexant_auto(struct hda_codec *codec) break; case 0x14f150f2: codec->power_save_node = 1; - /* Fall through */ + fallthrough; default: codec->pin_amp_workaround = 1; snd_hda_pick_fixup(codec, cxt5066_fixup_models, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d73f8beadb6..fcd303b696e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -374,7 +374,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0295: case 0x10ec0299: alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); - /* fallthrough */ + fallthrough; case 0x10ec0215: case 0x10ec0233: case 0x10ec0235: @@ -4709,7 +4709,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, break; case 0x10ec0867: alc_update_coefex_idx(codec, 0x57, 0x5, 0, 1<<14); - /* fallthru */ + fallthrough; case 0x10ec0221: case 0x10ec0662: snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 81929063b2fa..1d2a0287284b 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -691,7 +691,7 @@ static int snd_ice1712_delta_init(struct snd_ice1712 *ice) break; case ICE1712_SUBDEVICE_DELTADIO2496: ice->gpio.set_pro_rate = delta_1010_set_rate_val; - /* fall thru */ + fallthrough; case ICE1712_SUBDEVICE_DELTA66: ice->spdif.ops.open = delta_open_spdif; ice->spdif.ops.setup_rate = delta_setup_spdif; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 1781a1c081c3..675bb0da00a0 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -810,7 +810,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: val = ICH_IOCE | ICH_STARTBM; @@ -818,7 +818,7 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: val = 0; break; @@ -852,7 +852,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: ichdev->suspended = 0; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -869,7 +869,7 @@ static int snd_intel8x0_ali_trigger(struct snd_pcm_substream *substream, int cmd break; case SNDRV_PCM_TRIGGER_SUSPEND: ichdev->suspended = 1; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* pause */ diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 7ba487443c7f..efff220b26ea 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -169,7 +169,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr, case PIPE_RUNNING: if(rate != 0) break; - /* fall through */ + fallthrough; default: if(rate == 0) return 0; /* nothing to do */ diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 048a2660d18d..0bdd33b0af65 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -527,7 +527,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) dev_err(&mgr->pci->dev, "canceled notification %x !\n", msg); } - /* fall through */ + fallthrough; case MSG_TYPE_ANSWER: /* answer or notification to a message we are waiting for*/ mutex_lock(&mgr->msg_lock); diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index ebce4d259e06..240195e85e3d 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -560,7 +560,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: s->suspended = 0; - /* fallthru */ + fallthrough; case SNDRV_PCM_TRIGGER_START: if (! s->running) { snd_nm256_playback_start(chip, s, substream); @@ -569,7 +569,7 @@ snd_nm256_playback_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: s->suspended = 1; - /* fallthru */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: if (s->running) { snd_nm256_playback_stop(chip); @@ -1664,7 +1664,7 @@ static int snd_nm256_probe(struct pci_dev *pci, return -ENODEV; case NM_RESET_WORKAROUND_2: reset_workaround_2 = 1; - /* Fall-through */ + fallthrough; case NM_RESET_WORKAROUND: reset_workaround = 1; break; diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index db7d76a3cfeb..cacc6a9d14c8 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -400,7 +400,7 @@ static int snd_ymfpci_playback_trigger(struct snd_pcm_substream *substream, kctl = chip->pcm_mixer[substream->number].ctl; kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; } - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: chip->ctrl_playback[ypcm->voices[0]->number + 1] = 0; diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 6cbe5cb34358..dfc4295b69c4 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -45,7 +45,7 @@ static int pdacf_pcm_trigger(struct snd_pcm_substream *subs, int cmd) case SNDRV_PCM_TRIGGER_START: chip->pcm_hwptr = 0; chip->pcm_tdone = 0; - /* fall thru */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: mask = 0; @@ -132,7 +132,7 @@ static int pdacf_pcm_prepare(struct snd_pcm_substream *subs) case SNDRV_PCM_FORMAT_S24_3LE: case SNDRV_PCM_FORMAT_S24_3BE: chip->pcm_sample = 3; - /* fall through */ + fallthrough; default: /* 24-bit */ aval = AK4117_DIF_24R; chip->pcm_frame = 3; diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index cf7049999261..17b79d45d17f 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -580,16 +580,16 @@ static __u32 reverse_bytes(__u32 b, int len) switch (len) { case 32: b = ((b & 0xffff0000) >> 16) | ((b & 0x0000ffff) << 16); - /* fall through */ + fallthrough; case 16: b = ((b & 0xff00ff00) >> 8) | ((b & 0x00ff00ff) << 8); - /* fall through */ + fallthrough; case 8: b = ((b & 0xf0f0f0f0) >> 4) | ((b & 0x0f0f0f0f) << 4); - /* fall through */ + fallthrough; case 4: b = ((b & 0xcccccccc) >> 2) | ((b & 0x33333333) << 2); - /* fall through */ + fallthrough; case 2: b = ((b & 0xaaaaaaaa) >> 1) | ((b & 0x55555555) << 1); case 1: diff --git a/sound/usb/card.c b/sound/usb/card.c index 162bdd6eb4d4..696e788c5d31 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -230,7 +230,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) dev_warn(&dev->dev, "unknown interface protocol %#02x, assuming v1\n", protocol); - /* fall through */ + fallthrough; case UAC_VERSION_1: { struct uac1_ac_header_descriptor *h1; diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b118cf97607f..f3ca59005d91 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -670,7 +670,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, else return 0; } - /* fall through */ + fallthrough; case UAC_VERSION_2: return set_sample_rate_v2v3(chip, iface, alts, fmt, rate); } diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 8a05dcb1344f..6282c539bbb4 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1741,7 +1741,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea switch (cmd) { case SNDRV_PCM_TRIGGER_START: subs->trigger_tstamp_pending_update = true; - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: subs->data_endpoint->prepare_data_urb = prepare_playback_urb; subs->data_endpoint->retire_data_urb = retire_playback_urb; diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 15296f2c902c..4d1e6579e54d 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1146,9 +1146,8 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, dev_dbg(&dev->dev, "%u:%d: unknown interface protocol %#02x, assuming v1\n", iface_no, altno, protocol); protocol = UAC_VERSION_1; - /* fall through */ + fallthrough; case UAC_VERSION_1: - /* fall through */ case UAC_VERSION_2: { int bm_quirk = 0;

2 1

[PATCH 00/28] Clean-up ASoC's W=1 build warnings
by Lee Jones 07 Jul '20

07 Jul '20

This set is part of a larger effort attempting to clean-up W=1 kernel builds, which are currently overwhelmingly riddled with niggly little warnings. After these patches are applied, the build system no longer complains about any W=0 nor W=1 level warnings in sound/soc. Hurrah! Lee Jones (28): ASoC: atmel: atmel-pcm-dma: Demote function header which is clearly not kerneldoc ASoC: atmel: atmel_ssc_dai: Demote function header which is clearly not kerneldoc ASoC: fsl: fsl-asoc-card: Use correct format when providing struct documentation ASoC: fsl: fsl_asrc: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_ssi: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_ssi_dbg: Demote obvious misuse of kerneldoc to standard comment blocks ASoC: fsl: fsl_spdif: Use correct formatting when denoting struct documentation ASoC: fsl: fsl_spdif: Update 'struct fsl_spdif_priv's descriptions ASoC: fsl: fsl_esai: Fix a bunch of kerneldoc issues ASoC: fsl: fsl_esai: Add descriptions for 'tx_mask' and 'rx_mask' to 'fsl_esai' ASoC: pxa: pxa-ssp: Demote seemingly unintentional kerneldoc header ASoC: codecs: ak4458: Remove set but never checked variable 'ret' ASoC: qcom: qdsp6: q6asm: Provide documentation for WMA 'codec_profile' ASoC: samsung: pcm: Demote half-documented kerneldoc header ASoC: samsung: spdif: Fix 3 misspelled struct properties ASoC: sunxi: sun4i-i2s: Demote half-documented kerneldoc header ASoC: uniphier: aio-core: Fix incorrectly named struct property and remove another ASoC: sunxi: sun4i-spdif: Fix misspelling of 'reg_dac_txdata' ASoC: ux500: ux500_msp_i2s: Remove unused variables 'reg_val_DR' and 'reg_val_TSTDR' ASoC: sti: uniperif: Mark 'uni_tdm_hw' as __maybe_unused ASoC: ti: omap-mcbsp-st: Remove set, but unused variable 'w' ASoC: codecs: da7210: Fix formatting issues in da7210_set_dai_pll() header ASoC: soc-ac97: Demote seemingly unintentional kerneldoc header ASoC: codecs: jz4770: Remove defined but never used variable 'mic_boost_tlv' ASoC: codecs: rt5631: Demote misuse of kerneldoc to standard comment blocks ASoC: codecs: rt5659: Remove many unused const variables ASoC: codecs: tlv320aic26: Demote seemingly unintentional kerneldoc header ASoC: codecs: wm8400: Remove a bunch of unused variables sound/soc/atmel/atmel-pcm-dma.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/ak4458.c | 6 +++--- sound/soc/codecs/da7210.c | 12 ++++++----- sound/soc/codecs/jz4770.c | 1 - sound/soc/codecs/rt5631.c | 12 +++++------ sound/soc/codecs/rt5659.c | 37 --------------------------------- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/wm8400.c | 12 ----------- sound/soc/fsl/fsl-asoc-card.c | 16 +++++++------- sound/soc/fsl/fsl_asrc.c | 28 ++++++++++++------------- sound/soc/fsl/fsl_esai.c | 19 ++++++++++------- sound/soc/fsl/fsl_spdif.c | 5 +++-- sound/soc/fsl/fsl_ssi.c | 28 ++++++++++++------------- sound/soc/fsl/fsl_ssi_dbg.c | 4 ++-- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/qcom/qdsp6/q6asm.c | 1 + sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/spdif.c | 6 +++--- sound/soc/soc-ac97.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/ti/omap-mcbsp-st.c | 3 +-- sound/soc/uniphier/aio-core.c | 7 +++---- sound/soc/ux500/ux500_msp_i2s.c | 8 +++---- 26 files changed, 89 insertions(+), 134 deletions(-) -- 2.25.1

7 37

[PATCH] ASoC: amd: add ACPI dependency check
by Vijendar Mukunda 07 Jul '20

07 Jul '20

Add ACPI dependency for evaluating DMIC hardware runtime. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index d1faea5..165c93f 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -218,6 +218,7 @@ static int snd_rn_acp_probe(struct pci_dev *pci, ret = -ENODEV; goto de_init; } else if (dmic_acpi_check == ACP_DMIC_AUTO) { +#if defined(CONFIG_ACPI) handle = ACPI_HANDLE(&pci->dev); ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); if (ACPI_FAILURE(ret)) { @@ -228,6 +229,7 @@ static int snd_rn_acp_probe(struct pci_dev *pci, ret = -ENODEV; goto de_init; } +#endif } adata->res = devm_kzalloc(&pci->dev, -- 2.7.4

4 3

Ignored USB-audio implicit feedback in kernel 5.8rc3
by Pavel Hofman 07 Jul '20

07 Jul '20

Hi, Audio analyzer RTX6001 (XMOS-based) is using implicit feedback, yet the feedback is not used by the latest kernel 5.8 rc3 (i.e. already with the latest implicit-feedback patches). Two analyzers on two different PCs are getting clicks in duplex-mode loobpack, one every 10 secs, the other one every 50 secs. While running aplay/arecord loopback I captured USB packets with tcpdump/tshark. The dump contained 931484 IN packets with 48bytes payload, and 684 IN packets with 40 bytes payload. Packet interval 125us (bInterval = 1), 32bits, 2ch => precise samplerate 47,994.1298Hz. Clearly the incoming stream works OK. Yet stream0 file states for Capture: Momentary freq = 48000 Hz (0x6.0000) But ONLY 48bytes packets were sent by the host in OUT packets, i.e. exactly 48kHz output rate. 6 frames OUT/IN difference per second, the incoming FIFO in the device overflows after several seconds. Please may I ask for help with making the driver respect the implicit feedback stream? Best regards, Pavel. 5.8.0-050800-generic #202006282330 SMP Sun Jun 28 23:35:41 UTC 2020 x86_64 GNU/Linux cat /proc/asound/card4/stream0 RTX A/S RTX6001 USB Audio 2.0 at usb-0000:00:13.2-4, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 72 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 2 Endpoint: 1 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 32 Channel map: FL FR Interface 1 Altset 2 Format: S32_LE Channels: 2 Endpoint: 1 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR Interface 1 Altset 3 Format: S16_LE Channels: 2 Endpoint: 1 OUT (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 16 Channel map: FL FR Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 2 Endpoint: 1 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 32 Channel map: FL FR Interface 2 Altset 2 Format: S32_LE Channels: 2 Endpoint: 1 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 24 Channel map: FL FR Interface 2 Altset 3 Format: S16_LE Channels: 2 Endpoint: 1 IN (ASYNC) Rates: 44100, 48000, 88200, 96000, 176400, 192000 Data packet interval: 125 us Bits: 16 Channel map: FL FR lsusb: Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 007 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 006 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 002 Device 002: ID 058f:6362 Alcor Micro Corp. Flash Card Reader/Writer Bus 002 Device 003: ID 0d9a:00df RTX AS RTX6001 USB Audio 2.0 <---- Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 005 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 008: ID 2047:0301 Texas Instruments Arduino Micro Bus 004 Device 007: ID 18f8:0f97 [Maxxter] Elo TouchSystems 2216 AccuTouch® USB Touchmonitor Interface Bus 004 Device 006: ID 413c:2001 Dell Computer Corp. Keyboard HID Support Bus 004 Device 005: ID 413c:1001 Dell Computer Corp. Keyboard Hub Bus 004 Device 004: ID 0bd9:0001 Liberty Instruments, Inc. Liberty Praxis AudPod 1.1 Bus 004 Device 003: ID 2341:8037 Arduino SA Arduino Micro Bus 004 Device 002: ID 04e7:0050 Elo TouchSystems 2216 AccuTouch® Touchmonitor Interface Bus 004 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub Bus 009 Device 002: ID 18a5:0422 Verbatim, Ltd Portable SSD Bus 009 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 008 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub lsusb -v see the attachment

3 8

[PATCH v2] ASoC: Intel: bxt-da7219-max98357a: support MAX98390 speaker amp
by Brent Lu 07 Jul '20

07 Jul '20

Support MAX98390 speaker amplifier on cometlake platform. Driver now detects amplifier type in the probe function and installs corresponding controls and DAPM widgets/routes in the late_probe function. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 117 ++++++++++++++++++++-- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 13 +++ 3 files changed, 127 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 3d820e1..f176df2 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -288,6 +288,7 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC tristate select SND_SOC_DA7219 select SND_SOC_MAX98357A + select SND_SOC_MAX98390 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI @@ -298,14 +299,14 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON if SND_SOC_INTEL_APL config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH - tristate "Broxton with DA7219 and MAX98357A in I2S Mode" + tristate "Broxton with DA7219 and MAX98357A/MAX98390 in I2S Mode" depends on I2C && ACPI && GPIOLIB depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON help This adds support for ASoC machine driver for Broxton-P platforms - with DA7219 + MAX98357A I2S audio codec. + with DA7219 + MAX98357A/MAX98390 I2S audio codec. Say Y or m if you have such a device. This is a recommended option. If unsure select "N". diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 44016c1..4d39253 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -25,9 +25,14 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" +#define MAX98390_DEV0_NAME "i2c-MX98390:00" +#define MAX98390_DEV1_NAME "i2c-MX98390:01" #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 +#define SPKAMP_MAX98357A 1 +#define SPKAMP_MAX98390 2 + static struct snd_soc_jack broxton_headset; static struct snd_soc_jack broxton_hdmi[3]; @@ -40,6 +45,7 @@ struct bxt_hdmi_pcm { struct bxt_card_private { struct list_head hdmi_pcm_list; bool common_hdmi_codec_drv; + int spkamp; }; enum { @@ -85,13 +91,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, static const struct snd_kcontrol_new broxton_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_kcontrol_new max98357a_controls[] = { SOC_DAPM_PIN_SWITCH("Spk"), }; +static const struct snd_kcontrol_new max98390_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + static const struct snd_soc_dapm_widget broxton_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("SoC DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), @@ -100,14 +113,20 @@ static const struct snd_soc_dapm_widget broxton_widgets[] = { platform_clock_control, SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU), }; +static const struct snd_soc_dapm_widget max98357a_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_widget max98390_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + static const struct snd_soc_dapm_route audio_map[] = { /* HP jack connectors - unknown if we have jack detection */ {"Headphone Jack", NULL, "HPL"}, {"Headphone Jack", NULL, "HPR"}, - /* speaker */ - {"Spk", NULL, "Speaker"}, - /* other jacks */ {"MIC", NULL, "Headset Mic"}, @@ -134,6 +153,17 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Headset Mic", NULL, "Platform Clock" }, }; +static const struct snd_soc_dapm_route max98357a_routes[] = { + /* speaker */ + {"Spk", NULL, "Speaker"}, +}; + +static const struct snd_soc_dapm_route max98390_routes[] = { + /* Speaker */ + {"Left Spk", NULL, "Left BE_OUT"}, + {"Right Spk", NULL, "Right BE_OUT"}, +}; + static const struct snd_soc_dapm_route broxton_map[] = { {"HiFi Playback", NULL, "ssp5 Tx"}, {"ssp5 Tx", NULL, "codec0_out"}, @@ -404,6 +434,10 @@ SND_SOC_DAILINK_DEF(ssp5_pin, SND_SOC_DAILINK_DEF(ssp5_codec, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", BXT_MAXIM_CODEC_DAI))); +SND_SOC_DAILINK_DEF(max98390_codec, + DAILINK_COMP_ARRAY( + /* Left */ COMP_CODEC(MAX98390_DEV0_NAME, "max98390-aif1"), + /* Right */ COMP_CODEC(MAX98390_DEV1_NAME, "max98390-aif1"))); SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); @@ -601,15 +635,69 @@ static struct snd_soc_dai_link broxton_dais[] = { }, }; +static struct snd_soc_codec_conf max98390_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(MAX98390_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX98390_DEV1_NAME), + .name_prefix = "Right", + }, +}; + #define NAME_SIZE 32 static int bxt_card_late_probe(struct snd_soc_card *card) { struct bxt_card_private *ctx = snd_soc_card_get_drvdata(card); struct bxt_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; - int err, i = 0; + const struct snd_kcontrol_new *controls; + const struct snd_soc_dapm_widget *widgets; + const struct snd_soc_dapm_route *routes; + int num_controls, num_widgets, num_routes, err, i = 0; char jack_name[NAME_SIZE]; + switch (ctx->spkamp) { + case SPKAMP_MAX98357A: + controls = max98357a_controls; + num_controls = ARRAY_SIZE(max98357a_controls); + widgets = max98357a_widgets; + num_widgets = ARRAY_SIZE(max98357a_widgets); + routes = max98357a_routes; + num_routes = ARRAY_SIZE(max98357a_routes); + break; + case SPKAMP_MAX98390: + controls = max98390_controls; + num_controls = ARRAY_SIZE(max98390_controls); + widgets = max98390_widgets; + num_widgets = ARRAY_SIZE(max98390_widgets); + routes = max98390_routes; + num_routes = ARRAY_SIZE(max98390_routes); + break; + default: + dev_err(card->dev, "Invalid speaker amplifier %d\n", ctx->spkamp); + break; + } + + err = snd_soc_dapm_new_controls(&card->dapm, widgets, num_widgets); + if (err) { + dev_err(card->dev, "Fail to new widgets\n"); + return err; + } + + err = snd_soc_add_card_controls(card, controls, num_controls); + if (err) { + dev_err(card->dev, "Fail to add controls\n"); + return err; + } + + err = snd_soc_dapm_add_routes(&card->dapm, routes, num_routes); + if (err) { + dev_err(card->dev, "Fail to add routes\n"); + return err; + } + if (soc_intel_is_glk()) snd_soc_dapm_add_routes(&card->dapm, gemini_map, ARRAY_SIZE(gemini_map)); @@ -678,6 +766,11 @@ static int broxton_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + if (acpi_dev_present("MX98390", NULL, -1)) + ctx->spkamp = SPKAMP_MAX98390; + else + ctx->spkamp = SPKAMP_MAX98357A; + broxton_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&broxton_audio_card, ctx); if (soc_intel_is_glk()) { @@ -702,7 +795,13 @@ static int broxton_audio_probe(struct platform_device *pdev) } else if (soc_intel_is_cml()) { unsigned int i; - broxton_audio_card.name = "cmlda7219max"; + if (ctx->spkamp == SPKAMP_MAX98390) { + broxton_audio_card.name = "cml_max98390_da7219"; + + broxton_audio_card.codec_conf = max98390_codec_confs; + broxton_audio_card.num_configs = ARRAY_SIZE(max98390_codec_confs); + } else + broxton_audio_card.name = "cmlda7219max"; for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { /* MAXIM_CODEC is connected to SSP1. */ @@ -710,6 +809,11 @@ static int broxton_audio_probe(struct platform_device *pdev) BXT_MAXIM_CODEC_DAI)) { broxton_dais[i].name = "SSP1-Codec"; broxton_dais[i].cpus->dai_name = "SSP1 Pin"; + + if (ctx->spkamp == SPKAMP_MAX98390) { + broxton_dais[i].codecs = max98390_codec; + broxton_dais[i].num_codecs = ARRAY_SIZE(max98390_codec); + } } /* DIALOG_CODEC is connected to SSP0 */ else if (!strcmp(broxton_dais[i].codecs->dai_name, @@ -759,6 +863,7 @@ MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); MODULE_AUTHOR("Conrad Cooke <conrad.cooke(a)intel.com>"); MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); +MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:bxt_da7219_max98357a"); MODULE_ALIAS("platform:glk_da7219_max98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index cdea0c0..dee1f0f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -19,6 +19,11 @@ static struct snd_soc_acpi_codecs max98357a_spk_codecs = { .codecs = {"MX98357A"} }; +static struct snd_soc_acpi_codecs max98390_spk_codecs = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + /* * The order of the three entries with .id = "10EC5682" matters * here, because DSDT tables expose an ACPI HID for the MAX98357A @@ -55,6 +60,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", }, + { + .id = "DLGS7219", + .drv_name = "cml_da7219_max98357a", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &max98390_spk_codecs, + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); -- 2.7.4

4 4

[RFC PATCH 00/23] ALSA: fix kernel-doc and W=1 warnings
by Pierre-Louis Bossart 07 Jul '20

07 Jul '20

This cleanup touches a lot of old - and very old code for hardware designed in the late 1990s/early 2000s -, hence the RFC status. I mostly did dumb edits just to make warnings go away, suggestions and feedback welcome. Pierre-Louis Bossart (23): ALSA: isa/gus: remove -Wmissing-prototypes warnings ALSA: isa/gus: remove 'set but not used' warning ALSA: pci/asihpi: remove 'set but not used' warnings ALSA: pci/asihpi: fix kernel-doc ALSA: pci/asihpi: remove 'set but not used' warning ALSA: pci/echoaudio: remove 'set but not used' warning ALSA: pci/aw2-saa7146: remove 'set but not used' warning ALSA: pci/ctxfi/ctatc: fix kernel-doc ALSA: pci/ctxfi: fix kernel-doc warnings ALSA: usb/line6: remove 'defined but not used' warning ASoC: pci/emu10k1: remove "set but not used' warnings ALSA: firewire: fix kernel-doc ALSA: pci/emu10k1: remove 'set but not used' warning ALSA: pci/korg1212: remove 'set but not used' warnings ALSA: pci/rme9652/hdspm: remove always true condition ALSA: pci/oxygen/xonar_wm87x6: remove always true condition ALSA: xen: remove 'set but not used' warning ALSA: pci/es1938: remove 'set but not used' warning ALSA: pci/fm801: fix kernel-doc ALSA: pci/via82xx: remove 'set but not used' warnings ALSA: pcmcia/pdaudiocf: fix kernel-doc ALSA: vx_core: remove warning for empty loop body ALSA: pci/au88x0: remove "defined but not used" warnings include/sound/gus.h | 4 ++++ sound/drivers/vx/vx_core.c | 3 ++- sound/firewire/cmp.c | 1 - sound/isa/gus/gus_reset.c | 2 -- sound/isa/gus/gus_uart.c | 3 ++- sound/pci/asihpi/asihpi.c | 12 +++++----- sound/pci/asihpi/hpicmn.c | 26 +++++++++++---------- sound/pci/au88x0/au88x0_a3ddata.c | 8 +++---- sound/pci/au88x0/au88x0_xtalk.c | 36 +++++++++++++++--------------- sound/pci/aw2/aw2-saa7146.c | 2 +- sound/pci/ctxfi/ctatc.c | 4 ++++ sound/pci/ctxfi/cthardware.c | 2 +- sound/pci/ctxfi/cthw20k1.c | 2 +- sound/pci/ctxfi/cthw20k2.c | 2 +- sound/pci/ctxfi/ctimap.c | 2 +- sound/pci/ctxfi/ctmixer.c | 2 +- sound/pci/ctxfi/ctpcm.c | 2 +- sound/pci/echoaudio/echoaudio.c | 3 ++- sound/pci/emu10k1/emu10k1_main.c | 4 ++-- sound/pci/emu10k1/emu10k1_patch.c | 3 ++- sound/pci/es1938.c | 3 ++- sound/pci/fm801.c | 27 ++++++++++++++++++++++ sound/pci/korg1212/korg1212.c | 4 +++- sound/pci/oxygen/xonar_wm87x6.c | 6 +++-- sound/pci/rme9652/hdspm.c | 4 ++-- sound/pci/via82xx.c | 2 +- sound/pci/via82xx_modem.c | 2 +- sound/pcmcia/pdaudiocf/pdaudiocf.c | 1 + sound/usb/line6/driver.h | 4 ++-- sound/xen/xen_snd_front.c | 2 +- 30 files changed, 110 insertions(+), 68 deletions(-) -- 2.25.1

3 27

Localization
by GitHub issues - opened 07 Jul '20

07 Jul '20

alsa-project/alsa-tools issue #1 was opened from yasarciv: Will there be translations to other languages? Issue URL : https://github.com/alsa-project/alsa-tools/issues/1 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH v2] MAINTAINERS: Add Shengjiu to reviewer list of sound/soc/fsl
by Nicolin Chen 07 Jul '20

07 Jul '20

Add Shengjiu who's actively working on the latest fsl/nxp audio drivers. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Acked-by: Shengjiu Wang <shengjiu.wang(a)gmail.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> --- Changelog v1->v2: * Replaced Shengjiu's emaill address with his gmail one * Added Acked-by from Shengjiu and Reviewed-by from Fabio MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 496fd4eafb68..ff97b8cefaea 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6956,6 +6956,7 @@ M: Timur Tabi <timur(a)kernel.org> M: Nicolin Chen <nicoleotsuka(a)gmail.com> M: Xiubo Li <Xiubo.Lee(a)gmail.com> R: Fabio Estevam <festevam(a)gmail.com> +R: Shengjiu Wang <shengjiu.wang(a)gmail.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linuxppc-dev(a)lists.ozlabs.org S: Maintained -- 2.17.1

2 1

[PATCH 0/6] ASoC: amd: remove warnings with make W=1
by Pierre-Louis Bossart 07 Jul '20

07 Jul '20

Yet another cleanup series, with good findings of variables not used for anything. Pierre-Louis Bossart (6): ASoC: amd: acp-da7219-max98357a: fix 'defined but not used' warning ASoC: amd: acp-rt5645: fix 'defined but not used' warning ASoC: amd: raven: acp3x-pcm-dma: fix 'set but not used' warning ASoC: amd: raven: acp3x-i2s: fix 'set but not used' warning ASoC: amd: raven: acp3x-pcm-dma: remove unused-but-set variable ASoC: amd: raven: acp3x-i2s: remove unused-but-set variable sound/soc/amd/acp-da7219-max98357a.c | 2 ++ sound/soc/amd/acp-rt5645.c | 2 ++ sound/soc/amd/raven/acp3x-i2s.c | 4 ---- sound/soc/amd/raven/acp3x-pcm-dma.c | 4 ---- 4 files changed, 4 insertions(+), 8 deletions(-) -- 2.25.1

2 7

[PATCH] MAINTAINERS: Add Shengjiu to reviewer list of sound/soc/fsl
by Nicolin Chen 07 Jul '20

07 Jul '20

Add Shengjiu who's actively working on the latest fsl/nxp audio drivers. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- To Shengjiu, please ack if you feel okay with this (your email address too). MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 496fd4eafb68..54aab083bb88 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6956,6 +6956,7 @@ M: Timur Tabi <timur(a)kernel.org> M: Nicolin Chen <nicoleotsuka(a)gmail.com> M: Xiubo Li <Xiubo.Lee(a)gmail.com> R: Fabio Estevam <festevam(a)gmail.com> +R: Shengjiu Wang <shengjiu.wang(a)nxp.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linuxppc-dev(a)lists.ozlabs.org S: Maintained -- 2.17.1

4 3

[PATCH] ASoC: fsl: mpc8610_hpcd: Add missing of_node_put()
by Yi Wang 07 Jul '20

07 Jul '20

From: Liao Pingfang <liao.pingfang(a)zte.com.cn> After finishing using device node got from of_find_compatible_node(), of_node_put() needs to be called. Signed-off-by: Liao Pingfang <liao.pingfang(a)zte.com.cn> --- sound/soc/fsl/mpc8610_hpcd.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index f7bd900..b3090fe 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -426,9 +426,11 @@ static int __init mpc8610_hpcd_init(void) guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); if (of_address_to_resource(guts_np, 0, &res)) { pr_err("mpc8610-hpcd: missing/invalid global utilities node\n"); + of_node_put(guts_np); return -EINVAL; } guts_phys = res.start; + of_node_put(guts_np); return platform_driver_register(&mpc8610_hpcd_driver); } -- 2.9.5

2 1

[PATCH] ASoC: core: Remove only the registered component in devm functions
by Maxime Ripard 07 Jul '20

07 Jul '20

The ASoC devm_ functions that register a component (devm_snd_soc_register_component and devm_snd_dmaengine_pcm_register) will clean their component by running snd_soc_unregister_component. snd_soc_unregister_component will then remove all the components for the device that was used to register the component in the first place. However, some drivers register several components (such as a DAI and a dmaengine PCM) on the same device, and if the dmaengine PCM is registered first, then the DAI will be cleaned up first and snd_dmaengine_pcm_unregister will be called next. snd_dmaengine_pcm_unregister will then lookup the dmaengine PCM component on the device, and if there's one unregister that component and release its dmaengine channels. That doesn't happen in practice though since the first call to snd_soc_unregister_component removed all the components, so we never get the chance to release the dmaengine channels. In order to fix this, instead of removing all the components for a given device, we can simply remove the component that was registered in the first place. We should have the same number of component registration than we have components, so it should work just fine. Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- This was observed on a RaspberryPi that uses the vc4_hdmi driver (drivers/gpu/drm/vc4/vc4_hdmi.c). This driver will register a dmaengine PCM and two components. If the MIPI-DSI controller is enabled, it will create an EPROBE_DEFER across the entire display pipeline, leading to multiple bind/unbind cycles in the other display drivers including vc4_hdmi, leading to multiple warnings since we request the same dmaengine channels on the same device without ever releasing them. It's not really clear to me when that bug was introduced exactly, since it can only be seen on a rather unusual setup, and with all the drivers built-in (otherwise we probably wouldn't get an EPROBE_DEFER for DSI), but it still looks like something that should probably go to stable? Changes from v1: - Rebased on top of next-20200706 --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 27 +++++++++++++++++++++++++++ sound/soc/soc-devres.c | 8 +++++--- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 4 files changed, 35 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6791b7570a67..59235e553630 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -426,6 +426,8 @@ int devm_snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *component_driver, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_component(struct device *dev); +void snd_soc_unregister_component_by_driver(struct device *dev, + const struct snd_soc_component_driver *component_driver); struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev, const char *driver_name); struct snd_soc_component *snd_soc_lookup_component(struct device *dev, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index adedadcb0efb..7c58e45c1c3f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2513,6 +2513,33 @@ int snd_soc_register_component(struct device *dev, } EXPORT_SYMBOL_GPL(snd_soc_register_component); +/** + * snd_soc_unregister_component_by_driver - Unregister component using a given driver + * from the ASoC core + * + * @dev: The device to unregister + * @component_driver: The component driver to unregister + */ +void snd_soc_unregister_component_by_driver(struct device *dev, + const struct snd_soc_component_driver *component_driver) +{ + struct snd_soc_component *component; + + if (!component_driver) + return; + + mutex_lock(&client_mutex); + component = snd_soc_lookup_component_nolocked(dev, component_driver->name); + if (!component) + goto out; + + snd_soc_del_component_unlocked(component); + +out: + mutex_unlock(&client_mutex); +} +EXPORT_SYMBOL_GPL(snd_soc_unregister_component_by_driver); + /** * snd_soc_unregister_component - Unregister all related component * from the ASoC core diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 11e5d7962370..4534a1c03e8e 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -48,7 +48,9 @@ EXPORT_SYMBOL_GPL(devm_snd_soc_register_dai); static void devm_component_release(struct device *dev, void *res) { - snd_soc_unregister_component(*(struct device **)res); + const struct snd_soc_component_driver **cmpnt_drv = res; + + snd_soc_unregister_component_by_driver(dev, *cmpnt_drv); } /** @@ -65,7 +67,7 @@ int devm_snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai) { - struct device **ptr; + const struct snd_soc_component_driver **ptr; int ret; ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL); @@ -74,7 +76,7 @@ int devm_snd_soc_register_component(struct device *dev, ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai); if (ret == 0) { - *ptr = dev; + *ptr = cmpnt_drv; devres_add(dev, ptr); } else { devres_free(ptr); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 80a4e71f2d95..61844403f181 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -478,7 +478,7 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_component_to_pcm(component); - snd_soc_unregister_component(dev); + snd_soc_unregister_component_by_driver(dev, component->driver); dmaengine_pcm_release_chan(pcm); kfree(pcm); } -- 2.26.2

2 1

[PATCHv3 0/5] ASoC: da7213: support for usage with simple-card
by Sebastian Reichel 07 Jul '20

07 Jul '20

Hi, This extends the da7213 driver to be used with simple-audio-card in combination with a fixed clock. Here is a snippet of the downstream board's DT, that is supposed to be supported by this patchset. --------------------------------------------------------------------- / { sound { compatible = "simple-audio-card"; simple-audio-card,name = "audio-card"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&dailink_master>; simple-audio-card,frame-master = <&dailink_master>; simple-audio-card,widgets = "Speaker", "Ext Spk"; simple-audio-card,audio-routing = "Ext Spk", "LINE"; simple-audio-card,cpu { sound-dai = <&ssi1>; }; dailink_master: simple-audio-card,codec { sound-dai = <&codec>; }; }; clk_ext_audio_codec: clock-codec { compatible = "fixed-clock"; #clock-cells = <0>; clock-frequency = <12288000>; }; }; &i2c1 { codec: audio-codec@1a { compatible = "dlg,da7212"; reg = <0x1a>; #sound-dai-cells = <0>; VDDA-supply = <&reg_2v5_audio>; VDDSP-supply = <&reg_5v0_audio>; VDDMIC-supply = <&reg_3v3_audio>; VDDIO-supply = <&reg_3v3_audio>; clocks = <&clk_ext_audio_codec>; clock-names = "mclk"; }; }; --------------------------------------------------------------------- This is mostly a resend of PATCHv2. There was quite a bit of discussion for the last patch. As far as I can see no better solution was found and the patch is fine as is. Sorry for the delay in sending another revision. Changes since PATCHv2: * https://lore.kernel.org/alsa-devel/20191120152406.2744-1-sebastian.reichel@… * dropped patch converting DA7213 into selectable Kconfig option (merged) * fix compatible string in patch 1 (DT binding) as pointed out by Adam Thomson * collected Reviewed-by from Adam Thomson for patch 2-4 Changes since PATCHv1: * https://lore.kernel.org/alsa-devel/20191108174843.11227-1-sebastian.reichel… * add patch adding da7212 compatible to DT bindings * update regulator patch, so that VDDA is enabled together with VDDIO while the device is enabled to avoid device reset * update clock patch, so that automatic PLL handling is not enabled when PLL is configured manually * update clock patch, so that automatic PLL is disabled when the device is suspended * update clock patch, so that automatic PLL is configured into bypass mode when possible -- Sebastian Sebastian Reichel (5): ASoC: da7213: Add da7212 DT compatible ASoC: da7213: Add regulator support ASoC: da7213: move set_sysclk to codec level ASoC: da7213: move set_pll to codec level ASoC: da7213: add default clock handling .../devicetree/bindings/sound/da7213.txt | 8 +- sound/soc/codecs/da7213.c | 172 ++++++++++++++++-- sound/soc/codecs/da7213.h | 11 ++ 3 files changed, 175 insertions(+), 16 deletions(-) -- 2.26.2

3 10

[PATCH v3 0/2] Introduce multi PM domains helpers
by Daniel Baluta 07 Jul '20

07 Jul '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> i.MX8QXP/i.MX8QM has IPs that need multiple power domains to be up in order to work. In order to help drivers, we introduce multi PM domains helpers that are able to activate/deactivate multi PM domains. First patch introduces the helpers and second patch demonstrates how a driver can use them instead of hardcoding the PM domains handling. Changes since v2: » - fix kernel test robot reported issues (missing static inline for newly introduced functions in headers and arguments swapped for devm_kzalloc). Changes since v1: (addressed Ranjani's comments) » - enhanced description for dev_multi_pm_attach return value » - renamed exit_unroll_pm label to exit_detach_pm Ideally would be to have patch 1/2 merged via power tree and then I will submit again patch 2/2 on sound tree. Daniel Baluta (2): PM / domains: Introduce multi PM domains helpers ASoC: SOF: Use multi PM domains helpers drivers/base/power/common.c | 93 +++++++++++++++++++++++++++++++++++++ include/linux/pm_domain.h | 19 ++++++++ sound/soc/sof/imx/imx8.c | 60 ++++-------------------- 3 files changed, 121 insertions(+), 51 deletions(-) -- 2.17.1

2 3

[PATCH] ALSA: pcm: Fix build error on m68k and others
by Takashi Iwai 07 Jul '20

07 Jul '20

The commit 3ad796cbc36a ("ALSA: pcm: Use SG-buffer only when direct DMA is available") introduced a check of the DMA type and this caused a build error on m68k (and possibly some others) due to the lack of dma_is_direct() definition. Since the check is needed only for CONFIG_SND_DMA_SGBUF enablement (i.e. solely x86), use #ifdef instead of IS_ENABLED() for avoiding such a build error. Fixes: 3ad796cbc36a ("ALSA: pcm: Use SG-buffer only when direct DMA is available") Reported-by: Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_memory.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 8326d16d3596..795af1b88051 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -41,8 +41,9 @@ static int do_alloc_pages(struct snd_card *card, int type, struct device *dev, card->total_pcm_alloc_bytes + size > max_alloc_per_card) return -ENOMEM; - if (IS_ENABLED(CONFIG_SND_DMA_SGBUF) && - (type == SNDRV_DMA_TYPE_DEV_SG || type == SNDRV_DMA_TYPE_DEV_UC_SG) && + +#ifdef CONFIG_SND_DMA_SGBUF + if ((type == SNDRV_DMA_TYPE_DEV_SG || type == SNDRV_DMA_TYPE_DEV_UC_SG) && !dma_is_direct(get_dma_ops(dev))) { /* mutate to continuous page allocation */ dev_dbg(dev, "Use continuous page allocator\n"); @@ -51,6 +52,7 @@ static int do_alloc_pages(struct snd_card *card, int type, struct device *dev, else type = SNDRV_DMA_TYPE_DEV_UC; } +#endif /* CONFIG_SND_DMA_SGBUF */ err = snd_dma_alloc_pages(type, dev, size, dmab); if (!err) { -- 2.16.4

1 1

[PATCH 0/8] ASoC: fix W=1 warnings for various SOCs
by Pierre-Louis Bossart 07 Jul '20

07 Jul '20

One more series of trivial fixes for atmel, samsung, qualcomm, sunxi, tegra, uniphier. Pierre-Louis Bossart (8): ASoC: atmel: fix kernel-doc ASoC: samsung: spdif: fix kernel-doc ASoC: samsung: pcm: fix kernel-doc ASoC: qcom: q6asm: fix kernel-doc ASoC: sunxi: sun4i-i2s: fix kernel-doc ASoC: sunxi: sun4i-spdif: fix kernel-doc ASoC: tegra: tegra20_das: remove always-true comparison ASoC: uniphier: aio-core: fix kernel-doc sound/soc/atmel/atmel-pcm-dma.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 1 + sound/soc/qcom/qdsp6/q6asm.c | 2 +- sound/soc/samsung/pcm.c | 5 +++++ sound/soc/samsung/spdif.c | 6 +++--- sound/soc/sunxi/sun4i-i2s.c | 10 +++++++++- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/tegra/tegra20_das.c | 3 +-- sound/soc/uniphier/aio-core.c | 7 +++---- 9 files changed, 25 insertions(+), 13 deletions(-) -- 2.25.1

5 14

Re: mmotm 2020-07-06-18-53 uploaded (sound/soc/amd/renoir/rn-pci-acp3x.c:)
by Randy Dunlap 07 Jul '20

07 Jul '20

On 7/6/20 6:53 PM, Andrew Morton wrote: > The mm-of-the-moment snapshot 2020-07-06-18-53 has been uploaded to > > http://www.ozlabs.org/~akpm/mmotm/ > > mmotm-readme.txt says > > README for mm-of-the-moment: > > http://www.ozlabs.org/~akpm/mmotm/ > > This is a snapshot of my -mm patch queue. Uploaded at random hopefully > more than once a week. > > You will need quilt to apply these patches to the latest Linus release (5.x > or 5.x-rcY). The series file is in broken-out.tar.gz and is duplicated in > http://ozlabs.org/~akpm/mmotm/series > on i386: when CONFIG_ACPI is not set/enabled: ../sound/soc/amd/renoir/rn-pci-acp3x.c: In function ‘snd_rn_acp_probe’: ../sound/soc/amd/renoir/rn-pci-acp3x.c:222:9: error: implicit declaration of function ‘acpi_evaluate_integer’; did you mean ‘acpi_evaluate_object’? [-Werror=implicit-function-declaration] ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); ^~~~~~~~~~~~~~~~~~~~~ acpi_evaluate_object -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

2 3

[PATCH v3] ALSA: hda/hdmi: Add Intel silent stream support
by Harsha Priya 07 Jul '20

07 Jul '20

External HDMI receivers have analog circuitry that needs to be powered-on when exiting standby, and a mechanism to detect PCM v. IEC61937 data. These two steps take time and up to 2-3 seconds of audio may be muted when starting playback. Intel hardware (Haswell and beyond) can keep the link active with a 'silent stream', so that the receiver does not go through those two steps when valid audio is transmitted. This mechanism relies on an setting the channel_id as 0xf, sending info packet and preventing the codec from going to D3, which will increase the platform static power consumption. The info packet assumes a basic 2ch stereo, and the silent stream is enabled when connecting a monitor. In case of format changes the detection of PCM v. IEC61937 needs to be re-run. In this case there is no way to avoid the 2-3s mute. The silent stream is enabled with a Kconfig option, as well as a kernel parameter should there be a need to override the build time default. This approach is used based on the power_save capability as an example, but in the future, it may be used with a kcontrol, depending on UCM support for HDaudio legacy. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Emmanuel Jillela <emmanuel.jillela(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reported-by: kernel test robot <lkp(a)intel.com> --- Change History: v3: - Removed eld_valid check along side of silent stream enable check as its redundant with the monitor_present variable - Limited the change to take affect only for Haswell and beyond Intel platforms - As per the spec, setting the channel_id as 0xf before sending info packet - Fixed issue reported by Kernel Test Robot (tag added) - Tested on Intel Cometlake based Chromebook connected to few different monitors and docks. v2: - Kconfig symbol depends on SND_HDA_INTEL. - Added valid eld check to along side of silent stream enable check, under which silent stream is enabled. - Error prints added for codec power up, power down functions. - The silent stream is enabled with a Kconfig/module parameter following the power_save capability as an example. But in the future, it may be used with a kcontrol, depending on UCM support for HDaudio legacy. v1: - Initial version. sound/pci/hda/Kconfig | 14 ++++++++++ sound/pci/hda/patch_hdmi.c | 64 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 78 insertions(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 3f9abda..bf213e5 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -239,6 +239,20 @@ config SND_HDA_POWER_SAVE_DEFAULT The default time-out value in seconds for HD-audio automatic power-save mode. 0 means to disable the power-save mode. +config SND_HDA_INTEL_HDMI_SILENT_STREAM + bool "Enable Silent Stream always for HDMI" + depends on SND_HDA_INTEL + help + Intel hardware has a feature called 'silent stream', that + keeps external HDMI receiver's analog circuitry powered on + avoiding 2-3 sec silence during playback start. This mechanism + relies on setting channel_id as 0xf, sending info packet and + preventing codec D3 entry (increasing platform static power + consumption when HDMI receiver is plugged-in). 2-3 sec silence + at the playback start is expected whenever there is format change. + (default is 2 channel format). + Say Y to enable Silent Stream feature. + endif endmenu diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e2b21ef..18eabcf 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -42,6 +42,11 @@ static bool enable_acomp = true; module_param(enable_acomp, bool, 0444); MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)"); +static bool enable_silent_stream = +IS_ENABLED(CONFIG_SND_HDA_INTEL_HDMI_SILENT_STREAM); +module_param(enable_silent_stream, bool, 0644); +MODULE_PARM_DESC(enable_silent_stream, "Enable Silent Stream for HDMI devices"); + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -167,6 +172,7 @@ struct hdmi_spec { hda_nid_t vendor_nid; const int *port_map; int port_num; + bool send_silent_stream; /* Flag to enable silent stream feature */ }; #ifdef CONFIG_SND_HDA_COMPONENT @@ -1634,21 +1640,72 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, snd_hda_power_down_pm(codec); } +static void silent_stream_enable(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + unsigned int newval, oldval; + + codec_dbg(codec, "hdmi: enabling silent stream for NID %d\n", + per_pin->pin_nid); + + mutex_lock(&per_pin->lock); + + if (!per_pin->channels) + per_pin->channels = 2; + + oldval = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + AC_VERB_GET_CONV, 0); + newval = (oldval & 0xF0) | 0xF; + snd_hda_codec_write(codec, per_pin->pin_nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, newval); + + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); + + mutex_unlock(&per_pin->lock); +} + /* update ELD and jack state via audio component */ static void sync_eld_via_acomp(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin) { struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; + bool monitor_prev, monitor_next; mutex_lock(&per_pin->lock); eld->monitor_present = false; + monitor_prev = per_pin->sink_eld.monitor_present; eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, per_pin->dev_id, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); eld->eld_valid = (eld->eld_size > 0); update_eld(codec, per_pin, eld, 0); + monitor_next = per_pin->sink_eld.monitor_present; mutex_unlock(&per_pin->lock); + + /* + * Power-up will call hdmi_present_sense, so the PM calls + * have to be done without mutex held. + */ + + if (spec->send_silent_stream) { + int pm_ret; + + if (!monitor_prev && monitor_next) { + pm_ret = snd_hda_power_up_pm(codec); + if (pm_ret < 0) + codec_err(codec, + "Monitor plugged-in, Failed to power up codec ret=[%d]\n", + pm_ret); + silent_stream_enable(codec, per_pin); + } else if (monitor_prev && !monitor_next) { + pm_ret = snd_hda_power_down_pm(codec); + if (pm_ret < 0) + codec_err(codec, + "Monitor plugged-out, Failed to power down codec ret=[%d]\n", + pm_ret); + } + } } static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) @@ -2791,6 +2848,13 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, spec->ops.setup_stream = i915_hsw_setup_stream; spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup; + /* + * Enable silent stream feature, if it is enabled via + * module param or Kconfig option + */ + if (enable_silent_stream) + spec->send_silent_stream = true; + return parse_intel_hdmi(codec); } -- 2.7.4

2 1

[PATCH alsa-lib v2 0/2] Make dmix lockless operation optional
by Takashi Iwai 07 Jul '20

07 Jul '20

Hi, this is a revised patch set for the PCM dmix, to make the lockless operation optional and fix the semaphore usage in the lockless mode. Unfortunately I got a server problem in the last week and lost many posts during the weekend, so I couldn't follow up to the thread (I just read from the lore archive). In this patch set, to make my intention clearer, I swapped the patches: the first to disable the lockless operation as default and the second to fix the semaphore usage in the lockless mode. In the first patch, the magic number is changed as Jaroslav suggested, and the description has been updated. The second patch is equivalent with the previous one. thanks, Takashi === Takashi Iwai (2): pcm: dmix: make lockless operation optional pcm: dmix: Fix semaphore usage with lockless operation configure.ac | 13 +++++++++++++ src/pcm/pcm_direct.c | 16 +++++++++++++--- 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 + 7 files changed, 41 insertions(+), 11 deletions(-) -- 2.16.4

2 4

[PATCH v5 0/3] ALSA: compress: Document stream states and fix gapless SM
by Vinod Koul 07 Jul '20

07 Jul '20

Srini found issue with gapless implementation which prompted to look deeper into SM for compressed stream. So documenting SM was first step, so first two patches add that. Last patch fixes the issue by keeping track on partial_drain and then moving state to 'running' in snd_compr_drain_notify() for partial_drain case on success. While at it, noticed snd_compr_drain_notify() is lockless state change, so fixed that as well. I have tested this on Dragon board RB3, compressed audio works out of the box on that platform and Srini will send driver and fcplay patches for gapless soon. Changes in v5: - Update SM diagram as suggested by Amadeusz Changes in v4: - Add review tag by Charles - fix the lock deadlock pointed by Charles Changes in v3: - Add pause->stop->free transition - Add setup->free transition - remove running->free as that goes thru stop Changes in v2: - Added tested tag by Srini - Update compress SM with Free state and compr_stop() transitions Vinod Koul (3): ALSA: compress: document the compress audio state machine ALSA: compress: document the compress gapless audio state machine ALSA: compress: fix partial_drain completion state .../sound/designs/compress-offload.rst | 83 +++++++++++++++++++ include/sound/compress_driver.h | 10 ++- sound/core/compress_offload.c | 4 + 3 files changed, 96 insertions(+), 1 deletion(-) -- 2.26.2

3 8

[PATCH] Add implicit feedback quirk for RTX6001
by Pavel Hofman 07 Jul '20

07 Jul '20

USB Audio analyzer RTX6001 uses the same implicit feedback quirk as other XMOS-based devices. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> Tested-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a777d36c4f5a..40b7cd13fed9 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -368,6 +368,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, goto add_sync_ep_from_ifnum; case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ + case USB_ID(0x0d9a, 0x00df): /* RTX6001 */ ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; -- 2.25.1

2 1

[PATCH -next] sound: fix SND_HDA_GENERIC kconfig & build
by Randy Dunlap 07 Jul '20

07 Jul '20

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kconfig warnings and lots of subsequent build errors by adding yet another ugly select statement: WARNING: unmet direct dependencies detected for LEDS_CLASS Depends on [n]: NEW_LEDS [=n] Selected by [y]: - SND_HDA_GENERIC [=y] && SOUND [=y] && !UML && SND [=y] && SND_HDA [=y] && SND_HDA_GENERIC_LEDS [=y] WARNING: unmet direct dependencies detected for LEDS_TRIGGERS Depends on [n]: NEW_LEDS [=n] && LEDS_CLASS [=y] Selected by [y]: - SND_HDA_GENERIC [=y] && SOUND [=y] && !UML && SND [=y] && SND_HDA [=y] && SND_HDA_GENERIC_LEDS [=y] Selected by [m]: - MAC80211_LEDS [=y] && NET [=y] && WIRELESS [=y] && MAC80211 [=m] && LEDS_CLASS [=y] - IWLWIFI_LEDS [=y] && NETDEVICES [=y] && WLAN [=y] && WLAN_VENDOR_INTEL [=y] && IWLWIFI [=m] && (LEDS_CLASS [=y]=y || LEDS_CLASS [=y]=IWLWIFI [=m]) && (IWLMVM [=m] || IWLDVM [=m]) WARNING: unmet direct dependencies detected for LEDS_TRIGGER_AUDIO Depends on [n]: NEW_LEDS [=n] && LEDS_TRIGGERS [=y] Selected by [y]: - SND_HDA_GENERIC [=y] && SOUND [=y] && !UML && SND [=y] && SND_HDA [=y] && SND_HDA_GENERIC_LEDS [=y] Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/pci/hda/Kconfig | 1 + 1 file changed, 1 insertion(+) --- linux-next-20200701.orig/sound/pci/hda/Kconfig +++ linux-next-20200701/sound/pci/hda/Kconfig @@ -221,6 +221,7 @@ comment "Set to Y if you want auto-loadi config SND_HDA_GENERIC tristate "Enable generic HD-audio codec parser" + select NEW_LEDS if SND_HDA_GENERIC_LEDS select LEDS_CLASS if SND_HDA_GENERIC_LEDS select LEDS_TRIGGERS if SND_HDA_GENERIC_LEDS select LEDS_TRIGGER_AUDIO if SND_HDA_GENERIC_LEDS

2 1

[RESEND PATCH] ALSA: usb-audio: add quirk for MacroSilicon MS2109
by Hector Martin 07 Jul '20

07 Jul '20

These devices claim to be 96kHz mono, but actually are 48kHz stereo with swapped channels and unaligned transfers. Cc: stable(a)vger.kernel.org Signed-off-by: Hector Martin <marcan(a)marcan.st> --- 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

2 1

[PATCH] ALSA: core: pcm_iec958: fix kernel-doc
by Pierre-Louis Bossart 07 Jul '20

07 Jul '20

Fix W=1 warning. The kernel doc uses the hwparams type instead of the argument name. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/core/pcm_iec958.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_iec958.c b/sound/core/pcm_iec958.c index 073540f73b2f..f9a211cc1f2c 100644 --- a/sound/core/pcm_iec958.c +++ b/sound/core/pcm_iec958.c @@ -103,7 +103,7 @@ EXPORT_SYMBOL(snd_pcm_create_iec958_consumer); /** * snd_pcm_create_iec958_consumer_hw_params - create IEC958 channel status - * @hw_params: the hw_params instance for extracting rate and sample format + * @params: the hw_params instance for extracting rate and sample format * @cs: channel status buffer, at least four bytes * @len: length of channel status buffer * -- 2.25.1

2 1

[PATCH v2 1/3] ALSA: hda/realtek - Enable audio jacks of Acer vCopperbox with ALC269VC
by Jian-Hong Pan 07 Jul '20

07 Jul '20

The Acer desktop vCopperbox with ALC269VC cannot detect the MIC of headset, the line out and internal speaker until ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS quirk applied. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Chris Chiu <chiu(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 737ef82a75fd..dfb4bca07d3f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6149,6 +6149,7 @@ enum { ALC236_FIXUP_HP_MUTE_LED, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, + ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, }; static const struct hda_fixup alc269_fixups[] = { @@ -7327,6 +7328,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90100120 }, /* use as internal speaker */ + { 0x18, 0x02a111f0 }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01011020 }, /* use as line out */ + { }, + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7346,6 +7358,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK), + SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), -- 2.27.0

3 7

[PATCH v2 0/2] Fix Lenovo ThinkPad X1 Carbon 7th gen bass volume
by Benjamin Poirier 07 Jul '20

07 Jul '20

This is an updated version of the patchset to fix the low output volume on the Lenovo ThinkPad X1 Carbon 7th gen. With these patches, the volume output with the Linux driver matches the one with the Windows driver. Volume control is effective and headphone output is unaffected. Multiple users of this machine have reported a low output volume [1] when using a kernel with commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen"). Many have tried these patches or a similar workaround using `hda-verb` and prefer this configuration[2]. As noted in patch 2, the changes also depend on an associated alsa-ucm-conf change. [1] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… ... [2] https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcommen… ... Changes: v1 == https://lore.kernel.org/alsa-devel/20200211055651.4405-1-benjamin.poirier@g… v2 == * patch 1 Shortened log * patch 2 Change log to replace experimental observations by information from diagram posted on kernel.org bugzilla. Select exactly the desired connection for node 0x17 (bass speakers) instead of reusing alc295_fixup_disable_dac3(). Control node 0x21 (headphones) connection to avoid the volume "wobble" effect described in the log. Include SET_CONNECT_SEL command in fixup since it is not done by the hda core. Rename one control to reflect its new function and to use as a condition in the ucm config. Benjamin Poirier (2): ALSA: hda/realtek - Fix Lenovo Thinkpad X1 Carbon 7th quirk subdevice id ALSA: hda/realtek - Replace Lenovo Thinkpad X1 Carbon 7th quirk sound/pci/hda/patch_realtek.c | 57 ++++++++++++++++++++++++++++++++--- 1 file changed, 52 insertions(+), 5 deletions(-) -- 2.27.0

4 8

[PATCH 1/2] ALSA: hda/hdmi: fix failures at PCM open on Intel ICL and later
by Kai Vehmanen 07 Jul '20

07 Jul '20

When HDMI PCM devices are opened in a specific order, with at least one HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the connected monitor, on recent Intel platforms (ICL/JSL and newer). While this is not a typical sequence, at least Pulseaudio does this every time when it is started, to discover the available PCMs. The rootcause is an invalid assumption in hdmi_add_pin(), where the total number of converters is assumed to be known at the time the function is called. On older Intel platforms this held true, but after ICL/JSL, the order how pins and converters are in the subnode list as returned by snd_hda_get_sub_nodes(), was changed. As a result, information for some converters was not stored to per_pin->mux_nids. And this means some pins cannot be connected to all converters, and application instead gets -EBUSY instead at open. The assumption that converters are always before pins in the subnode list, is not really a valid one. Fix the problem in hdmi_parse_codec() by introducing separate loops for discovering converters and pins. BugLink: https://github.com/thesofproject/linux/issues/1978 BugLink: https://github.com/thesofproject/linux/issues/2216 BugLink: https://github.com/thesofproject/linux/issues/2217 Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 36 +++++++++++++++++++++++------------- 1 file changed, 23 insertions(+), 13 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e2b21ef5d7d1..295cc5a989d5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1804,33 +1804,43 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static int hdmi_parse_codec(struct hda_codec *codec) { - hda_nid_t nid; + hda_nid_t start_nid; + unsigned int caps; int i, nodes; - nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &nid); - if (!nid || nodes < 0) { + nodes = snd_hda_get_sub_nodes(codec, codec->core.afg, &start_nid); + if (!start_nid || nodes < 0) { codec_warn(codec, "HDMI: failed to get afg sub nodes\n"); return -EINVAL; } - for (i = 0; i < nodes; i++, nid++) { - unsigned int caps; - unsigned int type; + /* + * hdmi_add_pin() assumes total amount of converters to + * be known, so first discover all converters + */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; caps = get_wcaps(codec, nid); - type = get_wcaps_type(caps); if (!(caps & AC_WCAP_DIGITAL)) continue; - switch (type) { - case AC_WID_AUD_OUT: + if (get_wcaps_type(caps) == AC_WID_AUD_OUT) hdmi_add_cvt(codec, nid); - break; - case AC_WID_PIN: + } + + /* discover audio pins */ + for (i = 0; i < nodes; i++) { + hda_nid_t nid = start_nid + i; + + caps = get_wcaps(codec, nid); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + if (get_wcaps_type(caps) == AC_WID_PIN) hdmi_add_pin(codec, nid); - break; - } } return 0; -- 2.27.0

2 3

[PATCH 1/1] ALSA: opl3: fix infoleak in opl3
by Xidong Wang 07 Jul '20

07 Jul '20

From: xidongwang <wangxidong_97(a)163.com> The stack object “info” in snd_opl3_ioctl() has a leaking problem. It has 2 padding bytes which are not initialized and leaked via “copy_to_user”. Signed-off-by: xidongwang <wangxidong_97(a)163.com> --- sound/drivers/opl3/opl3_synth.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index e69a4ef..08c10ac 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -91,6 +91,8 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, { struct snd_dm_fm_info info; + memset(&info, 0, sizeof(info)); + info.fm_mode = opl3->fm_mode; info.rhythm = opl3->rhythm; if (copy_to_user(argp, &info, sizeof(struct snd_dm_fm_info))) -- 2.7.4

2 1

[RFC PATCH] ALSA: compress: add support to change codec profile in gapless playback
by Srinivas Kandagatla 06 Jul '20

06 Jul '20

For gapless playback its possible that each track can have different codec profile with same decoder, for example we have WMA album, we may have different tracks as WMA v9, WMA v10 and so on Existing code does not allow to change this profile while doing gapless playback. This patch adds new SNDRV_COMPRESS_SET_CODEC_PARAMS IOCTL to allow userspace to set this new parameters required for new codec profile. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- include/sound/compress_driver.h | 5 +++ include/sound/soc-component.h | 3 ++ include/sound/soc-dai.h | 5 +++ include/uapi/sound/compress_offload.h | 1 + sound/core/compress_offload.c | 54 ++++++++++++++++++++++++--- sound/soc/soc-compress.c | 30 +++++++++++++++ sound/soc/soc-dai.c | 14 +++++++ 7 files changed, 106 insertions(+), 6 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 70cbc5095e72..8d23351f7ad7 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -93,6 +93,9 @@ struct snd_compr_stream { * @set_params: Sets the compressed stream parameters, mandatory * This can be called in during stream creation only to set codec params * and the stream properties + * @set_codec_params: Sets the compressed stream codec parameters, mandatory + * This can be called in during gapless next track codec change only to set + * codec params * @get_params: retrieve the codec parameters, mandatory * @set_metadata: Set the metadata values for a stream * @get_metadata: retrieves the requested metadata values from stream @@ -112,6 +115,8 @@ struct snd_compr_ops { int (*free)(struct snd_compr_stream *stream); int (*set_params)(struct snd_compr_stream *stream, struct snd_compr_params *params); + int (*set_codec_params)(struct snd_compr_stream *stream, + struct snd_codec *params); int (*get_params)(struct snd_compr_stream *stream, struct snd_codec *params); int (*set_metadata)(struct snd_compr_stream *stream, diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5663891148e3..1e69c54ed0b9 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -36,6 +36,9 @@ struct snd_compress_ops { int (*get_params)(struct snd_soc_component *component, struct snd_compr_stream *stream, struct snd_codec *params); + int (*set_codec_params)(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_codec *params); int (*set_metadata)(struct snd_soc_component *component, struct snd_compr_stream *stream, struct snd_compr_metadata *metadata); diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 212257e84fac..526794ee555b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -192,6 +192,9 @@ int snd_soc_dai_compr_trigger(struct snd_soc_dai *dai, int snd_soc_dai_compr_set_params(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_compr_params *params); +int snd_soc_dai_compr_set_codec_params(struct snd_soc_dai *dai, + struct snd_compr_stream *cstream, + struct snd_codec *codec); int snd_soc_dai_compr_get_params(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, struct snd_codec *params); @@ -292,6 +295,8 @@ struct snd_soc_cdai_ops { struct snd_soc_dai *); int (*set_params)(struct snd_compr_stream *, struct snd_compr_params *, struct snd_soc_dai *); + int (*set_codec_params)(struct snd_compr_stream *, + struct snd_codec *, struct snd_soc_dai *); int (*get_params)(struct snd_compr_stream *, struct snd_codec *, struct snd_soc_dai *); int (*set_metadata)(struct snd_compr_stream *, diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h index 7184265c0b0d..c46286113a4b 100644 --- a/include/uapi/sound/compress_offload.h +++ b/include/uapi/sound/compress_offload.h @@ -172,6 +172,7 @@ struct snd_compr_metadata { struct snd_compr_metadata) #define SNDRV_COMPRESS_GET_METADATA _IOWR('C', 0x15,\ struct snd_compr_metadata) +#define SNDRV_COMPRESS_SET_CODEC_PARAMS _IOW('C', 0x16, struct snd_codec) #define SNDRV_COMPRESS_TSTAMP _IOR('C', 0x20, struct snd_compr_tstamp) #define SNDRV_COMPRESS_AVAIL _IOR('C', 0x21, struct snd_compr_avail) #define SNDRV_COMPRESS_PAUSE _IO('C', 0x30) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 0e53f6f31916..1c4b2cf450a0 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -566,6 +566,18 @@ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, return 0; } +static int snd_compress_check_codec_params(struct snd_codec *codec) +{ + /* now codec parameters */ + if (codec->id == 0 || codec->id > SND_AUDIOCODEC_MAX) + return -EINVAL; + + if (codec->ch_in == 0 || codec->ch_out == 0) + return -EINVAL; + + return 0; +} + static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ @@ -574,14 +586,41 @@ static int snd_compress_check_input(struct snd_compr_params *params) params->buffer.fragments == 0) return -EINVAL; - /* now codec parameters */ - if (params->codec.id == 0 || params->codec.id > SND_AUDIOCODEC_MAX) - return -EINVAL; + return snd_compress_check_codec_params(&params->codec); - if (params->codec.ch_in == 0 || params->codec.ch_out == 0) - return -EINVAL; +} - return 0; +static int snd_compr_set_codec_params(struct snd_compr_stream *stream, + unsigned long arg) +{ + struct snd_codec *params; + int retval; + + if (!stream->ops->set_codec_params) + return -EPERM; + + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + return -EPERM; + + /* codec params can be only set when next track has been signalled */ + if (stream->next_track == false) + return -EPERM; + + params = memdup_user((void __user *)arg, sizeof(*params)); + if (IS_ERR(params)) + return PTR_ERR(params); + + retval = snd_compress_check_codec_params(params); + if (retval) + goto out; + + retval = stream->ops->set_codec_params(stream, params); + if (retval) + goto out; + +out: + kfree(params); + return retval; } static int @@ -964,6 +1003,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) case _IOC_NR(SNDRV_COMPRESS_SET_PARAMS): retval = snd_compr_set_params(stream, arg); break; + case _IOC_NR(SNDRV_COMPRESS_SET_CODEC_PARAMS): + retval = snd_compr_set_codec_params(stream, arg); + break; case _IOC_NR(SNDRV_COMPRESS_GET_PARAMS): retval = snd_compr_get_params(stream, arg); break; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 4984b6a2c370..e549e0197aca 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -518,6 +518,34 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, return ret; } +static int soc_compr_set_codec_params(struct snd_compr_stream *cstream, + struct snd_codec *codec) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_component *component; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + int i, ret; + + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + + ret = snd_soc_dai_compr_set_codec_params(cpu_dai, cstream, codec); + if (ret < 0) + goto err; + + for_each_rtd_components(rtd, i, component) { + if (!component->driver->compress_ops || + !component->driver->compress_ops->set_codec_params) + continue; + + ret = component->driver->compress_ops->set_codec_params(component, cstream, + codec); + break; + } +err: + mutex_unlock(&rtd->card->pcm_mutex); + return ret; +} + static int soc_compr_get_params(struct snd_compr_stream *cstream, struct snd_codec *params) { @@ -728,6 +756,7 @@ static struct snd_compr_ops soc_compr_ops = { .open = soc_compr_open, .free = soc_compr_free, .set_params = soc_compr_set_params, + .set_codec_params = soc_compr_set_codec_params, .set_metadata = soc_compr_set_metadata, .get_metadata = soc_compr_get_metadata, .get_params = soc_compr_get_params, @@ -744,6 +773,7 @@ static struct snd_compr_ops soc_compr_dyn_ops = { .free = soc_compr_free_fe, .set_params = soc_compr_set_params_fe, .get_params = soc_compr_get_params, + .set_codec_params = soc_compr_set_codec_params, .set_metadata = soc_compr_set_metadata, .get_metadata = soc_compr_get_metadata, .trigger = soc_compr_trigger_fe, diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index b05e18b63a1c..06481d0278b8 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -594,6 +594,20 @@ int snd_soc_dai_compr_get_params(struct snd_soc_dai *dai, } EXPORT_SYMBOL_GPL(snd_soc_dai_compr_get_params); +int snd_soc_dai_compr_set_codec_params(struct snd_soc_dai *dai, + struct snd_compr_stream *cstream, + struct snd_codec *codec) +{ int ret = 0; + + if (dai->driver->cops && + dai->driver->cops->set_codec_params) + ret = dai->driver->cops->set_codec_params(cstream, codec, dai); + + return soc_dai_ret(dai, ret); + +} +EXPORT_SYMBOL_GPL(snd_soc_dai_compr_set_codec_params); + int snd_soc_dai_compr_ack(struct snd_soc_dai *dai, struct snd_compr_stream *cstream, size_t bytes) -- 2.21.0

3 5

[PATCH] ASoC: codecs: wm8400: remove unused variables
by Pierre-Louis Bossart 06 Jul '20

06 Jul '20

Fix W=1 warning by removing unused variables Suggested-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/wm8400.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 2551eb0f1868..35ffa7765c85 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -67,16 +67,12 @@ static void wm8400_component_reset(struct snd_soc_component *component) wm8400_reset_codec_reg_cache(wm8400->wm8400); } -static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 600, 0); - static const DECLARE_TLV_DB_SCALE(in_pga_tlv, -1650, 3000, 0); static const DECLARE_TLV_DB_SCALE(out_mix_tlv, -2100, 0, 0); static const DECLARE_TLV_DB_SCALE(out_pga_tlv, -7300, 600, 0); -static const DECLARE_TLV_DB_SCALE(out_omix_tlv, -600, 0, 0); - static const DECLARE_TLV_DB_SCALE(out_dac_tlv, -7163, 0, 0); static const DECLARE_TLV_DB_SCALE(in_adc_tlv, -7163, 1763, 0); @@ -439,14 +435,6 @@ static SOC_ENUM_SINGLE_DECL(wm8400_ainrmux_enum, static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls = SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum); -/* RXVOICE */ -static const struct snd_kcontrol_new wm8400_dapm_rxvoice_controls[] = { -SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8400_INPUT_MIXER5, WM8400_LR4BVOL_SHIFT, - WM8400_LR4BVOL_MASK, 0, in_mix_tlv), -SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8400_INPUT_MIXER6, WM8400_RL4BVOL_SHIFT, - WM8400_RL4BVOL_MASK, 0, in_mix_tlv), -}; - /* LOMIX */ static const struct snd_kcontrol_new wm8400_dapm_lomix_controls[] = { SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER1, -- 2.25.1

2 1

Re: [PATCH v2 1/7] Documentation: device-tree: sound: Update lpass-cpu driver binding
by Rohit Kumar 06 Jul '20

06 Jul '20

On 5/16/2020 11:07 PM, Ajit Pandey wrote: > On 5/14/2020 10:14 PM, Mark Brown wrote: >> On Thu, May 14, 2020 at 10:08:12PM +0530, Ajit Pandey wrote: >>> Done the required cleanups to update lpass-cpu binding with newer >>> yaml formats. >> Please do YAML conversions as the last thing in any patch series, there >> is quite a big backlog of reviews of YAML conversions so they can slow >> down the code development. It's good to do them, just try to make sure >> other patches don't depend on them. >> >> Please submit patches using subject lines reflecting the style for the >> subsystem, this makes it easier for people to identify relevant patches. >> Look at what existing commits in the area you're changing are doing and >> make sure your subject lines visually resemble what they're doing. >> There's no need to resubmit to fix this alone. > > Ok.. we'll do the yaml conversion as a separate patch chain so that it > won't block > > merging of this feature in ASoC core. @Mark, I was planning to post v3 patchset series for this and keep yaml change as last patch in the series as suggested. However, we have one update in the driver to support another compatible string. If we keep Documentation change at the last, the driver change will have checkpatch errors. Can you please suggest if should make first Documentation change in text file itself before driver change and finally have a patch to convert it to yaml at the end? Thanks, Rohit -- Qualcomm INDIA, on behalf of Qualcomm Innovation Center, Inc.is a member of the Code Aurora Forum, hosted by the Linux Foundation.

2 1

[PATCH v2] ASoC: cros_ec_codec: Log results when EC commands fail
by Yu-Hsuan Hsu 06 Jul '20

06 Jul '20

Log results of failed EC commands to identify a problem more easily. Replace cros_ec_cmd_xfer_status with cros_ec_cmd_xfer because the result has already been checked in this function. The wrapper is not needed. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/cros_ec_codec.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8d45c628e988e..a4ab62f59efa6 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -90,10 +90,17 @@ static int send_ec_host_command(struct cros_ec_device *ec_dev, uint32_t cmd, if (outsize) memcpy(msg->data, out, outsize); - ret = cros_ec_cmd_xfer_status(ec_dev, msg); + ret = cros_ec_cmd_xfer(ec_dev, msg); if (ret < 0) goto error; + if (msg->result != EC_RES_SUCCESS) { + dev_err(ec_dev->dev, "Command %d failed: %d\n", cmd, + msg->result); + ret = -EPROTO; + goto error; + } + if (insize) memcpy(in, msg->data, insize); -- 2.27.0.212.ge8ba1cc988-goog

4 12

Move ownership of snd-firewire-ctl-services repository
by Takashi Sakamoto 05 Jul '20

05 Jul '20

Hi Jaroslav, I'm currently working for the idea of 'ALSA control service in userspace'[1] for audio and music units on IEEE 1394 bus under my personal repository[2], and the work mostly done for Digidesign Digi 002/003 family and Tascam FireWire series[3]. As well as libhinawa and alsa-gobject repositories[4], I'd like to move ownership of the repository to 'GObject Introspection' team of alsa-project account, then continue the work. I'd like you to arrange access rights and so on, again. Besides, as I tols you in previous message[5], the nested team is not good in reasons. I strongly request you to leave 'GObject Introspection' team and drop 'Hinawa' team, again. Would I expect your arrangement for it? [1] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169141.html [2] https://github.com/takaswie/snd-firewire-ctl-services/ [3] https://github.com/takaswie/snd-firewire-ctl-services/tree/topic/services-f… [4] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169141.html [5] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169191.html Thanks Takashi Sakamoto

1 1

Some questions about asoc codec mixer elements naming
by Станислав Александров 05 Jul '20

05 Jul '20

Hello. I am writing an asoc codec driver and stuck with mixer and output naming. Could someone clarify a few things? This codec has two playback PGAs and 4 outputs: PGA1 -> [Speaker,Headphone,Line Out] PGA2 -> [Mono Output] [PGA1] controls volume and mute for all three outputs. [Speaker,Headphone,Line Out] have only mute. [PGA2] controls volume and mute. [Mono Output] also has mute. Codec is bound to platform driver via simple-audio-card: simple-audio-card,widgets = "Headphone", "Headphones", "Speaker", "Speakers", "Line", "Line Out Jack"; simple-audio-card,routing = "Headphones", "HPOUT", "Speakers", "SPKOUT", "Line Out Jack", "LINEOUT"; Is it possible to name mixer controls so that no further configuration is required when the driver is loaded? i.e. pulseaudio automatically detects all these outputs and controls, correctly maps them and allows switching between them? Currently i have this configuration(simplified): SOC_DOUBLE_S8_TLV("Master Playback Volume", PGA1, ...) SOC_DOUBLE("Master Playback Switch", PGA1, ...) SOC_SINGLE("Speaker Playback Switch", ...) SOC_SINGLE("Headphone Playback Switch", ...) SOC_SINGLE("Line Out Playback Switch", ...) SOC_SINGLE_S8_TLV("Mono Output Playback Volume", PGA2...) SOC_SINGLE("Mono Output Playback Switch", PGA2....) With these settings pulseaudio only detects Speaker and Headphone (but not Line Out because it wants to find its Jack) But as I understand "Master Playback" assumes control for all sound streams, so this is probably not correct =\ If it is impossible to name those elements that way so everything is autodetected, what should i use then? Write a machine audio driver or use alsaucm?

1 0

Asus Xonar SE
by Jefferson Smith Pizarro Gutierrez 05 Jul '20

05 Jul '20

hi, i have issues with my Asus Xonar SE souncard, i need to get a solution, please helpme to get sound through the spdif of the soundcard. i dont have sound

1 1

Looking for guidance for porting home made audio driver to ASoC/FSL
by Christophe Leroy 04 Jul '20

04 Jul '20

Hello I'm looking for guidance for porting a home made audio driver to ASoC/FSL on a custom board. The board has an MPC8321 freescale/NXP cpu. The board has a TDM bus connected to one of the cpu TDM port. On that TDM bus, there are three IDT 821034 quad codecs (PCM G.711) and an E1 chip that carries IP over E1. Today, there is a home made audio driver providing raw read/write devices plus an home made WAN HDLC driver for the E1. The CPU Quicc Engine (QE) is configured to connect the TDM port to one UCC. QUICC Multi-Channel Controller (QMC) is used to distribute E1 timeslots to the HDLC driver and each of the 12 codec timeslots to the audio driver. (The QMC is able to manage two types of channels: transparent and HDLC). We would like to make all this more standard and port it to use ALSA ASoC for the audio part, and the Freescale Quicc Engine HDLC. As of today, the Freescale QE HDLC wan driver uses the UCC directly, it doesn't interface through the QMC. And the FSL ASoC drivers seem to neither handle QE TDM via QE QMC nor QE UCC. We are thinking about adding a QE QMC layer in driver/soc/fsl/ which would more or less present to the consumer the same kind of API as the FSL UCC layer, and enhance the WAN FSL HDLC driver to be able to use that new layer. For the ALSA ASoC part, could some parts of existing FSL ASoC drivers be re-used of would we have to start implementing an ASoC from scratch ? Any recommendation and/or guidance would be welcomed and appreciated. Thanks in advance Christophe

1 0

[PATCH] ASoC: amd: fix Kconfig warning for Renoir
by Randy Dunlap 04 Jul '20

04 Jul '20

From: Randy Dunlap <rdunlap(a)infradead.org> SND_SOC_DMIC depends on GPIOLIB so this driver should also depend on GPIOLIB to make kconfig happy. WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_AMD_RENOIR_MACH [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AMD_RENOIR [=y] Fixes: 1d3776669323 ("ASoC: amd: enable build for RN machine driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Cc: Mark Brown <broonie(a)kernel.org> --- Found in mmotm. There are no object file build errors. I suppose that gpiolib stubs take care of that. Maybe some other patch would be more appropriate. sound/soc/amd/Kconfig | 1 + 1 file changed, 1 insertion(+) --- mmotm-2020-0703-1503.orig/sound/soc/amd/Kconfig +++ mmotm-2020-0703-1503/sound/soc/amd/Kconfig @@ -47,5 +47,6 @@ config SND_SOC_AMD_RENOIR_MACH tristate "AMD Renoir support for DMIC" select SND_SOC_DMIC depends on SND_SOC_AMD_RENOIR + depends on GPIOLIB help This option enables machine driver for DMIC

1 0

[PATCH v2] ASoC: fsl_asrc: Add an option to select internal ratio mode
by Shengjiu Wang 04 Jul '20

04 Jul '20

The ASRC not only supports ideal ratio mode, but also supports internal ratio mode. For internal rato mode, the rate of clock source should be divided with no remainder by sample rate, otherwise there is sound distortion. Add function fsl_asrc_select_clk() to find proper clock source for internal ratio mode, if the clock source is available then internal ratio mode will be selected. With change, the ideal ratio mode is not the only option for user. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - update according to Nicolin's comments sound/soc/fsl/fsl_asrc.c | 54 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 52 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 95f6a9617b0b..4105ef2c4f99 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -582,11 +582,55 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints); } +/** + * Select proper clock source for internal ratio mode + */ +static int fsl_asrc_select_clk(struct fsl_asrc_priv *asrc_priv, + struct fsl_asrc_pair *pair, + int in_rate, + int out_rate) +{ + struct fsl_asrc_pair_priv *pair_priv = pair->private; + struct asrc_config *config = pair_priv->config; + int rate[2], select_clk[2]; /* Array size 2 means IN and OUT */ + int clk_rate, clk_index; + int i = 0, j = 0; + + rate[0] = in_rate; + rate[1] = out_rate; + + /* Select proper clock source for internal ratio mode */ + for (j = 0; j < 2; j++) { + for (i = 0; i < ASRC_CLK_MAP_LEN; i++) { + clk_index = asrc_priv->clk_map[j][i]; + clk_rate = clk_get_rate(asrc_priv->asrck_clk[clk_index]); + /* Only match a perfect clock source with no remainder */ + if (clk_rate != 0 && (clk_rate / rate[j]) <= 1024 && + (clk_rate % rate[j]) == 0) + break; + } + + select_clk[j] = i; + } + + /* Switch to ideal ratio mode if there is no proper clock source */ + if (select_clk[IN] == ASRC_CLK_MAP_LEN || select_clk[OUT] == ASRC_CLK_MAP_LEN) { + select_clk[IN] = INCLK_NONE; + select_clk[OUT] = OUTCLK_ASRCK1_CLK; + } + + config->inclk = select_clk[IN]; + config->outclk = select_clk[OUT]; + + return 0; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct fsl_asrc *asrc = snd_soc_dai_get_drvdata(dai); + struct fsl_asrc_priv *asrc_priv = asrc->private; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; struct fsl_asrc_pair_priv *pair_priv = pair->private; @@ -605,8 +649,6 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.pair = pair->index; config.channel_num = channels; - config.inclk = INCLK_NONE; - config.outclk = OUTCLK_ASRCK1_CLK; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config.input_format = params_format(params); @@ -620,6 +662,14 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } + ret = fsl_asrc_select_clk(asrc_priv, pair, + config.input_sample_rate, + config.output_sample_rate); + if (ret) { + dev_err(dai->dev, "fail to select clock\n"); + return ret; + } + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); -- 2.21.0

3 5

[PATCH v2 0/2] ASoC: rt5670: 2 small cleanups
by Hans de Goede 03 Jul '20

03 Jul '20

Hi All, Here is in essence a resend of my 2 cleanup patches for the rt5670 ASoC codec code, rebased on top of broonie/sound/for-5.9 with broonie/sound/for-5.8 merged in. Regards, Hans p.s. I'll also send out the patch improving the comment in cht_bsw_rt5672.c cht_codec_fixup() which Pierre-Louis requested soon.

2 4

[PATCH v2 0/6] ASoC: fsl: fix kernel-doc
by Pierre-Louis Bossart 03 Jul '20

03 Jul '20

This cleanup patchset is only about kernel-doc, mostly trivial edits and format correction. v2: added Nicolin Chen's Acked-by tags and included the three suggested edits. Pierre-Louis Bossart (6): ASoC: fsl: fsl_ssi_dbg: remove spurious kernel-doc comment start ASoC: fsl: fsl_ssi: fix kernel-doc ASoC: fsl: fsl-asoc-card: fix kernel-doc ASoC: fsl: fsl_spdif: fix kernel-doc ASoC: fsl: fsl_asrc: fix kernel-doc ASoC: fsl: fsl_esai: fix kernel-doc sound/soc/fsl/fsl-asoc-card.c | 21 +++++------ sound/soc/fsl/fsl_asrc.c | 57 +++++++++++++++++++--------- sound/soc/fsl/fsl_esai.c | 32 +++++++++------- sound/soc/fsl/fsl_spdif.c | 5 ++- sound/soc/fsl/fsl_ssi.c | 70 ++++++++++++++++++++++------------- sound/soc/fsl/fsl_ssi_dbg.c | 4 +- 6 files changed, 115 insertions(+), 74 deletions(-) -- 2.25.1

2 7

[RESEND PATCH V3] ASoC: amd: add logic to check dmic hardware runtime
by Vijendar Mukunda 03 Jul '20

03 Jul '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> --- v3: changed dmic_acpi_check module param as bint v2: changed module param usage -1 as default, 0 - skip, 1 - force sound/soc/amd/renoir/rn-pci-acp3x.c | 29 +++++++++++++++++++++++++++++ sound/soc/amd/renoir/rn_acp3x.h | 2 ++ 2 files changed, 31 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 859ed67..d1faea5 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 = -1 - Checks ACPI method to know DMIC hardware status runtime + * = 0 - Skips the DMIC device creation and returns probe failure + * = 1 - Assumes that platform has DMIC support and skips ACPI + * method check + */ +static int dmic_acpi_check = ACP_DMIC_AUTO; +module_param(dmic_acpi_check, bint, 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..1462039 100644 --- a/sound/soc/amd/renoir/rn_acp3x.h +++ b/sound/soc/amd/renoir/rn_acp3x.h @@ -55,6 +55,8 @@ #define MAX_BUFFER (CAPTURE_MAX_PERIOD_SIZE * CAPTURE_MAX_NUM_PERIODS) #define MIN_BUFFER MAX_BUFFER +#define ACP_DMIC_AUTO -1 + struct pdm_dev_data { u32 pdm_irq; void __iomem *acp_base; -- 2.7.4

2 1

[PATCH -next] ASoC: ti: j721e-evm: Fix missing unlock on error in j721e_audio_hw_params()
by Wei Yongjun 03 Jul '20

03 Jul '20

Add the missing unlock before return from function j721e_audio_hw_params() in the error handling case. Fixes: 6748d0559059 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/ti/j721e-evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index 3a2a8b1f3aa3..174306cf53ad 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -330,7 +330,7 @@ static int j721e_audio_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, slot_width); if (ret && ret != -ENOTSUPP) - return ret; + goto out; } ret = j721e_configure_refclk(priv, domain_id, params_rate(params));

3 2

[PATCH] ASoC: soc-ac97: fix kernel-doc
by Pierre-Louis Bossart 03 Jul '20

03 Jul '20

Fix W=1 warning. Add missing arguments Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/soc-ac97.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 906106ed8ca1..65db083e242b 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -393,6 +393,8 @@ EXPORT_SYMBOL_GPL(snd_soc_set_ac97_ops); /** * snd_soc_set_ac97_ops_of_reset - Set ac97 ops with generic ac97 reset functions + * @ops: bus ops + * @pdev: platform device * * This function sets the reset and warm_reset properties of ops and parses * the device node of pdev to get pinctrl states and gpio numbers to use. -- 2.25.1

2 1

[PATCH][next] ASoC: Intel: bxt-da7219-max98357a: return -EINVAL on unrecognized speaker amplifier
by Colin King 03 Jul '20

03 Jul '20

From: Colin Ian King <colin.king(a)canonical.com> Currently if the ctx->spkamp is not recognized an error message is reported but the code continues to set up the device with uninitialized variables such as the number of widgets. Fix this by returning -EINVAL for unrecognized speaker amplifier types. Addresses-Coverity: ("Uninitialized scalar variable") Fixes: e1435a1feb18 ("ASoC: Intel: bxt-da7219-max98357a: support MAX98390 speaker amp") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 4d39253e796b..0c0a717823c4 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -677,7 +677,7 @@ static int bxt_card_late_probe(struct snd_soc_card *card) break; default: dev_err(card->dev, "Invalid speaker amplifier %d\n", ctx->spkamp); - break; + return -EINVAL; } err = snd_soc_dapm_new_controls(&card->dapm, widgets, num_widgets); -- 2.27.0

3 2

[PATCH] ASoC: core: Remove only the registered component in devm functions
by Maxime Ripard 03 Jul '20

03 Jul '20

The ASoC devm_ functions that register a component (devm_snd_soc_register_component and devm_snd_dmaengine_pcm_register) will clean their component by running snd_soc_unregister_component. snd_soc_unregister_component will then remove all the components for the device that was used to register the component in the first place. However, some drivers register several components (such as a DAI and a dmaengine PCM) on the same device, and if the dmaengine PCM is registered first, then the DAI will be cleaned up first and snd_dmaengine_pcm_unregister will be called next. snd_dmaengine_pcm_unregister will then lookup the dmaengine PCM component on the device, and if there's one unregister that component and release its dmaengine channels. That doesn't happen in practice though since the first call to snd_soc_unregister_component removed all the components, so we never get the chance to release the dmaengine channels. In order to fix this, instead of removing all the components for a given device, we can simply remove the component that was registered in the first place. We should have the same number of component registration than we have components, so it should work just fine. Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- This was observed on a RaspberryPi that uses the vc4_hdmi driver (drivers/gpu/drm/vc4/vc4_hdmi.c). This driver will register a dmaengine PCM and two components. If the MIPI-DSI controller is enabled, it will create an EPROBE_DEFER across the entire display pipeline, leading to multiple bind/unbind cycles in the other display drivers including vc4_hdmi, leading to multiple warnings since we request the same dmaengine channels on the same device without ever releasing them. It's not really clear to me when that bug was introduced exactly, since it can only be seen on a rather unusual setup, and with all the drivers built-in (otherwise we probably wouldn't get an EPROBE_DEFER for DSI), but it still looks like something that should probably go to stable? --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 27 +++++++++++++++++++++++++++ sound/soc/soc-devres.c | 8 +++++--- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 4 files changed, 35 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ef5dd28e10a9..0b3b31803678 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -444,6 +444,8 @@ int devm_snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *component_driver, struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_component(struct device *dev); +void snd_soc_unregister_component_by_driver(struct device *dev, + const struct snd_soc_component_driver *component_driver); struct snd_soc_component *snd_soc_lookup_component(struct device *dev, const char *driver_name); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7b387202c5db..9d24cbd9111f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2571,6 +2571,33 @@ int snd_soc_register_component(struct device *dev, } EXPORT_SYMBOL_GPL(snd_soc_register_component); +/** + * snd_soc_unregister_component_by_driver - Unregister component using a given driver + * from the ASoC core + * + * @dev: The device to unregister + * @component_driver: The component driver to unregister + */ +void snd_soc_unregister_component_by_driver(struct device *dev, + const struct snd_soc_component_driver *component_driver) +{ + struct snd_soc_component *component; + + if (!component_driver) + return; + + mutex_lock(&client_mutex); + component = snd_soc_lookup_component_nolocked(dev, component_driver->name); + if (!component) + goto out; + + snd_soc_del_component_unlocked(component); + +out: + mutex_unlock(&client_mutex); +} +EXPORT_SYMBOL_GPL(snd_soc_unregister_component_by_driver); + /** * snd_soc_unregister_component - Unregister all related component * from the ASoC core diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index a9ea172a66a7..c6364caabc0e 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -11,7 +11,9 @@ static void devm_component_release(struct device *dev, void *res) { - snd_soc_unregister_component(*(struct device **)res); + const struct snd_soc_component_driver **cmpnt_drv = res; + + snd_soc_unregister_component_by_driver(dev, *cmpnt_drv); } /** @@ -28,7 +30,7 @@ int devm_snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai) { - struct device **ptr; + const struct snd_soc_component_driver **ptr; int ret; ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL); @@ -37,7 +39,7 @@ int devm_snd_soc_register_component(struct device *dev, ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai); if (ret == 0) { - *ptr = dev; + *ptr = cmpnt_drv; devres_add(dev, ptr); } else { devres_free(ptr); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index f728309a0833..ab600be9e69f 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -490,7 +490,7 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_component_to_pcm(component); - snd_soc_unregister_component(dev); + snd_soc_unregister_component_by_driver(dev, component->driver); dmaengine_pcm_release_chan(pcm); kfree(pcm); } -- 2.26.2

2 1

Xone32c bNumConfigurations issue
by Tim Nugent 03 Jul '20

03 Jul '20

Hi alsa-devel, Having a few problems with the Xone32c though not sure if this is an ALSA issue or general USB stack problem. It's a USB soundcard that is supposed to have 4 inputs and 4 outputs. It seems to have 2 USB configurations (bNumConfigurations : 0x2). When plugged in, bConfigurationValue is 0x1 and only 2 inputs and 2 outputs show up. Change this value to 0x2 and it shows up (as it should) as 4 inputs and 4 outputs. Here's the full USB description: https://pastebin.com/DDPXv5c1 I can change the bConfigurationValue with this udev rule, or with a `echo 2 > /sys/bus/usb/devices/1-1.4/bConfigurationValue`: https://gist.github.com/timnugent/ed65a79b2bd6c63788bfada3624756a4 This has all been working OK on Ubuntu 20.04 running kernel 5.4.0-33-generic. However, I just updated to 5.4.0-37-generic and now there is only a single USB configuration (bNumConfigurations : 0x1) and bConfigurationValue can't be changed to 0x2: https://pastebin.com/th9VRH0q echo 2 > /sys/bus/usb/devices/1-1.4/bConfigurationValue bash: echo: write error: Invalid argument So this could be a regression in snd_usb_audio, or maybe usbhid - which `lsudb -t` says are the drivers. This behaviour (only 1 configuration) also happens on a Raspberry Pi (Raspbian 4.19.66-v7+) - I haven't been able to get it to work at all. Thanks Tim

1 0

[PATCH 00/11] ASoC: codecs: fix kernel-doc and W=1 warnings
by Pierre-Louis Bossart 03 Jul '20

03 Jul '20

Second batch of cleanups to remove all the warnings reported by buildbots/compile tests. Pierre-Louis Bossart (11): ASoC: codecs: ak4458: fix 'set but not used' warning ASoC: codecs: cros_ec_codec: fix 'defined but not used' warning ASoC: codecs: da7210: fix kernel-doc ASoC: codecs: da7219: fix 'defined but not used' warning ASoC: codecs: es8316: fix 'defined but not used' warning ASoC: codecs: max98390: fix 'defined but not used' warning ASoC: codecs: rt*: fix 'defined but not used' warning ASoC: codecs: rt5659: add _maybe_unused as needed ASoC: codecs: rt5631: fix kernel-doc ASoC: codecs: tlv320aic26: fix kernel-doc warning ASoC: codecs: jz4770: add _maybe_unused as needed sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/cros_ec_codec.c | 2 ++ sound/soc/codecs/da7210.c | 12 +++++++----- sound/soc/codecs/da7219.c | 2 ++ sound/soc/codecs/es8316.c | 2 ++ sound/soc/codecs/jz4770.c | 2 +- sound/soc/codecs/max98390.c | 8 -------- sound/soc/codecs/rt274.c | 2 ++ sound/soc/codecs/rt286.c | 2 ++ sound/soc/codecs/rt298.c | 2 ++ sound/soc/codecs/rt5631.c | 8 ++++++-- sound/soc/codecs/rt5659.c | 14 +++++++------- sound/soc/codecs/rt5660.c | 2 ++ sound/soc/codecs/rt5677-spi.c | 2 ++ sound/soc/codecs/tlv320aic26.c | 2 +- 15 files changed, 39 insertions(+), 25 deletions(-) base-commit: a2b782d59c57cb6f673dbb4804ffc500336d7a54 -- 2.25.1

2 14

[HDAudio][ALC295] speakers not working on Acer Travelmate P614
by Stanislav Kazmin 03 Jul '20

03 Jul '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.

2 5

[PATCH 0/2] ASoC: fsl-asoc-card: add support for generic codecs
by Arnaud Ferraris 03 Jul '20

03 Jul '20

fsl-asoc-card currently only works with AC97 or a selection of codecs, although the hardware is capable of more. Supporting generic codecs when acting as I2S slave (codec is master) would be useful, especially when using Bluetooth audio, as these are generally simple I2S devices not controlled by the sound subsystem. This will allow using simple/dummy codecs along with ASRC. Arnaud Ferraris (2): dt-bindings: sound: fsl-asoc-card: add new compatible for I2S slave ASoC: fsl-asoc-card: add support for generic I2S slave use-case Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 23 ++++++++++++++++++++++- sound/soc/fsl/fsl-asoc-card.c | 46 +++++++++++++++++++++++++++++++++++----------- 2 files changed, 57 insertions(+), 12 deletions(-)

2 6

How to identify Alsa eLements?
by Tanjeff Moos 03 Jul '20

03 Jul '20

Hi alsa-devel, I'm programming a GUI for the mixer in my Focusrite Scarlett 6i6 interface. The Alsa driver supports that functionality, i.e. I can control the mixer using alsamixer, for example. Now I'm writing a Qt GUI for it. I wonder how I can reliably identify a mixer control element (e.g. the master volume control). Each element has an ID (struct snd_ctl_elem_id), consisting of multiple elements: - numid (seems to change at each boot) - iface - device - subdevice - name (is always constant) - index The docs says I can identify an element by the combination of name and index. My questions: 1) Are name and index enough to identify an element? 2) How can I obtain the index? The name and numid are shown by 'amixer contents', for example. Kind regards, Tanjeff

2 4

sof-hda-dsp UCM doesn't work with the latest alsa-lib
by GitHub issues - opened 03 Jul '20

03 Jul '20

alsa-project/alsa-lib issue #67 was opened from libinyang: In the latest UCM for sof-hda-dsp, it has supported Syntax 3 and it uses $var for hdmi. In sof-hda-dsp/Hdmi.conf, It defines "HdmiNum 1", and later it will redefine "HdmiNum 2", then "HdmiNum 3". It will try to `Include.hdmi1.File "/codecs/hda/hdmi.conf"` to use these definitions. But actually, only "HdmiNum 3" will be used for HDMI1, HDMI2 and HDMI3. The old value of "1" and "2" will be overridden when parsing HDMI1 and HDMI2. This is the bug of alsa-lib initialization sequence issue. HDA-Intel/Hdmi.conf should have the same issue. Issue URL : https://github.com/alsa-project/alsa-lib/issues/67 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH RFC alsa-ucm-conf] sof-hda-dsp: Use Master mixer to control volume with dual stereo speakers
by Benjamin Poirier 03 Jul '20

03 Jul '20

The Lenovo ThinkPad X1 Carbon 7th is a laptop with two sets of stereo speakers driven from a single stereo output. The volume of each set of speakers can be controlled independently through two separate mixers. In that case, one of those mixers serves a dual purpose: it also controls headphone volume. With the current ucm profile, when outputting to speakers, pulseaudio uses only the mixer for the front speakers to control volume. This is ineffective; rear speaker volume must be controlled as well. In order to control both sets of speakers at once, indicate that the Master mixer should be used. Moreover, when switching between speaker and headphone outputs, the level of the dual purpose mixer should be restored. Since this kind of constraint cannot be described with ucm, indicate that the Master mixer should be used to control volume instead when outputting to headphones. This patch depends on a modification to the snd_hda_codec_realtek kernel driver used on that system to rename one of the controls. Signed-off-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> --- Note that I plan to submit the related kernel patch upstream shortly, in the meantime, it is available here: https://github.com/gobenji/thinkpad-x1-gen7-sound Also note that this patch is based on commit 38e5906cd1b1 ("sof-hda-dsp: fix the device order (Hdmi devices)"). While testing with libasound built from alsa-lib git head 485930ea5dc8 ("ucm: substitution - remove duplicate allow_empty assignment"), alsa-ucm-conf versions including dcef48f13d4f ("HDA-Intel: add support for AMD acp microphone devices") and newer do not work for me. I've tested up to current head, ffe0cab5cfce ("sof-hda-dsp: use sof-hda-dsp/Hdmi.conf"). Looking at the output of `pulseaudio --log-level=debug`, things go wrong at: I: [pulseaudio] (alsa-lib)main.c: error: failed to import sof-hda-dsp use case configuration -17 ucm2/sof-hda-dsp/HiFi.conf | 32 ++++++++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) diff --git a/ucm2/sof-hda-dsp/HiFi.conf b/ucm2/sof-hda-dsp/HiFi.conf index 62358e2..661a086 100644 --- a/ucm2/sof-hda-dsp/HiFi.conf +++ b/ucm2/sof-hda-dsp/HiFi.conf @@ -25,10 +25,22 @@ SectionDevice."Headphones" { Value { PlaybackPriority 200 PlaybackPCM "hw:${CardId}" - PlaybackMixerElem "Headphone" PlaybackMasterElem "Master" - PlaybackVolume "Headphone Playback Volume" PlaybackSwitch "Headphone Playback Switch" + If.hpmixer { + Condition { + Type ControlExists + Control "name='Headphone/Bass Speaker Playback Volume'" + } + True { + PlaybackMixerElem "Master" + PlaybackVolume "Master Playback Volume" + } + False { + PlaybackMixerElem "Headphone" + PlaybackVolume "Headphone Playback Volume" + } + } If.jack { Condition { Type ControlExists @@ -77,10 +89,22 @@ SectionDevice."Speaker" { Value { PlaybackPriority 100 PlaybackPCM "hw:${CardId}" - PlaybackMixerElem "Speaker" PlaybackMasterElem "Master" - PlaybackVolume "Speaker Playback Volume" PlaybackSwitch "Speaker Playback Switch" + If.spkmixer { + Condition { + Type ControlExists + Control "name='Headphone/Bass Speaker Playback Volume'" + } + True { + PlaybackMixerElem "Master" + PlaybackVolume "Master Playback Volume" + } + False { + PlaybackMixerElem "Speaker" + PlaybackVolume "Speaker Playback Volume" + } + } } } -- 2.27.0

1 0

[PATCH 1/3] ALSA: hda/realtek - Enable audio jacks of Acer vCopperbox with ALC269VC
by Jian-Hong Pan 03 Jul '20

03 Jul '20

The Acer desktop vCopperbox with ALC269VC cannot detect the MIC of headset, the line out and internal speaker until ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS quirk applied. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Chris Chiu <chiu(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 737ef82a75fd..dfb4bca07d3f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6149,6 +6149,7 @@ enum { ALC236_FIXUP_HP_MUTE_LED, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, + ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, }; static const struct hda_fixup alc269_fixups[] = { @@ -7327,6 +7328,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90100120 }, /* use as internal speaker */ + { 0x18, 0x02a111f0 }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01011020 }, /* use as line out */ + { }, + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7346,6 +7358,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK), + SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), -- 2.27.0

2 3

[PATCH] ASoC: cros_ec_codec: Log results when EC commands fail
by Yu-Hsuan Hsu 03 Jul '20

03 Jul '20

Log results of failed EC commands to identify a problem more easily. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/cros_ec_codec.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8d45c628e988e..a4ab62f59efa6 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -90,10 +90,17 @@ static int send_ec_host_command(struct cros_ec_device *ec_dev, uint32_t cmd, if (outsize) memcpy(msg->data, out, outsize); - ret = cros_ec_cmd_xfer_status(ec_dev, msg); + ret = cros_ec_cmd_xfer(ec_dev, msg); if (ret < 0) goto error; + if (msg->result != EC_RES_SUCCESS) { + dev_err(ec_dev->dev, "Command %d failed: %d\n", cmd, + msg->result); + ret = -EPROTO; + goto error; + } + if (insize) memcpy(in, msg->data, insize); -- 2.27.0.212.ge8ba1cc988-goog

2 1

alsa-lib: ctl: improve documentation for identifier of control element
by GitHub pull_request - opened 03 Jul '20

03 Jul '20

alsa-project/alsa-lib pull request #66 was opened from takaswie: In documentation, there are two ways relevant to the identifier of control element. However, the case of combination has the lack of parameters. This commit improves documentation in this point. Fixes: f3c24de8c0df ("ctl: add an overview for design of ALSA control interface") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-lib/pull/66 Patch URL : https://github.com/alsa-project/alsa-lib/pull/66.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: dt-bindings: simple-card: Fix 'make dt_binding_check' warnings
by Fabio Estevam 03 Jul '20

03 Jul '20

The following build warnings are seen with 'make dt_binding_check': Documentation/devicetree/bindings/sound/simple-card.example.dts:209.46-211.15: Warning (unit_address_vs_reg): /example-4/sound/simple-audio-card,cpu@0: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:213.37-215.15: Warning (unit_address_vs_reg): /example-4/sound/simple-audio-card,cpu@1: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:250.42-261.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@0: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:263.42-288.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:270.32-272.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@0: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:273.23-275.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@1: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:276.23-278.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@2: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:279.23-281.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@3: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:290.42-303.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@2: node has a unit name, but no reg or ranges property Fix them all. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- .../devicetree/bindings/sound/simple-card.yaml | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml index 8132d0c0f00a..35e669020296 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.yaml +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -378,6 +378,8 @@ examples: - | sound { compatible = "simple-audio-card"; + #address-cells = <1>; + #size-cells = <0>; simple-audio-card,name = "rsnd-ak4643"; simple-audio-card,format = "left_j"; @@ -391,10 +393,12 @@ examples: "ak4642 Playback", "DAI1 Playback"; dpcmcpu: simple-audio-card,cpu@0 { + reg = <0>; sound-dai = <&rcar_sound 0>; }; simple-audio-card,cpu@1 { + reg = <1>; sound-dai = <&rcar_sound 1>; }; @@ -418,6 +422,8 @@ examples: - | sound { compatible = "simple-audio-card"; + #address-cells = <1>; + #size-cells = <0>; simple-audio-card,routing = "pcm3168a Playback", "DAI1 Playback", @@ -426,6 +432,7 @@ examples: "pcm3168a Playback", "DAI4 Playback"; simple-audio-card,dai-link@0 { + reg = <0>; format = "left_j"; bitclock-master = <&sndcpu0>; frame-master = <&sndcpu0>; @@ -439,22 +446,23 @@ examples: }; simple-audio-card,dai-link@1 { + reg = <1>; format = "i2s"; bitclock-master = <&sndcpu1>; frame-master = <&sndcpu1>; convert-channels = <8>; /* TDM Split */ - sndcpu1: cpu@0 { + sndcpu1: cpu0 { sound-dai = <&rcar_sound 1>; }; - cpu@1 { + cpu1 { sound-dai = <&rcar_sound 2>; }; - cpu@2 { + cpu2 { sound-dai = <&rcar_sound 3>; }; - cpu@3 { + cpu3 { sound-dai = <&rcar_sound 4>; }; codec { @@ -466,6 +474,7 @@ examples: }; simple-audio-card,dai-link@2 { + reg = <2>; format = "i2s"; bitclock-master = <&sndcpu2>; frame-master = <&sndcpu2>; -- 2.17.1

4 6

[PATCH 0/6] ASoC: fsl: fix kernel-doc
by Pierre-Louis Bossart 02 Jul '20

02 Jul '20

This cleanup patchset is only about kernel-doc, mostly trivial edits and format correction. Pierre-Louis Bossart (6): ASoC: fsl: fsl_ssi_dbg: remove spurious kernel-doc comment start ASoC: fsl: fsl_ssi: fix kernel-doc ASoC: fsl: fsl-asoc-card: fix kernel-doc ASoC: fsl: fsl_spdif: fix kernel-doc ASoC: fsl: fsl_asrc: fix kernel-doc ASoC: fsl: fsl_esai: fix kernel-doc sound/soc/fsl/fsl-asoc-card.c | 21 +++++------ sound/soc/fsl/fsl_asrc.c | 57 +++++++++++++++++++--------- sound/soc/fsl/fsl_esai.c | 32 +++++++++------- sound/soc/fsl/fsl_spdif.c | 5 ++- sound/soc/fsl/fsl_ssi.c | 70 ++++++++++++++++++++++------------- sound/soc/fsl/fsl_ssi_dbg.c | 4 +- 6 files changed, 115 insertions(+), 74 deletions(-) -- 2.25.1

2 14

[PATCH 0/7] ASoC: codecs: wm/cs: fix kernel-doc and W=1 warnings
by Pierre-Louis Bossart 02 Jul '20

02 Jul '20

I've been doing a lot of cleanups to get to zero warnings with W=1, here's the first batch for Cirrus/Wolfson codecs. Pierre-Louis Bossart (7): ASoC: codecs: cs4270: fix kernel-doc ASoC: codecs: cs42l42: remove always-true comparisons ASoC: codecs: wm8986: fix missing kernel-doc arguments ASoC: codecs: wm8960: fix kernel-doc ASoC: codecs: wm9713: remove spurious kernel-doc comment start ASoC: codecs: wm8994: fix kernel-doc ASoC: codecs: wm8400: add _maybe_unused as needed sound/soc/codecs/cs4270.c | 4 ++-- sound/soc/codecs/cs42l42.c | 9 +++------ sound/soc/codecs/wm8400.c | 6 +++--- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8994.c | 4 ++++ sound/soc/codecs/wm8996.c | 3 +++ sound/soc/codecs/wm9713.c | 2 +- 7 files changed, 17 insertions(+), 13 deletions(-) base-commit: a2b782d59c57cb6f673dbb4804ffc500336d7a54 -- 2.25.1

3 14

[PATCH alsa-lib 1/8] topology: decode: fix channel map memory allocation
by Piotr Maziarz 02 Jul '20

02 Jul '20

Without proper memory allocation behaviour was undefined. Signed-off-by: Piotr Maziarz <piotrx.maziarz(a)linux.intel.com> --- src/topology/ctl.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/src/topology/ctl.c b/src/topology/ctl.c index 90241b6..c8c7e94 100644 --- a/src/topology/ctl.c +++ b/src/topology/ctl.c @@ -1330,7 +1330,6 @@ int tplg_decode_control_enum1(snd_tplg_t *tplg, void *bin, size_t size) { struct snd_soc_tplg_enum_control *ec = bin; - struct snd_tplg_channel_map_template cmt; int i; if (size < sizeof(*ec)) { @@ -1375,11 +1374,11 @@ int tplg_decode_control_enum1(snd_tplg_t *tplg, } } - et->map = &cmt; - memset(&cmt, 0, sizeof(cmt)); - cmt.num_channels = ec->num_channels; - for (i = 0; i < cmt.num_channels; i++) { - struct snd_tplg_channel_elem *channel = &cmt.channel[i]; + et->map = tplg_calloc(heap, sizeof(struct snd_tplg_channel_map_template)); + et->map->num_channels = ec->num_channels; + for (i = 0; i < et->map->num_channels; i++) { + struct snd_tplg_channel_elem *channel = &et->map->channel[i]; + tplg_log(tplg, 'D', pos + ((void *)&ec->channel[i] - (void *)ec), "enum: channel size %d", ec->channel[i].size); channel->reg = ec->channel[i].reg; -- 2.7.4

2 11

Re: [PATCH] riscv: asm/gdb_xml.h: use __maybe_used to make W=1 warnings go away
by Pierre-Louis Bossart 02 Jul '20

02 Jul '20

On 7/1/20 9:04 PM, Vincent Chen wrote: > On Thu, Jul 2, 2020 at 2:48 AM Pierre-Louis Bossart > <pierre-louis.bossart(a)linux.intel.com> wrote: >> >> 0day/kbuild reports warnings with the ASoC codecs compiled with W=1. >> >> In file included from arch/riscv/include/asm/kgdb.h:109, >> from include/linux/kgdb.h:20, >> from include/linux/fb.h:5, >> from include/drm/drm_crtc.h:31, >> from sound/soc/codecs/hdmi-codec.c:19: >> >> arch/riscv/include/asm/gdb_xml.h:23:19: warning: >> 'riscv_gdb_stub_cpuxml' defined but not used [-Wunused-const-variable=] >> 23 | static const char riscv_gdb_stub_cpuxml[2048] = >> | ^~~~~~~~~~~~~~~~~~~~~ >> arch/riscv/include/asm/gdb_xml.h:16:19: warning: >> 'riscv_gdb_stub_target_desc' defined but not used [-Wunused-const-variable=] >> 16 | static const char riscv_gdb_stub_target_desc[256] = >> | ^~~~~~~~~~~~~~~~~~~~~~~~~~ >> arch/riscv/include/asm/gdb_xml.h:13:19: warning: >> 'gdb_xfer_read_cpuxml' defined but not used [-Wunused-const-variable=] >> 13 | static const char gdb_xfer_read_cpuxml[39] = >> | ^~~~~~~~~~~~~~~~~~~~ >> arch/riscv/include/asm/gdb_xml.h:10:19: warning: >> 'gdb_xfer_read_target' defined but not used [-Wunused-const-variable=] >> 10 | static const char gdb_xfer_read_target[31] = "qXfer:features:read:target.xml:"; >> | ^~~~~~~~~~~~~~~~~~~~ >> arch/riscv/include/asm/gdb_xml.h:7:19: warning: >> 'riscv_gdb_stub_feature' defined but not used [-Wunused-const-variable=] >> 7 | static const char riscv_gdb_stub_feature[64] = >> | ^~~~~~~~~~~~~~~~~~~~~~ >> >> Reported-by: kernel test robot <lkp(a)intel.com> >> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> >> --- >> >> I don't know if this is the right way of solving this issue but this >> error is now consistently thrown in kbuild compile-test reports w/ >> W=1. >> > > Dear Pierre-Louis Bossart: > Thanks for your modifications. I have fixed the warnings in my latest > KGDB patchset and sent it out, but the merged KGDB patchset is not my > latest version. > I have sent a patch set to complement the missing ports, and I think > these warnings would disappear after this patch has been accepted and > merged. > > Here is the patch to fix this warning: > http://lists.infradead.org/pipermail/linux-riscv/2020-June/000660.html Thank you Vincent, let's drop my hacky patch then.

1 0

[PATCH] ASoC: dt-bindings: simple-card: care missing address #address-cells
by Kuninori Morimoto 02 Jul '20

02 Jul '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current simple-card will get below error, because it doesn't care about #address-cells at some part. DTC Documentation/devicetree/bindings/sound/simple-card.example.dt.yaml Documentation/devicetree/bindings/sound/simple-card.example.dts:171.46-173.15: \ Warning (unit_address_vs_reg): /example-4/sound/simple-audio-card,cpu@0: \ node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:175.37-177.15: \ Warning (unit_address_vs_reg): /example-4/sound/simple-audio-card,cpu@1: \ node has a unit name, but no reg or ranges property ... This patch fixup this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../bindings/sound/simple-card.yaml | 25 ++++++++++++++++++- 1 file changed, 24 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml index cb2bb5fac0e1..6c4c2c6d6d3c 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.yaml +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -208,6 +208,11 @@ patternProperties: reg: maxItems: 1 + "#address-cells": + const: 1 + "#size-cells": + const: 0 + # common properties frame-master: $ref: "#/definitions/frame-master" @@ -288,7 +293,6 @@ examples: #address-cells = <1>; #size-cells = <0>; - simple-audio-card,dai-link@0 { /* I2S - HDMI */ reg = <0>; format = "i2s"; @@ -392,11 +396,15 @@ examples: simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", "ak4642 Playback", "DAI1 Playback"; + #address-cells = <1>; + #size-cells = <0>; dpcmcpu: simple-audio-card,cpu@0 { + reg = <0>; sound-dai = <&rcar_sound 0>; }; simple-audio-card,cpu@1 { + reg = <1>; sound-dai = <&rcar_sound 1>; }; @@ -427,7 +435,12 @@ examples: "pcm3168a Playback", "DAI3 Playback", "pcm3168a Playback", "DAI4 Playback"; + #address-cells = <1>; + #size-cells = <0>; + simple-audio-card,dai-link@0 { + reg = <0>; + format = "left_j"; bitclock-master = <&sndcpu0>; frame-master = <&sndcpu0>; @@ -441,22 +454,30 @@ examples: }; simple-audio-card,dai-link@1 { + reg = <1>; + format = "i2s"; bitclock-master = <&sndcpu1>; frame-master = <&sndcpu1>; convert-channels = <8>; /* TDM Split */ + #address-cells = <1>; + #size-cells = <0>; sndcpu1: cpu@0 { + reg = <0>; sound-dai = <&rcar_sound 1>; }; cpu@1 { + reg = <1>; sound-dai = <&rcar_sound 2>; }; cpu@2 { + reg = <2>; sound-dai = <&rcar_sound 3>; }; cpu@3 { + reg = <3>; sound-dai = <&rcar_sound 4>; }; codec { @@ -468,6 +489,8 @@ examples: }; simple-audio-card,dai-link@2 { + reg = <2>; + format = "i2s"; bitclock-master = <&sndcpu2>; frame-master = <&sndcpu2>; -- 2.17.1

2 3

[PATCH 2/2] ASoC: wm8974: remove unsupported clock mode
by puyou.lu＠gmail.com 02 Jul '20

02 Jul '20

From: Puyou Lu <puyou.lu(a)gmail.com> In DSP_A mode, BIT7 of IFACE should bit 0 according to datasheet (ie. inverted frame clock is not support in this mode). Signed-off-by: Puyou Lu <puyou.lu(a)gmail.com> --- sound/soc/codecs/wm8974.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 764bf93..7cfc896 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -474,6 +474,10 @@ static int wm8974_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0008; break; case SND_SOC_DAIFMT_DSP_A: + if ((fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_IB_IF || + (fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_NB_IF) { + return -EINVAL; + } iface |= 0x00018; break; default: -- 2.7.4

2 1

[PATCH 1/2] ASoC: wm8974: fix Boost Mixer Aux Switch
by puyou.lu＠gmail.com 02 Jul '20

02 Jul '20

From: Puyou Lu <puyou.lu(a)gmail.com> Clear BIT6 of INPPGA means not muted (Switch On). Signed-off-by: Puyou Lu <puyou.lu(a)gmail.com> --- sound/soc/codecs/wm8974.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 06ba365..764bf93 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -186,7 +186,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0), /* Boost mixer */ static const struct snd_kcontrol_new wm8974_boost_mixer[] = { -SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 0), +SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 1), }; /* Input PGA */ -- 2.7.4

2 1

[PATCH AUTOSEL 5.4 13/40] ALSA: hda: Intel: add missing PCI IDs for ICL-H, TGL-H and EKL
by Sasha Levin 02 Jul '20

02 Jul '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit d50313a5a0d803bcf55121a2b82086633060d05e ] 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> Link: https://lore.kernel.org/r/20200617164909.18225-1-pierre-louis.bossart@linux… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 612441508e80a..95b0fdffc504c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2433,6 +2433,9 @@ 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}, @@ -2441,9 +2444,14 @@ 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 }, -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 11/40] ASoC: SOF: Intel: add PCI IDs for ICL-H and TGL-H
by Sasha Levin 02 Jul '20

02 Jul '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit c8d2e2bfaeffa0f914330e8b4e45b986c8d30b58 ] Usually the DSP is not traditionally enabled on H skews but this might be used moving forward. 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> Link: https://lore.kernel.org/r/20200617164755.18104-4-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/sof-pci-dev.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 3f79cd03507c9..9e2ae6e62e7f9 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -410,8 +410,11 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&skl_desc}, #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) - { PCI_DEVICE(0x8086, 0x34C8), + { PCI_DEVICE(0x8086, 0x34C8), /* ICL-LP */ .driver_data = (unsigned long)&icl_desc}, + { PCI_DEVICE(0x8086, 0x3dc8), /* ICL-H */ + .driver_data = (unsigned long)&icl_desc}, + #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) { PCI_DEVICE(0x8086, 0x02c8), @@ -424,8 +427,11 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&cml_desc}, #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) - { PCI_DEVICE(0x8086, 0xa0c8), + { PCI_DEVICE(0x8086, 0xa0c8), /* TGL-LP */ .driver_data = (unsigned long)&tgl_desc}, + { PCI_DEVICE(0x8086, 0x43c8), /* TGL-H */ + .driver_data = (unsigned long)&tgl_desc}, + #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) { PCI_DEVICE(0x8086, 0x4b55), -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 10/40] ASoC: SOF: Intel: add PCI ID for CometLake-S
by Sasha Levin 02 Jul '20

02 Jul '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 258fb4f4c34a0db9d3834aba6784d7b322176bb9 ] Mirror ID added for legacy HDaudio 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> Link: https://lore.kernel.org/r/20200617164755.18104-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/sof-pci-dev.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a778739..3f79cd03507c9 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -420,6 +420,8 @@ static const struct pci_device_id sof_pci_ids[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) { PCI_DEVICE(0x8086, 0x06c8), .driver_data = (unsigned long)&cml_desc}, + { PCI_DEVICE(0x8086, 0xa3f0), /* CML-S */ + .driver_data = (unsigned long)&cml_desc}, #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) { PCI_DEVICE(0x8086, 0xa0c8), -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 18/53] ALSA: hda: Intel: add missing PCI IDs for ICL-H, TGL-H and EKL
by Sasha Levin 02 Jul '20

02 Jul '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit d50313a5a0d803bcf55121a2b82086633060d05e ] 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> Link: https://lore.kernel.org/r/20200617164909.18225-1-pierre-louis.bossart@linux… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 41a03c61a74b3..11ec5c56c80e9 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 }, -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 16/53] ASoC: hdac_hda: fix memleak with regmap not freed on remove
by Sasha Levin 02 Jul '20

02 Jul '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit a94eaccefea1186947c5c5451fcae2245dd7e714 ] 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> Link: https://lore.kernel.org/r/20200617164144.17859-1-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/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 de003acb1951f..473efe9ef998a 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[] = { -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 14/53] ASoC: SOF: Intel: add PCI ID for CometLake-S
by Sasha Levin 02 Jul '20

02 Jul '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 258fb4f4c34a0db9d3834aba6784d7b322176bb9 ] Mirror ID added for legacy HDaudio 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> Link: https://lore.kernel.org/r/20200617164755.18104-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/sof-pci-dev.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index cec631a1389b5..7b1846aeadd59 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -427,6 +427,8 @@ static const struct pci_device_id sof_pci_ids[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) { PCI_DEVICE(0x8086, 0x06c8), .driver_data = (unsigned long)&cml_desc}, + { PCI_DEVICE(0x8086, 0xa3f0), /* CML-S */ + .driver_data = (unsigned long)&cml_desc}, #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) { PCI_DEVICE(0x8086, 0xa0c8), -- 2.25.1

1 0

[PATCH] ASoC: SOF: sof-acpi-dev: fix 'defined but unused' warning
by Pierre-Louis Bossart 02 Jul '20

02 Jul '20

Fix W=1 warning Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/sof-acpi-dev.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/sof-acpi-dev.c b/sound/soc/sof/sof-acpi-dev.c index c5eaaa978054..8aecc46b3647 100644 --- a/sound/soc/sof/sof-acpi-dev.c +++ b/sound/soc/sof/sof-acpi-dev.c @@ -35,7 +35,7 @@ MODULE_PARM_DESC(sof_acpi_debug, "SOF ACPI debug options (0x0 all off)"); #define SOF_ACPI_DISABLE_PM_RUNTIME BIT(0) -#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) +#if IS_ENABLED(CONFIG_ACPI) && IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) static const struct sof_dev_desc sof_acpi_broadwell_desc = { .machines = snd_soc_acpi_intel_broadwell_machines, .resindex_lpe_base = 0, @@ -51,7 +51,7 @@ static const struct sof_dev_desc sof_acpi_broadwell_desc = { }; #endif -#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) +#if IS_ENABLED(CONFIG_ACPI) && IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) /* BYTCR uses different IRQ index */ static const struct sof_dev_desc sof_acpi_baytrailcr_desc = { @@ -133,7 +133,7 @@ static int sof_acpi_probe(struct platform_device *pdev) if (!desc) return -ENODEV; -#if IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) +#if IS_ENABLED(CONFIG_ACPI) && IS_ENABLED(CONFIG_SND_SOC_SOF_BAYTRAIL) if (desc == &sof_acpi_baytrail_desc && soc_intel_is_byt_cr(pdev)) desc = &sof_acpi_baytrailcr_desc; #endif @@ -191,6 +191,7 @@ static int sof_acpi_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_ACPI static const struct acpi_device_id sof_acpi_match[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL) { "INT3438", (unsigned long)&sof_acpi_broadwell_desc }, @@ -202,6 +203,7 @@ static const struct acpi_device_id sof_acpi_match[] = { { } }; MODULE_DEVICE_TABLE(acpi, sof_acpi_match); +#endif /* acpi_driver definition */ static struct platform_driver snd_sof_acpi_driver = { base-commit: a2b782d59c57cb6f673dbb4804ffc500336d7a54 -- 2.25.1

2 1

[PATCH] ASoC: Intel: KeemBay: Fix header guard
by Nathan Chancellor 02 Jul '20

02 Jul '20

Clang warns: In file included from sound/soc/intel/keembay/kmb_platform.c:14: sound/soc/intel/keembay/kmb_platform.h:9:9: warning: 'KMB_PLATFORM_H_' is used as a header guard here, followed by #define of a different macro [-Wheader-guard] #ifndef KMB_PLATFORM_H_ ^~~~~~~~~~~~~~~ sound/soc/intel/keembay/kmb_platform.h:10:9: note: 'KMB_PLATFORMP_H_' is defined here; did you mean 'KMB_PLATFORM_H_'? #define KMB_PLATFORMP_H_ ^~~~~~~~~~~~~~~~ KMB_PLATFORM_H_ 1 warning generated. Fix the typo so that the header guard works as intended. Fixes: c5477e966728 ("ASoC: Intel: Add KeemBay platform driver") Link: https://github.com/ClangBuiltLinux/linux/issues/1053 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> --- sound/soc/intel/keembay/kmb_platform.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/keembay/kmb_platform.h b/sound/soc/intel/keembay/kmb_platform.h index 29600652d8f4..6bf221aa8fff 100644 --- a/sound/soc/intel/keembay/kmb_platform.h +++ b/sound/soc/intel/keembay/kmb_platform.h @@ -7,7 +7,7 @@ */ #ifndef KMB_PLATFORM_H_ -#define KMB_PLATFORMP_H_ +#define KMB_PLATFORM_H_ #include <linux/bits.h> #include <linux/bitfield.h> base-commit: 27f70ec4fa0e0f419031f1b8d61b1a788244e313 -- 2.27.0

4 5

[PATCH] ASoC: amd: Rectifying Unbalanced pm_runtime_enable! issue
by Ravulapati Vishnu vardhan rao 02 Jul '20

02 Jul '20

When snd_pci_acp3x driver loads we see: WARNING kernel:snd_pci_acp3x 0000:04:00.5: Unbalanced pm_runtime_enable! at boot time. same can be observed in /var/log/messages/. Modifying pm runtime sequence for fixing unbalanced pm issue. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index f25ce50f1a90..ebf4388b6262 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -232,9 +232,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pm_runtime_set_autosuspend_delay(&pci->dev, 2000); pm_runtime_use_autosuspend(&pci->dev); - pm_runtime_set_active(&pci->dev); pm_runtime_put_noidle(&pci->dev); - pm_runtime_enable(&pci->dev); pm_runtime_allow(&pci->dev); return 0; @@ -303,7 +301,7 @@ static void snd_acp3x_remove(struct pci_dev *pci) ret = acp3x_deinit(adata->acp3x_base); if (ret) dev_err(&pci->dev, "ACP de-init failed\n"); - pm_runtime_disable(&pci->dev); + pm_runtime_forbid(&pci->dev); pm_runtime_get_noresume(&pci->dev); pci_disable_msi(pci); pci_release_regions(pci); -- 2.17.1

2 1

[PATCH] ASoC: rt5682: cancel jack_detect_work if hs_jack is set to null even soundwire mode
by Oder Chiou 02 Jul '20

02 Jul '20

Base on https://patchwork.kernel.org/patch/11237953/ Soundwire mode also should follow it. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5682.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index e9514c81b9ba..de40b6cd16cf 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -992,16 +992,17 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component, rt5682->hs_jack = hs_jack; - if (!rt5682->is_sdw) { - if (!hs_jack) { - regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, - RT5682_JD1_EN_MASK, RT5682_JD1_DIS); - regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, - RT5682_POW_JDH | RT5682_POW_JDL, 0); - cancel_delayed_work_sync(&rt5682->jack_detect_work); - return 0; - } + if (!hs_jack) { + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK, RT5682_JD1_DIS); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_JDH | RT5682_POW_JDL, 0); + cancel_delayed_work_sync(&rt5682->jack_detect_work); + return 0; + } + + if (!rt5682->is_sdw) { switch (rt5682->pdata.jd_src) { case RT5682_JD1: snd_soc_component_update_bits(component, -- 2.27.0

2 1

[PATCH 0/2] ASoC: Intel: atom: fix kernel-doc and W=1 warnings
by Pierre-Louis Bossart 02 Jul '20

02 Jul '20

Yet another series to fix broken kernel-doc and mark unused variables as such. Pierre-Louis Bossart (2): ASoC: Intel: atom: fix kernel-doc ASoC: Intel: atom: fix 'defined but not used' warning sound/soc/intel/atom/sst-atom-controls.c | 65 ++++++++++++++++-------- sound/soc/intel/atom/sst/sst_loader.c | 4 ++ sound/soc/intel/atom/sst/sst_stream.c | 43 +++++++++------- 3 files changed, 74 insertions(+), 38 deletions(-) base-commit: a2b782d59c57cb6f673dbb4804ffc500336d7a54 -- 2.25.1

2 5

[PATCH] ASoC: amd: Enable interrupt in dma_open
by Akshu Agrawal 02 Jul '20

02 Jul '20

Fixes interrupt enable condition check with which now interrupt gets enabled in dma_open. Prior to this patch it was getting enabled in runtime_resume only. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index e6386de20ac7..5bd458e0fe31 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -238,7 +238,7 @@ static int acp3x_dma_open(struct snd_soc_component *component, } if (!adata->play_stream && !adata->capture_stream && - adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) + !adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); i2s_data->acp3x_base = adata->acp3x_base; -- 2.20.1

2 1

[PATCH] ASoC: dt-bindings: keembay-i2s: Fix reg descriptions
by Fabio Estevam 02 Jul '20

02 Jul '20

intel,keembay-i2s has two register regions: - I2S registers - I2S gen configuration Describe these regions accordingly to fix the following warning seen with 'make dt_binding_check': Documentation/devicetree/bindings/sound/intel,keembay-i2s.example.dt.yaml: example-0: i2s@20140000:reg:0: [538181632, 512, 539623588, 4] is too long Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- .../devicetree/bindings/sound/intel,keembay-i2s.yaml | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml index 175e4fb0c315..2e0bbc1c868a 100644 --- a/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml +++ b/Documentation/devicetree/bindings/sound/intel,keembay-i2s.yaml @@ -23,7 +23,8 @@ properties: reg: items: - - description: I2S configuration + - description: I2S registers + - description: I2S gen configuration reg-names: items: @@ -60,7 +61,8 @@ examples: i2s3: i2s@20140000 { compatible = "intel,keembay-i2s"; #sound-dai-cells = <0>; - reg = <0x20140000 0x200 0x202a00a4 0x4>; + reg = <0x20140000 0x200>, /* I2S registers */ + <0x202a00a4 0x4>; /* I2S gen configuration */ reg-names = "i2s-regs", "i2s_gen_cfg"; interrupts = <GIC_SPI 120 IRQ_TYPE_LEVEL_HIGH>; clock-names = "osc", "apb_clk"; -- 2.17.1

2 1

[PATCH] ASoC: fsl_sai: Refine regcache usage with pm runtime
by Shengjiu Wang 02 Jul '20

02 Jul '20

When there is dedicated power domain bound with device, after probing the power will be disabled, then registers are not accessible in fsl_sai_dai_probe(), so regcache only need to be enabled in end of probe() and regcache_mark_dirty should be moved to pm runtime resume callback function. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 9d436b0c5718..a22562f2df47 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1016,6 +1016,7 @@ static int fsl_sai_probe(struct platform_device *pdev) platform_set_drvdata(pdev, sai); pm_runtime_enable(&pdev->dev); + regcache_cache_only(sai->regmap, true); ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, &fsl_sai_dai, 1); @@ -1107,7 +1108,6 @@ static int fsl_sai_runtime_suspend(struct device *dev) clk_disable_unprepare(sai->bus_clk); regcache_cache_only(sai->regmap, true); - regcache_mark_dirty(sai->regmap); return 0; } @@ -1137,6 +1137,7 @@ static int fsl_sai_runtime_resume(struct device *dev) } regcache_cache_only(sai->regmap, false); + regcache_mark_dirty(sai->regmap); regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR); regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), FSL_SAI_CSR_SR); usleep_range(1000, 2000); -- 2.21.0

3 2

[PATCH] ASoC: rockchip: add format and rate constraints on rk3399
by Yu-Hsuan Hsu 02 Jul '20

02 Jul '20

S8 and S24 formats does not work on this machine driver so force to use S16_LE instead. In addition, add constraint to limit the max value of rate because the rate higher than 96000(172000, 192000) is not stable either. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/rockchip/rk3399_gru_sound.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index f45e5aaa4b302..9539b0d024fed 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -219,19 +219,32 @@ static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream, return 0; } +static int rockchip_sound_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + return snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, + 8000, 96000); +} + static const struct snd_soc_ops rockchip_sound_max98357a_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_max98357a_hw_params, }; static const struct snd_soc_ops rockchip_sound_rt5514_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_rt5514_hw_params, }; static const struct snd_soc_ops rockchip_sound_da7219_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_da7219_hw_params, }; static const struct snd_soc_ops rockchip_sound_dmic_ops = { + .startup = rockchip_sound_startup, .hw_params = rockchip_sound_dmic_hw_params, }; -- 2.27.0.212.ge8ba1cc988-goog

2 3

[PATCH v4 0/3] ASoC: ti: Add support for audio on J721e EVM
by Peter Ujfalusi 02 Jul '20

02 Jul '20

Hi, Changes since v3: - Fix the single clock source handling and typo Changes since v2: - DT binding: - use proper (?) patch subject for the binding docuemtn patch - drop pll4 and pll15 from DT - driver should check the rate via clk_get_parent. If it is not available (as it is not currently) then use the match_data provided rates. - add simple explanation for the clocking setup - Use descriptive names for clocks: cpb/ivi-mcasp-auxclk and cpb/ivi-codec-scki - dt_binding_check shows no errors/warnings - ASoC machine driver: - Try to read the PLL4/15 rate with clk API (parent of the two clock divider) if it is not available then use the match_data provided numbers. - Support for single PLL setup Changes since v1: - Fixed DT binding documentation errors - Rebased on ASoC head and updated the driver to compile and work This series adds support for the analog audio setup on the j721e EVM. The audio setup of the EVM is: Common Processor Board (CPB): McASP10 <-> pcm3168a Infotainment Expansion Board (IVI): McASP0 <-> 2x pcm3168a Both CPB and IVI wired in parallel serializer setup. The first patch adds the stream_name for McASP driver as it is needed in multicodec (and would be needed in DPCM) setup for proper DAPM handling. The second patch adds two DT schema, one for the cpb and one for the cpb+ivi card. Regards, Peter --- Peter Ujfalusi (3): ASoC: ti: davinci-mcasp: Specify stream_name for playback/capture ASoC: dt-bindings: Add documentation for TI j721e EVM (CPB and IVI) ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI) .../bindings/sound/ti,j721e-cpb-audio.yaml | 95 ++ .../sound/ti,j721e-cpb-ivi-audio.yaml | 150 +++ sound/soc/ti/Kconfig | 8 + sound/soc/ti/Makefile | 2 + sound/soc/ti/davinci-mcasp.c | 3 + sound/soc/ti/j721e-evm.c | 896 ++++++++++++++++++ 6 files changed, 1154 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml create mode 100644 sound/soc/ti/j721e-evm.c -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 4

[PATCH v3] ASoC: fsl_asrc: Add an option to select internal ratio mode
by Shengjiu Wang 02 Jul '20

02 Jul '20

The ASRC not only supports ideal ratio mode, but also supports internal ratio mode. For internal rato mode, the rate of clock source should be divided with no remainder by sample rate, otherwise there is sound distortion. Add function fsl_asrc_select_clk() to find proper clock source for internal ratio mode, if the clock source is available then internal ratio mode will be selected. With change, the ideal ratio mode is not the only option for user. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3 - convert fsl_asrc_select_clk to void type changes in v2 - update according to Nicolin's comments sound/soc/fsl/fsl_asrc.c | 46 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 44 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 95f6a9617b0b..462ce9f9ab48 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -582,11 +582,51 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints); } +/* Select proper clock source for internal ratio mode */ +static void fsl_asrc_select_clk(struct fsl_asrc_priv *asrc_priv, + struct fsl_asrc_pair *pair, + int in_rate, + int out_rate) +{ + struct fsl_asrc_pair_priv *pair_priv = pair->private; + struct asrc_config *config = pair_priv->config; + int rate[2], select_clk[2]; /* Array size 2 means IN and OUT */ + int clk_rate, clk_index; + int i = 0, j = 0; + + rate[IN] = in_rate; + rate[OUT] = out_rate; + + /* Select proper clock source for internal ratio mode */ + for (j = 0; j < 2; j++) { + for (i = 0; i < ASRC_CLK_MAP_LEN; i++) { + clk_index = asrc_priv->clk_map[j][i]; + clk_rate = clk_get_rate(asrc_priv->asrck_clk[clk_index]); + /* Only match a perfect clock source with no remainder */ + if (clk_rate != 0 && (clk_rate / rate[j]) <= 1024 && + (clk_rate % rate[j]) == 0) + break; + } + + select_clk[j] = i; + } + + /* Switch to ideal ratio mode if there is no proper clock source */ + if (select_clk[IN] == ASRC_CLK_MAP_LEN || select_clk[OUT] == ASRC_CLK_MAP_LEN) { + select_clk[IN] = INCLK_NONE; + select_clk[OUT] = OUTCLK_ASRCK1_CLK; + } + + config->inclk = select_clk[IN]; + config->outclk = select_clk[OUT]; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct fsl_asrc *asrc = snd_soc_dai_get_drvdata(dai); + struct fsl_asrc_priv *asrc_priv = asrc->private; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; struct fsl_asrc_pair_priv *pair_priv = pair->private; @@ -605,8 +645,6 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.pair = pair->index; config.channel_num = channels; - config.inclk = INCLK_NONE; - config.outclk = OUTCLK_ASRCK1_CLK; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config.input_format = params_format(params); @@ -620,6 +658,10 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } + fsl_asrc_select_clk(asrc_priv, pair, + config.input_sample_rate, + config.output_sample_rate); + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); -- 2.21.0

4 3

[PATCH] riscv: asm/gdb_xml.h: use __maybe_used to make W=1 warnings go away
by Pierre-Louis Bossart 01 Jul '20

01 Jul '20

0day/kbuild reports warnings with the ASoC codecs compiled with W=1. In file included from arch/riscv/include/asm/kgdb.h:109, from include/linux/kgdb.h:20, from include/linux/fb.h:5, from include/drm/drm_crtc.h:31, from sound/soc/codecs/hdmi-codec.c:19: arch/riscv/include/asm/gdb_xml.h:23:19: warning: 'riscv_gdb_stub_cpuxml' defined but not used [-Wunused-const-variable=] 23 | static const char riscv_gdb_stub_cpuxml[2048] = | ^~~~~~~~~~~~~~~~~~~~~ arch/riscv/include/asm/gdb_xml.h:16:19: warning: 'riscv_gdb_stub_target_desc' defined but not used [-Wunused-const-variable=] 16 | static const char riscv_gdb_stub_target_desc[256] = | ^~~~~~~~~~~~~~~~~~~~~~~~~~ arch/riscv/include/asm/gdb_xml.h:13:19: warning: 'gdb_xfer_read_cpuxml' defined but not used [-Wunused-const-variable=] 13 | static const char gdb_xfer_read_cpuxml[39] = | ^~~~~~~~~~~~~~~~~~~~ arch/riscv/include/asm/gdb_xml.h:10:19: warning: 'gdb_xfer_read_target' defined but not used [-Wunused-const-variable=] 10 | static const char gdb_xfer_read_target[31] = "qXfer:features:read:target.xml:"; | ^~~~~~~~~~~~~~~~~~~~ arch/riscv/include/asm/gdb_xml.h:7:19: warning: 'riscv_gdb_stub_feature' defined but not used [-Wunused-const-variable=] 7 | static const char riscv_gdb_stub_feature[64] = | ^~~~~~~~~~~~~~~~~~~~~~ Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- I don't know if this is the right way of solving this issue but this error is now consistently thrown in kbuild compile-test reports w/ W=1. arch/riscv/include/asm/gdb_xml.h | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/arch/riscv/include/asm/gdb_xml.h b/arch/riscv/include/asm/gdb_xml.h index 041b45f5b997..c28fc1a1d230 100644 --- a/arch/riscv/include/asm/gdb_xml.h +++ b/arch/riscv/include/asm/gdb_xml.h @@ -4,23 +4,23 @@ #define __ASM_GDB_XML_H_ #define kgdb_arch_gdb_stub_feature riscv_gdb_stub_feature -static const char riscv_gdb_stub_feature[64] = +static __maybe_unused const char riscv_gdb_stub_feature[64] = "PacketSize=800;qXfer:features:read+;"; -static const char gdb_xfer_read_target[31] = "qXfer:features:read:target.xml:"; +static __maybe_unused const char gdb_xfer_read_target[31] = "qXfer:features:read:target.xml:"; #ifdef CONFIG_64BIT -static const char gdb_xfer_read_cpuxml[39] = +static __maybe_unused const char gdb_xfer_read_cpuxml[39] = "qXfer:features:read:riscv-64bit-cpu.xml"; -static const char riscv_gdb_stub_target_desc[256] = +static __maybe_unused const char riscv_gdb_stub_target_desc[256] = "l<?xml version=\"1.0\"?>" "<!DOCTYPE target SYSTEM \"gdb-target.dtd\">" "<target>" "<xi:include href=\"riscv-64bit-cpu.xml\"/>" "</target>"; -static const char riscv_gdb_stub_cpuxml[2048] = +static __maybe_unused const char riscv_gdb_stub_cpuxml[2048] = "l<?xml version=\"1.0\"?>" "<!DOCTYPE feature SYSTEM \"gdb-target.dtd\">" "<feature name=\"org.gnu.gdb.riscv.cpu\">" -- 2.25.1

1 0

[PATCH] ASoC: Intel: bxt-da7219-max98357a: support MAX98390 speaker amp
by Brent Lu 01 Jul '20

01 Jul '20

Support MAX98390 speaker amplifier on cometlake platform. Driver now detects amplifier type in the probe function and installs corresponding controls and DAPM widgets/routes in the late_probe function. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 20 ++++ sound/soc/intel/boards/bxt_da7219_max98357a.c | 129 ++++++++++++++++++++-- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 4 +- 3 files changed, 139 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 3d820e1..b3b863e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -291,9 +291,17 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC select SND_SOC_DMIC select SND_SOC_HDAC_HDMI +config SND_SOC_INTEL_DA7219_MAX98390_GENERIC + tristate + select SND_SOC_DA7219 + select SND_SOC_MAX98390 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON tristate select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC + select SND_SOC_INTEL_DA7219_MAX98390_GENERIC if SND_SOC_INTEL_APL @@ -309,6 +317,18 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". +config SND_SOC_INTEL_BXT_DA7219_MAX98390_MACH + tristate "Broxton with DA7219 and MAX98390 in I2S Mode" + depends on I2C && ACPI && GPIOLIB + depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on SND_HDA_CODEC_HDMI + select SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON + help + This adds support for ASoC machine driver for Broxton-P platforms + with DA7219 + MAX98390 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_BXT_RT298_MACH tristate "Broxton with RT298 I2S mode" depends on I2C && ACPI && GPIOLIB diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 44016c1..12f07e1 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -25,9 +25,14 @@ #define BXT_DIALOG_CODEC_DAI "da7219-hifi" #define BXT_MAXIM_CODEC_DAI "HiFi" +#define MAX98390_DEV0_NAME "i2c-MX98390:00" +#define MAX98390_DEV1_NAME "i2c-MX98390:01" #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 +#define SPKAMP_MAX98357A 1 +#define SPKAMP_MAX98390 2 + static struct snd_soc_jack broxton_headset; static struct snd_soc_jack broxton_hdmi[3]; @@ -40,6 +45,7 @@ struct bxt_hdmi_pcm { struct bxt_card_private { struct list_head hdmi_pcm_list; bool common_hdmi_codec_drv; + int spkamp; }; enum { @@ -85,13 +91,20 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, static const struct snd_kcontrol_new broxton_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_kcontrol_new max98357a_controls[] = { SOC_DAPM_PIN_SWITCH("Spk"), }; +static const struct snd_kcontrol_new max98390_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + static const struct snd_soc_dapm_widget broxton_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), SND_SOC_DAPM_MIC("SoC DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), @@ -100,14 +113,20 @@ static const struct snd_soc_dapm_widget broxton_widgets[] = { platform_clock_control, SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU), }; +static const struct snd_soc_dapm_widget max98357a_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_widget max98390_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + static const struct snd_soc_dapm_route audio_map[] = { /* HP jack connectors - unknown if we have jack detection */ {"Headphone Jack", NULL, "HPL"}, {"Headphone Jack", NULL, "HPR"}, - /* speaker */ - {"Spk", NULL, "Speaker"}, - /* other jacks */ {"MIC", NULL, "Headset Mic"}, @@ -134,6 +153,17 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Headset Mic", NULL, "Platform Clock" }, }; +static const struct snd_soc_dapm_route max98357a_routes[] = { + /* speaker */ + {"Spk", NULL, "Speaker"}, +}; + +static const struct snd_soc_dapm_route max98390_routes[] = { + /* Speaker */ + {"Left Spk", NULL, "Left BE_OUT"}, + {"Right Spk", NULL, "Right BE_OUT"}, +}; + static const struct snd_soc_dapm_route broxton_map[] = { {"HiFi Playback", NULL, "ssp5 Tx"}, {"ssp5 Tx", NULL, "codec0_out"}, @@ -404,6 +434,10 @@ SND_SOC_DAILINK_DEF(ssp5_pin, SND_SOC_DAILINK_DEF(ssp5_codec, DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", BXT_MAXIM_CODEC_DAI))); +SND_SOC_DAILINK_DEF(max98390_codec, + DAILINK_COMP_ARRAY( + /* Left */ COMP_CODEC(MAX98390_DEV0_NAME, "max98390-aif1"), + /* Right */ COMP_CODEC(MAX98390_DEV1_NAME, "max98390-aif1"))); SND_SOC_DAILINK_DEF(ssp1_pin, DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); @@ -601,15 +635,69 @@ static struct snd_soc_dai_link broxton_dais[] = { }, }; +static struct snd_soc_codec_conf max98390_codec_confs[] = { + { + .dlc = COMP_CODEC_CONF(MAX98390_DEV0_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX98390_DEV1_NAME), + .name_prefix = "Right", + }, +}; + #define NAME_SIZE 32 static int bxt_card_late_probe(struct snd_soc_card *card) { struct bxt_card_private *ctx = snd_soc_card_get_drvdata(card); struct bxt_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; - int err, i = 0; + const struct snd_kcontrol_new *controls; + const struct snd_soc_dapm_widget *widgets; + const struct snd_soc_dapm_route *routes; + int num_controls, num_widgets, num_routes, ret, i = 0; char jack_name[NAME_SIZE]; + switch (ctx->spkamp) { + case SPKAMP_MAX98357A: + controls = max98357a_controls; + num_controls = ARRAY_SIZE(max98357a_controls); + widgets = max98357a_widgets; + num_widgets = ARRAY_SIZE(max98357a_widgets); + routes = max98357a_routes; + num_routes = ARRAY_SIZE(max98357a_routes); + break; + case SPKAMP_MAX98390: + controls = max98390_controls; + num_controls = ARRAY_SIZE(max98390_controls); + widgets = max98390_widgets; + num_widgets = ARRAY_SIZE(max98390_widgets); + routes = max98390_routes; + num_routes = ARRAY_SIZE(max98390_routes); + break; + default: + dev_err(card->dev, "Invalid speaker amplifier %d\n", ctx->spkamp); + break; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, widgets, num_widgets); + if (ret) { + dev_err(card->dev, "Fail to new widgets\n"); + return ret; + } + + ret = snd_soc_add_card_controls(card, controls, num_controls); + if (ret) { + dev_err(card->dev, "Fail to add controls\n"); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, routes, num_routes); + if (ret) { + dev_err(card->dev, "Fail to add routes\n"); + return ret; + } + if (soc_intel_is_glk()) snd_soc_dapm_add_routes(&card->dapm, gemini_map, ARRAY_SIZE(gemini_map)); @@ -631,17 +719,17 @@ static int bxt_card_late_probe(struct snd_soc_card *card) component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, + ret = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &broxton_hdmi[i], NULL, 0); - if (err) - return err; + if (ret) + return ret; - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + ret = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &broxton_hdmi[i]); - if (err < 0) - return err; + if (ret < 0) + return ret; i++; } @@ -678,6 +766,11 @@ static int broxton_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + if (acpi_dev_present("MX98390", NULL, -1)) + ctx->spkamp = SPKAMP_MAX98390; + else + ctx->spkamp = SPKAMP_MAX98357A; + broxton_audio_card.dev = &pdev->dev; snd_soc_card_set_drvdata(&broxton_audio_card, ctx); if (soc_intel_is_glk()) { @@ -702,7 +795,13 @@ static int broxton_audio_probe(struct platform_device *pdev) } else if (soc_intel_is_cml()) { unsigned int i; - broxton_audio_card.name = "cmlda7219max"; + if (ctx->spkamp == SPKAMP_MAX98390) { + broxton_audio_card.name = "cml_max98390_da7219"; + + broxton_audio_card.codec_conf = max98390_codec_confs; + broxton_audio_card.num_configs = ARRAY_SIZE(max98390_codec_confs); + } else + broxton_audio_card.name = "cmlda7219max"; for (i = 0; i < ARRAY_SIZE(broxton_dais); i++) { /* MAXIM_CODEC is connected to SSP1. */ @@ -710,6 +809,11 @@ static int broxton_audio_probe(struct platform_device *pdev) BXT_MAXIM_CODEC_DAI)) { broxton_dais[i].name = "SSP1-Codec"; broxton_dais[i].cpus->dai_name = "SSP1 Pin"; + + if (ctx->spkamp == SPKAMP_MAX98390) { + broxton_dais[i].codecs = max98390_codec; + broxton_dais[i].num_codecs = ARRAY_SIZE(max98390_codec); + } } /* DIALOG_CODEC is connected to SSP0 */ else if (!strcmp(broxton_dais[i].codecs->dai_name, @@ -759,6 +863,7 @@ MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); MODULE_AUTHOR("Conrad Cooke <conrad.cooke(a)intel.com>"); MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); +MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:bxt_da7219_max98357a"); MODULE_ALIAS("platform:glk_da7219_max98357a"); diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index cdea0c0..7dc9ba9 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -15,8 +15,8 @@ static struct snd_soc_acpi_codecs rt1011_spk_codecs = { }; static struct snd_soc_acpi_codecs max98357a_spk_codecs = { - .num_codecs = 1, - .codecs = {"MX98357A"} + .num_codecs = 2, + .codecs = {"MX98357A", "MX98390"} }; /* -- 2.7.4

3 3

Translation for alsa-utils
by GitHub issues - opened 01 Jul '20

01 Jul '20

alsa-project/alsa-utils issue #41 was opened from Porrumentzio: I found the [po folder](https://github.com/alsa-project/alsa-utils/tree/master/po) but seems that there are not all the translations (I know there's a Spanish localization for example). Is it enough to translate the po files and PR them? Or is there any other procedure? Issue URL : https://github.com/alsa-project/alsa-utils/issues/41 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2] ALSA: hda/hdmi: Add Intel silent stream support
by Harsha Priya 01 Jul '20

01 Jul '20

External HDMI receivers have analog circuitry that needs to be powered-on when exiting standby, and a mechanism to detect PCM v. IEC61937 data. These two steps take time and up to 2-3 seconds of audio may be muted when starting playback. Intel hardware can keep the link active with a 'silent stream', so that the receiver does not go through those two steps when valid audio is transmitted. This mechanism relies on an info packet and preventing the codec from going to D3, which will increase the platform static power consumption. The info packet assumes a basic 2ch stereo, and the silent stream is enabled when connecting a monitor. In case of format changes the detection of PCM v. IEC61937 needs to be re-run. In this case there is no way to avoid the 2-3s mute. The silent stream is enabled with a Kconfig option, as well as a kernel parameter should there be a need to override the build time default. This approach is used based on the power_save capability as an example, but in the future, it may be used with a kcontrol, depending on UCM support for HDaudio legacy. Silent stream is supported in Intel platforms Skylake and beyond. Tested HDMI plug-out plug-in from Intel Cometlake based Chromebook connected to few different monitors. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Emmanuel Jillela <emmanuel.jillela(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- Change History: v2: - Kconfig symbol depends on SND_HDA_INTEL. - Added valid eld check to along side of silent stream enable check, under which silent stream is enabled. - Error prints added for codec power up, power down functions. - The silent stream is enabled with a Kconfig/module parameter following the power_save capability as an example. But in the future, it may be used with a kcontrol, depending on UCM support for HDaudio legacy. v1: - Initial version. sound/pci/hda/Kconfig | 16 ++++++++++++++ sound/pci/hda/patch_hdmi.c | 45 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 61 insertions(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 7ba542e45a3d..1c4758d58fc3 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -232,4 +232,20 @@ config SND_HDA_POWER_SAVE_DEFAULT endif +config SND_HDA_INTEL_HDMI_SILENT_STREAM + bool "Enable Silent Stream always for HDMI" + depends on SND_HDA_INTEL + help + Intel hardware has a feature called 'silent stream', that + keeps external HDMI receiver's analog circuitry powered on + avoiding 2-3 sec silence during playback start. This mechanism + relies on an info packet and preventing the codec from going to + D3. (increasing the platform static power consumption when a + HDMI receiver is plugged-in). 2-3 sec silence at the playback + start is expected whenever there is format change. (default is + 2 channel format). + Say Y to enable Silent Stream feature. + +endif + endmenu diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index fbd7cc6026d8..68b81ea176c6 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -42,6 +42,11 @@ static bool enable_acomp = true; module_param(enable_acomp, bool, 0444); MODULE_PARM_DESC(enable_acomp, "Enable audio component binding (default=yes)"); +static bool enable_silent_stream = +IS_ENABLED(CONFIG_SND_HDA_INTEL_HDMI_SILENT_STREAM); +module_param(enable_silent_stream, bool, 0644); +MODULE_PARM_DESC(enable_silent_stream, "Enable Silent Stream for HDMI devices"); + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -1634,21 +1639,61 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, snd_hda_power_down_pm(codec); } +static void silent_stream_enable(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + codec_dbg(codec, "hdmi: enabling silent stream for NID %d\n", + per_pin->pin_nid); + + mutex_lock(&per_pin->lock); + if (!per_pin->channels) + per_pin->channels = 2; + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); + mutex_unlock(&per_pin->lock); +} + /* update ELD and jack state via audio component */ static void sync_eld_via_acomp(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin) { struct hdmi_spec *spec = codec->spec; struct hdmi_eld *eld = &spec->temp_eld; + bool monitor_prev, monitor_next; mutex_lock(&per_pin->lock); eld->monitor_present = false; + monitor_prev = per_pin->sink_eld.monitor_present; eld->eld_size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, per_pin->dev_id, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); eld->eld_valid = (eld->eld_size > 0); update_eld(codec, per_pin, eld, 0); + monitor_next = per_pin->sink_eld.monitor_present; mutex_unlock(&per_pin->lock); + + /* + * Power-up will call hdmi_present_sense, so the PM calls + * have to be done without mutex held. + */ + + if (enable_silent_stream && eld->eld_valid) { + int pm_ret; + + if (!monitor_prev && monitor_next && eld->eld_valid) { + pm_ret = snd_hda_power_up_pm(codec); + if (pm_ret < 0) + codec_err(codec, + "Monitor plugged-in, Failed to power up codec ret=[%d]\n", + pm_ret); + silent_stream_enable(codec, per_pin); + } else if (monitor_prev && !monitor_next && !eld->eld_valid) { + pm_ret = snd_hda_power_down_pm(codec); + if (pm_ret < 0) + codec_err(codec, + "Monitor plugged-out, Failed to power down codec ret=[%d]\n", + pm_ret); + } + } } static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) -- 2.17.1

3 2

[PATCH] ALSA: hda/realtek: Serialize setting GPIO LED
by Kai-Heng Feng 01 Jul '20

01 Jul '20

If a system has two GPIO controlled LED, one for mute and another one for micmute, and both of them are on before system suspend, sometimes one of them won't be turned off by system suspend. The codec doesn't seem to be able to control multiple GPIO LEDs at the same time, so introduce a new mutex to serialize setting the LED, to prevent the issue from happening. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- include/sound/hda_codec.h | 1 + sound/pci/hda/hda_codec.c | 1 + sound/pci/hda/patch_realtek.c | 3 +++ 3 files changed, 5 insertions(+) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index d16a4229209b..3a1792bbb7ac 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -206,6 +206,7 @@ struct hda_codec { struct mutex spdif_mutex; struct mutex control_mutex; + struct mutex led_mutex; struct snd_array spdif_out; unsigned int spdif_in_enable; /* SPDIF input enable? */ const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7e3ae4534df9..ec477cd8afed 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -946,6 +946,7 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, codec->addr = codec_addr; mutex_init(&codec->spdif_mutex); mutex_init(&codec->control_mutex); + mutex_init(&codec->led_mutex); snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53e0eef8b042..96dac365fbb8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4101,7 +4101,10 @@ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, { if (polarity) enabled = !enabled; + + mutex_lock(&codec->led_mutex); alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */ + mutex_unlock(&codec->led_mutex); } /* turn on/off mute LED via GPIO per vmaster hook */ -- 2.17.1

2 2

[PATCH] ASoC: rt1015: DAC volume range modification.
by jack.yu＠realtek.com 01 Jul '20

01 Jul '20

From: Jack Yu <jack.yu(a)realtek.com> RT1015 dac volume range modification. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 4 ++-- sound/soc/codecs/rt1015.h | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 548f68649064..90bd7e22ec3b 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -401,7 +401,7 @@ static bool rt1015_readable_register(struct device *dev, unsigned int reg) } } -static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9525, 75, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -95625, 375, 0); static const char * const rt1015_din_source_select[] = { "Left", @@ -517,7 +517,7 @@ static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new rt1015_snd_controls[] = { SOC_SINGLE_TLV("DAC Playback Volume", RT1015_DAC1, RT1015_DAC_VOL_SFT, - 127, 0, dac_vol_tlv), + 255, 0, dac_vol_tlv), SOC_DOUBLE("DAC Playback Switch", RT1015_DAC3, RT1015_DA_MUTE_SFT, RT1015_DVOL_MUTE_FLAG_SFT, 1, 1), SOC_ENUM_EXT("Boost Mode", rt1015_boost_mode_enum, diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 7bd159e8f958..cc1ae33ca753 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -214,8 +214,8 @@ #define RT1015_ID_VERB 0x1 /* 0x0102 */ -#define RT1015_DAC_VOL_MASK (0x7f << 9) -#define RT1015_DAC_VOL_SFT 9 +#define RT1015_DAC_VOL_MASK (0xff << 8) +#define RT1015_DAC_VOL_SFT 8 /* 0x0104 */ #define RT1015_DAC_CLK (0x1 << 13) -- 2.27.0

1 0

Sof hda dsp fixup
by GitHub pull_request - opened 01 Jul '20

01 Jul '20

alsa-project/alsa-ucm-conf pull request #38 was opened from libinyang: This PR does such fixup: fixup typo "codec" => "codecs" sof-hda-dsp should include sof-hda-dsp/Hdmi.conf not HDA-Intel/Hdmi.conf Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/38 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/38.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/5] soundwire: handle stream at the dailink level
by Bard Liao 01 Jul '20

01 Jul '20

Currently, stream is handled at the dai level. But we have to handle stream at the dailink level in the multi-cpu dailink usage. Pierre-Louis Bossart (5): soundwire: intel: implement get_sdw_stream() operations soundwire: stream: add helper to startup/shutdown streams soundwire: intel: remove stream allocation/free soundwire: cadence: allocate/free dma_data in set_sdw_stream soundwire: intel: don't free dma_data in DAI shutdown drivers/soundwire/cadence_master.c | 52 ++++++++++++----- drivers/soundwire/intel.c | 82 +++++++-------------------- drivers/soundwire/stream.c | 90 ++++++++++++++++++++++++++++++ include/linux/soundwire/sdw.h | 2 + 4 files changed, 149 insertions(+), 77 deletions(-) -- 2.17.1

3 8

[GIT PULL] ASoC fixes for v5.8-rc3
by Mark Brown 30 Jun '20

30 Jun '20

The following changes since commit f141a422159a199f4c8dedb7e0df55b3b2cf16cd: ASoC: rockchip: Fix a reference count leak. (2020-06-18 17:21:58 +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-rc3 for you to fetch changes up to bc5c7f55f5ea91e137fc7939435ed2e2bb6e5a15: AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module (2020-06-25 15:06:56 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.8 A few small driver specific fixes, nothing particularly dramatic. ---------------------------------------------------------------- Jaroslav Kysela (1): AsoC: amd: add missing snd- module prefix to the acp3x-rn driver kernel module Shengjiu Wang (2): ASoC: fsl_mqs: Don't check clock is NULL before calling clk API ASoC: fsl_mqs: Fix unchecked return value for clk_prepare_enable Shuming Fan (1): ASoC: rt5682: fix the pop noise while OMTP type headset plugin sound/soc/amd/renoir/Makefile | 7 ++++--- sound/soc/codecs/rt5682.c | 9 ++++++++- sound/soc/fsl/fsl_mqs.c | 23 ++++++++++++++--------- 3 files changed, 26 insertions(+), 13 deletions(-)

2 1

[PATCH] ALSA: usb-audio: Fix some typos
by Alexander Tsoy 30 Jun '20

30 Jun '20

Fix the following typos in comments and in the code: - KHz -> kHz - procssed -> processed Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/mixer_s1810c.c | 6 +++--- sound/usb/mixer_scarlett_gen2.c | 4 ++-- sound/usb/pcm.c | 4 ++-- sound/usb/quirks.c | 14 +++++++------- 4 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c index 6483e47bafd0..c53a9773f310 100644 --- a/sound/usb/mixer_s1810c.c +++ b/sound/usb/mixer_s1810c.c @@ -554,11 +554,11 @@ int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) dev_info(&dev->dev, "Presonus Studio 1810c, device_setup: %u\n", chip->setup); if (chip->setup == 1) - dev_info(&dev->dev, "(8out/18in @ 48KHz)\n"); + dev_info(&dev->dev, "(8out/18in @ 48kHz)\n"); else if (chip->setup == 2) - dev_info(&dev->dev, "(6out/8in @ 192KHz)\n"); + dev_info(&dev->dev, "(6out/8in @ 192kHz)\n"); else - dev_info(&dev->dev, "(8out/14in @ 96KHz)\n"); + dev_info(&dev->dev, "(8out/14in @ 96kHz)\n"); ret = snd_s1810c_init_mixer_maps(chip); if (ret < 0) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 74c00c905d24..0ffff7640892 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -332,7 +332,7 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }, [SCARLETT2_PORT_TYPE_SPDIF] = { .id = 0x180, - /* S/PDIF outputs aren't available at 192KHz + /* S/PDIF outputs aren't available at 192kHz * but are included in the USB mux I/O * assignment message anyway */ @@ -401,7 +401,7 @@ static const struct scarlett2_device_info s18i20_gen2_info = { .dst_descr = "Analogue Output %02d Playback" }, [SCARLETT2_PORT_TYPE_SPDIF] = { - /* S/PDIF outputs aren't available at 192KHz + /* S/PDIF outputs aren't available at 192kHz * but are included in the USB mux I/O * assignment message anyway */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a777d36c4f5a..e42241087b77 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1694,8 +1694,8 @@ static void retire_playback_urb(struct snd_usb_substream *subs, int processed = urb->transfer_buffer_length / ep->stride; int est_delay; - /* ignore the delay accounting when procssed=0 is given, i.e. - * silent payloads are procssed before handling the actual data + /* ignore the delay accounting when processed=0 is given, i.e. + * silent payloads are processed before handling the actual data */ if (!processed) return; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index fca72730a802..c551141f337e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1144,14 +1144,14 @@ static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev) #define MAUDIO_SET 0x01 /* parse device_setup */ #define MAUDIO_SET_COMPATIBLE 0x80 /* use only "win-compatible" interfaces */ #define MAUDIO_SET_DTS 0x02 /* enable DTS Digital Output */ -#define MAUDIO_SET_96K 0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */ +#define MAUDIO_SET_96K 0x04 /* 48-96kHz rate if set, 8-48kHz otherwise */ #define MAUDIO_SET_24B 0x08 /* 24bits sample if set, 16bits otherwise */ #define MAUDIO_SET_DI 0x10 /* enable Digital Input */ #define MAUDIO_SET_MASK 0x1f /* bit mask for setup value */ -#define MAUDIO_SET_24B_48K_DI 0x19 /* 24bits+48KHz+Digital Input */ -#define MAUDIO_SET_24B_48K_NOTDI 0x09 /* 24bits+48KHz+No Digital Input */ -#define MAUDIO_SET_16B_48K_DI 0x11 /* 16bits+48KHz+Digital Input */ -#define MAUDIO_SET_16B_48K_NOTDI 0x01 /* 16bits+48KHz+No Digital Input */ +#define MAUDIO_SET_24B_48K_DI 0x19 /* 24bits+48kHz+Digital Input */ +#define MAUDIO_SET_24B_48K_NOTDI 0x09 /* 24bits+48kHz+No Digital Input */ +#define MAUDIO_SET_16B_48K_DI 0x11 /* 16bits+48kHz+Digital Input */ +#define MAUDIO_SET_16B_48K_NOTDI 0x01 /* 16bits+48kHz+No Digital Input */ static int quattro_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno) @@ -1597,7 +1597,7 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) /* * M-Audio Fast Track C400/C600 - when packets are not skipped, real - * world latency varies by approx. +/- 50 frames (at 96KHz) each time + * world latency varies by approx. +/- 50 frames (at 96kHz) each time * the stream is (re)started. When skipping packets 16 at endpoint * start up, the real world latency is stable within +/- 1 frame (also * across power cycles). @@ -1839,7 +1839,7 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, /* * MaxPacketsOnly attribute is erroneously set in endpoint * descriptors. As a result this card produces noise with - * all sample rates other than 96 KHz. + * all sample rates other than 96 kHz. */ fp->attributes &= ~UAC_EP_CS_ATTR_FILL_MAX; break; -- 2.26.2

2 1

[PATCH 1/2] ALSA: usb-audio: Fix packet size calculation
by Alexander Tsoy 30 Jun '20

30 Jun '20

Commit f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation") introduced a regression for devices which have playback endpoints with bInterval > 1. Fix this by taking ep->datainterval into account. Note that frame and fps are actually mean packet and packets per second in the code introduces by the mentioned commit. This will be fixed in a follow-up patch. Fixes: f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208353 Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/endpoint.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 9bea7d3f99f8..11f23778f0a5 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1093,6 +1093,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->freqn = get_usb_high_speed_rate(rate); ep->fps = 8000; } + ep->fps >>= ep->datainterval; ep->sample_rem = rate % ep->fps; ep->framesize[0] = rate / ep->fps; -- 2.26.2

2 3

[PATCH] ALSA: core: Warn on empty module
by Takashi Iwai 30 Jun '20

30 Jun '20

The module argument passed to snd_card_new() must be a valid non-NULL pointer when the module support is enabled. Since ASoC driver passes the argument from each snd_soc_card definition, one may forget to set the owner field and lead to a NULL module easily. For catching such an overlook, add a WARN_ON() in snd_card_new(). Also, put the card->module assignment in the ifdef block for a very minor optimization. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/init.c b/sound/core/init.c index b02a99766351..0478847ba2b8 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -203,7 +203,10 @@ int snd_card_new(struct device *parent, int idx, const char *xid, mutex_unlock(&snd_card_mutex); card->dev = parent; card->number = idx; +#ifdef MODULE + WARN_ON(!module); card->module = module; +#endif INIT_LIST_HEAD(&card->devices); init_rwsem(&card->controls_rwsem); rwlock_init(&card->ctl_files_rwlock); -- 2.16.4

5 9

[PATCH 0/4] soundwire: add SoundWire 1.2 spec support
by Bard Liao 30 Jun '20

30 Jun '20

This series adds basic support for SoundWire 1.2. Pierre-Louis Bossart (4): soundwire: add definitions for 1.2 spec soundwire: bus_type: convert open-coded while() to for() loop soundwire: extend SDW_SLAVE_ENTRY soundwire: bus: initialize bus clock base and scale registers drivers/soundwire/bus.c | 107 ++++++++++++++++++++++ drivers/soundwire/bus_type.c | 19 ++-- include/linux/mod_devicetable.h | 2 + include/linux/soundwire/sdw.h | 11 ++- include/linux/soundwire/sdw_registers.h | 117 +++++++++++++++++++++++- scripts/mod/devicetable-offsets.c | 2 + scripts/mod/file2alias.c | 6 +- 7 files changed, 250 insertions(+), 14 deletions(-) -- 2.17.1

2 5

Re: XHCI vs PCM2903B/PCM2904 part 2
by Mathias Nyman 30 Jun '20

30 Jun '20

On 30.6.2020 16.08, Rik van Riel wrote: > I misread the code, it's not a bitfield, so state 1 means an endpoint marked with running state. The next urb is never getting a response, though. > > However, the xhci spec says an endpoint is halted upon a babble error. I was looking at the same, so according to specs this state shouldn't be possible. > > The code right above the babble handling case adds halted into the endpoint state itself. Does the code handling the babble error need to do something similar to trigger cleanup elsewhere? It's a flag to prevent ringing the doorbell for a halted endpoint. Anyway, reset endpoint is meant to recover an endpoint in a halted state. Resetting non-halted endpoints will just lead to a context state error, and besides, isoc endpoints shouldn't halt. Anyways, I haven't got any better idea at the moment. You can try and see what a forced reset does with: diff --git a/drivers/usb/host/xhci-ring.c b/drivers/usb/host/xhci-ring.c index 2c255d0620b0..d79aca0df6d4 100644 --- a/drivers/usb/host/xhci-ring.c +++ b/drivers/usb/host/xhci-ring.c @@ -1900,8 +1900,7 @@ static int xhci_requires_manual_halt_cleanup(struct xhci_hcd *xhci, * endpoint anyway. Check if a babble halted the * endpoint. */ - if (GET_EP_CTX_STATE(ep_ctx) == EP_STATE_HALTED) - return 1; + return 1; return 0; } Traces also showed thet endpoint doorbell was rang after th babble error, so we know that didn't help restarting the endpoint. -Mathias

1 0

[PATCH 1/2] dt-bindings: sound: convert rk3328 codec binding to yaml
by Katsuhiro Suzuki 30 Jun '20

30 Jun '20

Just convert Rockchip rk3328 audio codec binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/rockchip,rk3328-codec.txt | 28 --------- .../bindings/sound/rockchip,rk3328-codec.yaml | 61 +++++++++++++++++++ 2 files changed, 61 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt deleted file mode 100644 index 1ecd75d2032a..000000000000 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ /dev/null @@ -1,28 +0,0 @@ -* Rockchip Rk3328 internal codec - -Required properties: - -- compatible: "rockchip,rk3328-codec" -- reg: physical base address of the controller and length of memory mapped - region. -- rockchip,grf: the phandle of the syscon node for GRF register. -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should be "pclk". -- spk-depop-time-ms: speak depop time msec. - -Optional properties: - -- mute-gpios: GPIO specifier for external line driver control (typically the - dedicated GPIO_MUTE pin) - -Example for rk3328 internal codec: - -codec: codec@ff410000 { - compatible = "rockchip,rk3328-codec"; - reg = <0x0 0xff410000 0x0 0x1000>; - rockchip,grf = <&grf>; - clocks = <&cru PCLK_ACODEC>; - clock-names = "pclk"; - mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; - spk-depop-time-ms = 100; -}; diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml new file mode 100644 index 000000000000..08a74e28e483 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip,rk3328-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk3328 internal codec + +maintainers: + - Heiko Stuebner <heiko(a)sntech.de> + +properties: + compatible: + const: rockchip,rk3328-codec + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for audio codec + + clock-names: + items: + - const: pclk + + rockchip,grf: + $ref: /schemas/types.yaml#/definitions/phandle + description: + The phandle of the syscon node for the GRF register. + + spk-depop-time-ms: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 200 + description: + Speaker depop time in msec. + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + +required: + - compatible + - reg + - clocks + - clock-names + - rockchip,grf + +examples: + - | + codec: codec@ff410000 { + compatible = "rockchip,rk3328-codec"; + reg = <0xff410000 0x1000>; + clocks = <&cru PCLK_ACODEC>; + clock-names = "pclk"; + rockchip,grf = <&grf>; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; + spk-depop-time-ms = 100; + }; -- 2.27.0

2 3

[PATCH] ASoC: fsl_asrc: Add an option to select internal ratio mode
by Shengjiu Wang 30 Jun '20

30 Jun '20

The ASRC not only supports ideal ratio mode, but also supports internal ratio mode. For internal rato mode, the rate of clock source should be divided with no remainder by sample rate, otherwise there is sound distortion. Add function fsl_asrc_select_clk() to find proper clock source for internal ratio mode, if the clock source is available then internal ratio mode will be selected. With change, the ideal ratio mode is not the only option for user. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 58 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 56 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 95f6a9617b0b..fcafc8ecb131 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -582,11 +582,59 @@ static int fsl_asrc_dai_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, &fsl_asrc_rate_constraints); } +/** + * Select proper clock source for internal ratio mode + */ +static int fsl_asrc_select_clk(struct fsl_asrc_priv *asrc_priv, + struct fsl_asrc_pair *pair, + int in_rate, + int out_rate) +{ + struct fsl_asrc_pair_priv *pair_priv = pair->private; + struct asrc_config *config = pair_priv->config; + int rate[2], select_clk[2]; /* Array size 2 means IN and OUT */ + int clk_rate, clk_index; + int i = 0, j = 0; + bool clk_sel[2]; + + rate[0] = in_rate; + rate[1] = out_rate; + + /* Select proper clock source for internal ratio mode */ + for (j = 0; j < 2; j++) { + for (i = 0; i < ASRC_CLK_MAP_LEN; i++) { + clk_index = asrc_priv->clk_map[j][i]; + clk_rate = clk_get_rate(asrc_priv->asrck_clk[clk_index]); + if (clk_rate != 0 && (clk_rate / rate[j]) <= 1024 && + (clk_rate % rate[j]) == 0) + break; + } + + if (i == ASRC_CLK_MAP_LEN) { + select_clk[j] = OUTCLK_ASRCK1_CLK; + clk_sel[j] = false; + } else { + select_clk[j] = i; + clk_sel[j] = true; + } + } + + /* Switch to ideal ratio mode if there is no proper clock source */ + if (!clk_sel[IN] || !clk_sel[OUT]) + select_clk[IN] = INCLK_NONE; + + config->inclk = select_clk[IN]; + config->outclk = select_clk[OUT]; + + return 0; +} + static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct fsl_asrc *asrc = snd_soc_dai_get_drvdata(dai); + struct fsl_asrc_priv *asrc_priv = asrc->private; struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; struct fsl_asrc_pair_priv *pair_priv = pair->private; @@ -605,8 +653,6 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.pair = pair->index; config.channel_num = channels; - config.inclk = INCLK_NONE; - config.outclk = OUTCLK_ASRCK1_CLK; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config.input_format = params_format(params); @@ -620,6 +666,14 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } + ret = fsl_asrc_select_clk(asrc_priv, pair, + config.input_sample_rate, + config.output_sample_rate); + if (ret) { + dev_err(dai->dev, "fail to select clock\n"); + return ret; + } + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); -- 2.21.0

4 4

ucm2: Fixes for moving DragonBoard UCM files.
by GitHub pull_request - opened 30 Jun '20

30 Jun '20

alsa-project/alsa-ucm-conf pull request #37 was opened from Srinivas-Kandagatla: Hi Jaroslav, Nico just noticed that you have moved and renamed all the DragonBoard ucm2 files. We have below few issues with that patches: 1> DB820c belongs to APQ8096 SoC, so it should be moved to ucm2/apq8096 folder rather than apq8016. 2> DB410c belongs to APQ8016 SoC, so it should be under ucm2/apq8016 rather than apq8016-sbc. 3> All the board specific ucm conf file names with actual board file are now replaced with SoC names, like DB845c is replace with sdm845. Any particular reason for doing this? Are you suggesting us to use some matching on card long names to include correct board files from top conf file? Can you provide us some pointers on how board specific config files should endup looking like? This PR fixes (1), and (2). Thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/37 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/37.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] asoc: Update supported rate and format for dummy dai
by Rohit kumar 29 Jun '20

29 Jun '20

Add support for 384KHz sample rate and S24_3LE bitwidth for dummy dai. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/soc-utils.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 922eac9..364b248 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -86,12 +86,13 @@ static const struct snd_soc_component_driver dummy_codec = { .non_legacy_dai_naming = 1, }; -#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_RATES SNDRV_PCM_RATE_8000_384000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_U8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_U16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ SNDRV_PCM_FMTBIT_U24_LE | \ SNDRV_PCM_FMTBIT_S32_LE | \ SNDRV_PCM_FMTBIT_U32_LE | \ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

[PATCH -next] ASoC: Documentation: fix reference to renamed source file
by Randy Dunlap 29 Jun '20

29 Jun '20

From: Randy Dunlap <rdunlap(a)infradead.org> sound/soc/soc-io.c was merged into sound/soc/soc-component.c, so fixup the Documentation to use the updated file name. Error: Cannot open file ../sound/soc/soc-io.c WARNING: kernel-doc '../scripts/kernel-doc -rst -enable-lineno ../sound/soc/soc-io.c' failed with return code 1 Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Mark Brown <broonie(a)kernel.org> --- Documentation/sound/kernel-api/alsa-driver-api.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20200626.orig/Documentation/sound/kernel-api/alsa-driver-api.rst +++ linux-next-20200626/Documentation/sound/kernel-api/alsa-driver-api.rst @@ -99,7 +99,7 @@ ASoC Core API .. kernel-doc:: include/sound/soc.h .. kernel-doc:: sound/soc/soc-core.c .. kernel-doc:: sound/soc/soc-devres.c -.. kernel-doc:: sound/soc/soc-io.c +.. kernel-doc:: sound/soc/soc-component.c .. kernel-doc:: sound/soc/soc-pcm.c .. kernel-doc:: sound/soc/soc-ops.c .. kernel-doc:: sound/soc/soc-compress.c

3 2

[PATCH] ASoC: qcom: Drop HAS_DMA dependency to fix link failure
by Geert Uytterhoeven 29 Jun '20

29 Jun '20

When building on allyesconfig kernel for a NO_DMA=y platform (e.g. Sun-3), CONFIG_SND_SOC_QCOM_COMMON=y, but CONFIG_SND_SOC_QDSP6_AFE=n, leading to a link failure: sound/soc/qcom/common.o: In function `qcom_snd_parse_of': common.c:(.text+0x2e2): undefined reference to `q6afe_is_rx_port' While SND_SOC_QDSP6 depends on HAS_DMA, SND_SOC_MSM8996 and SND_SOC_SDM845 don't, so the following warning is seen: WARNING: unmet direct dependencies detected for SND_SOC_QDSP6 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && HAS_DMA [=n] Selected by [y]: - SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] - SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && CROS_EC [=y] && I2C [=y] && SOUNDWIRE [=y] Until recently, this warning was harmless (from a compile-testing point-of-view), but the new user of q6afe_is_rx_port() turned this into a hard failure. As the QDSP6 driver itself builds fine if NO_DMA=y, and it depends on QCOM_APR (which in turns depends on ARCH_QCOM || COMPILE_TEST), it is safe to increase compile testing coverage. Hence fix the link failure by dropping the HAS_DMA dependency of SND_SOC_QDSP6. Fixes: a2120089251f1fe2 ("ASoC: qcom: common: set correct directions for dailinks") Fixes: 6b1687bf76ef84cb ("ASoC: qcom: add sdm845 sound card support") Fixes: a6f933f63f2ffdb2 ("ASoC: qcom: apq8096: Add db820c machine driver") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/soc/qcom/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index f51b28d1b94d87b5..92f51d0e9fe2bf02 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -72,7 +72,7 @@ config SND_SOC_QDSP6_ASM_DAI config SND_SOC_QDSP6 tristate "SoC ALSA audio driver for QDSP6" - depends on QCOM_APR && HAS_DMA + depends on QCOM_APR select SND_SOC_QDSP6_COMMON select SND_SOC_QDSP6_CORE select SND_SOC_QDSP6_AFE -- 2.17.1

2 1

[PATCH v2 0/3] ASoC: rt1015: fix compilation errors and warnings
by Tzung-Bi Shih 29 Jun '20

29 Jun '20

The series fixes compilation errors and warnings discovered in the thread https://patchwork.kernel.org/patch/11622319/. The original patch: (1) adds acpi.h for fixing implicit declaration of function 'ACPI_PTR' (2) sorts header inclusions in alphabetical The compilation errors and warnings are likely introduced by (2). The 1st and 2nd patches fix the newly discovered errors and warnings. The 3rd patch is the same as the original patch. Tzung-Bi Shih (3): ASoC: core: move definition of enum snd_soc_bias_level ASoC: dapm: declare missing structure prototypes ASoC: rt1015: add missing header inclusion include/sound/soc-dapm.h | 20 ++++++++++++++++++++ include/sound/soc.h | 18 ------------------ sound/soc/codecs/rt1015.c | 17 +++++++++-------- 3 files changed, 29 insertions(+), 26 deletions(-) -- 2.27.0.212.ge8ba1cc988-goog

2 6

[PATCH v4 0/3] ALSA: compress: Document stream states and fix gapless SM
by Vinod Koul 29 Jun '20

29 Jun '20

Srini found issue with gapless implementation which prompted to look deeper into SM for compressed stream. So documenting SM was first step, so first two patches add that. Last patch fixes the issue by keeping track on partial_drain and then moving state to 'running' in snd_compr_drain_notify() for partial_drain case on success. While at it, noticed snd_compr_drain_notify() is lockless state change, so fixed that as well. I have tested this on Dragon board RB3, compressed audio works out of the box on that platform and Srini will send driver and fcplay patches for gapless soon. Changes in v4: - Add review tag by Charles - fix the lock deadlock pointed by Charles Changes in v3: - Add pause->stop->free transition - Add setup->free transition - remove running->free as that goes thru stop Changes in v2: - Added tested tag by Srini - Update compress SM with Free state and compr_stop() transitions Vinod Koul (3): ALSA: compress: document the compress audio state machine ALSA: compress: document the compress gapless audio state machine ALSA: compress: fix partial_drain completion state .../sound/designs/compress-offload.rst | 83 +++++++++++++++++++ include/sound/compress_driver.h | 10 ++- sound/core/compress_offload.c | 4 + 3 files changed, 96 insertions(+), 1 deletion(-) -- 2.26.2

2 7

[PATCH v3 0/3] ALSA: compress: Document stream states and fix gapless SM
by Vinod Koul 29 Jun '20

29 Jun '20

Srini found issue with gapless implementation which prompted to look deeper into SM for compressed stream. So documenting SM was first step, so first two patches add that. Last patch fixes the issue by keeping track on partial_drain and then moving state to 'running' in snd_compr_drain_notify() for partial_drain case on success. While at it, noticed snd_compr_drain_notify() is lockless state change, so fixed that as well. I have tested this on Dragon board RB3, compressed audio works out of the box on that platform and Srini will send driver and fcplay patches for gapless soon. Changes in v3: - Add pause->stop->free transition - Add setup->free transition - remove running->free as that goes thru stop Changes in v2: - Added tested tag by Srini - Update compress SM with Free state and compr_stop() transitions Vinod Koul (3): ALSA: compress: document the compress audio state machine ALSA: compress: document the compress gapless audio state machine ALSA: compress: fix partial_drain completion state .../sound/designs/compress-offload.rst | 83 +++++++++++++++++++ include/sound/compress_driver.h | 12 ++- sound/core/compress_offload.c | 4 + 3 files changed, 98 insertions(+), 1 deletion(-) -- 2.26.2

2 6

Re: [PATCH v2 08/21] ASoC: codecs: merge .digital_mute() into .mute_stream()
by Kuninori Morimoto 29 Jun '20

29 Jun '20

Hi Peter, Mark Thank you for your review > > static const struct snd_soc_dai_ops tfa9879_dai_ops = { > > .hw_params = tfa9879_hw_params, > > - .digital_mute = tfa9879_digital_mute, > > + .mute_stream = tfa9879_mute, > > .set_fmt = tfa9879_set_fmt, > > + .no_capture_mute = 1, > > }; (snip) > - .digital_mute = tfa9879_digital_mute, > + .mute_stream = tfa9879_mute_stream, Yes, indeed. I can fix it by additional patch if v2 was accepted, or, in v3 if needed. Is it OK for you, and Mark ? Thank you for your help !! Best regards --- Kuninori Morimoto

1 0

[PATCH 10/21] ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()
by Kuninori Morimoto 29 Jun '20

29 Jun '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_dai_digital_mute() is internally using both mute_stream() (1) or digital_mute() (2), but the difference between these 2 are only handling direction. We can merge digital_mute() into mute_stream int snd_soc_dai_digital_mute(xxx, int direction) { ... else if (dai->driver->ops->mute_stream) (1) return dai->driver->ops->mute_stream(xxx, direction); else if (direction == SNDRV_PCM_STREAM_PLAYBACK && dai->driver->ops->digital_mute) (2) return dai->driver->ops->digital_mute(xxx); ... } Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This was missed on v2 patch-set. sound/soc/codecs/tas2552.c | 5 +++-- sound/soc/codecs/tas2562.c | 5 +++-- sound/soc/codecs/tas2770.c | 5 +++-- sound/soc/codecs/tas571x.c | 5 +++-- sound/soc/codecs/tas5720.c | 5 +++-- sound/soc/codecs/tas6424.c | 5 +++-- 6 files changed, 18 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 529c0fb93f9b..3a153526e47f 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -465,7 +465,7 @@ static int tas2552_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } -static int tas2552_mute(struct snd_soc_dai *dai, int mute) +static int tas2552_mute(struct snd_soc_dai *dai, int mute, int direction) { u8 cfg1_reg = 0; struct snd_soc_component *component = dai->component; @@ -519,7 +519,8 @@ static const struct snd_soc_dai_ops tas2552_speaker_dai_ops = { .set_sysclk = tas2552_set_dai_sysclk, .set_fmt = tas2552_set_dai_fmt, .set_tdm_slot = tas2552_set_dai_tdm_slot, - .digital_mute = tas2552_mute, + .mute_stream = tas2552_mute, + .no_capture_mute = 1, }; /* Formats supported by TAS2552 driver. */ diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 7fae88655a0f..8721b0ecc403 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -334,7 +334,7 @@ static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static int tas2562_mute(struct snd_soc_dai *dai, int mute) +static int tas2562_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; @@ -552,7 +552,8 @@ static const struct snd_soc_dai_ops tas2562_speaker_dai_ops = { .hw_params = tas2562_hw_params, .set_fmt = tas2562_set_dai_fmt, .set_tdm_slot = tas2562_set_dai_tdm_slot, - .digital_mute = tas2562_mute, + .mute_stream = tas2562_mute, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver tas2562_dai[] = { diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 54c8135fe43c..4538b2d0216f 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -189,7 +189,7 @@ static const struct snd_soc_dapm_route tas2770_audio_map[] = { {"VSENSE", "Switch", "VMON"}, }; -static int tas2770_mute(struct snd_soc_dai *dai, int mute) +static int tas2770_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; int ret; @@ -530,10 +530,11 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, } static struct snd_soc_dai_ops tas2770_dai_ops = { - .digital_mute = tas2770_mute, + .mute_stream = tas2770_mute, .hw_params = tas2770_hw_params, .set_fmt = tas2770_set_fmt, .set_tdm_slot = tas2770_set_dai_tdm_slot, + .no_capture_mute = 1, }; #define TAS2770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 5b7f9fcf6cbf..835a723ce5bc 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -301,7 +301,7 @@ static int tas571x_hw_params(struct snd_pcm_substream *substream, TAS571X_SDI_FMT_MASK, val); } -static int tas571x_mute(struct snd_soc_dai *dai, int mute) +static int tas571x_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; u8 sysctl2; @@ -354,7 +354,8 @@ static int tas571x_set_bias_level(struct snd_soc_component *component, static const struct snd_soc_dai_ops tas571x_dai_ops = { .set_fmt = tas571x_set_dai_fmt, .hw_params = tas571x_hw_params, - .digital_mute = tas571x_mute, + .mute_stream = tas571x_mute, + .no_capture_mute = 1, }; diff --git a/sound/soc/codecs/tas5720.c b/sound/soc/codecs/tas5720.c index e159f839d928..139ac5e683bf 100644 --- a/sound/soc/codecs/tas5720.c +++ b/sound/soc/codecs/tas5720.c @@ -199,7 +199,7 @@ static int tas5720_set_dai_tdm_slot(struct snd_soc_dai *dai, return ret; } -static int tas5720_mute(struct snd_soc_dai *dai, int mute) +static int tas5720_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; int ret; @@ -604,7 +604,8 @@ static const struct snd_soc_dai_ops tas5720_speaker_dai_ops = { .hw_params = tas5720_hw_params, .set_fmt = tas5720_set_dai_fmt, .set_tdm_slot = tas5720_set_dai_tdm_slot, - .digital_mute = tas5720_mute, + .mute_stream = tas5720_mute, + .no_capture_mute = 1, }; /* diff --git a/sound/soc/codecs/tas6424.c b/sound/soc/codecs/tas6424.c index aaba39295079..6198138e693a 100644 --- a/sound/soc/codecs/tas6424.c +++ b/sound/soc/codecs/tas6424.c @@ -252,7 +252,7 @@ static int tas6424_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } -static int tas6424_mute(struct snd_soc_dai *dai, int mute) +static int tas6424_mute(struct snd_soc_dai *dai, int mute, int direction) { struct snd_soc_component *component = dai->component; struct tas6424_data *tas6424 = snd_soc_component_get_drvdata(component); @@ -382,7 +382,8 @@ static const struct snd_soc_dai_ops tas6424_speaker_dai_ops = { .hw_params = tas6424_hw_params, .set_fmt = tas6424_set_dai_fmt, .set_tdm_slot = tas6424_set_dai_tdm_slot, - .digital_mute = tas6424_mute, + .mute_stream = tas6424_mute, + .no_capture_mute = 1, }; static struct snd_soc_dai_driver tas6424_dai[] = { -- 2.25.1

1 0

[PATCH v2 0/7] ASoC: QCOM: Add support for SC7180 lpass variant
by Ajit Pandey 29 Jun '20

29 Jun '20

We've common ASoC cpu driver for QCOM LPASS soc varaints. This patch chain add support for new variant SC7180 soc by doing the required modification in existing common lpass-cpu driver. Below is a brief summary of patch series: PATCH v2 0001 ... 0004: Update lpass-cpu driver and documentation to make it more generic and support newer soc registers configuration. PATCH v2 0005 ... 0007: Add documentation and platform driver for newer SC7180 SOC variant. Ajit Pandey (7): Documentation: device-tree: sound: Update lpass-cpu driver binding ASoC: qcom: Add common array to initialize soc based core clocks ASoC: qcom: lpass-cpu: Make "ahbix-clk" an optional clock. ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers include: dt-bindings: sound: Add sc7180-lpass bindings header device-tree: bindings: sound: lpass-cpu: Add new compatible soc ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.txt | 79 ------- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 154 +++++++++++++ include/dt-bindings/sound/sc7180-lpass.h | 10 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 100 ++++++-- sound/soc/qcom/lpass-cpu.c | 116 ++++++---- sound/soc/qcom/lpass-lpaif-reg.h | 203 ++++++++++------- sound/soc/qcom/lpass-platform.c | 86 ++++--- sound/soc/qcom/lpass-sc7180.c | 252 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 40 +++- 11 files changed, 790 insertions(+), 257 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml create mode 100644 include/dt-bindings/sound/sc7180-lpass.h create mode 100644 sound/soc/qcom/lpass-sc7180.c -- ('The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum,\na Linux Foundation Collaborative Project')

4 18

[PATCH alsa-lib 1/5] Fix invalid "Regex" Type in various Condition blocks
by Hans de Goede 28 Jun '20

28 Jun '20

Regex is not a valid Condition type, using it leads to errors like these: ALSA lib ucm_cond.c:300:(if_eval) unknown If.Condition.Type ALSA lib main.c:983:(snd_use_case_mgr_open) error: failed to import cht-bsw-rt5672 use case configuration -22 alsaucm: error failed to open sound card cht-bsw-rt5672: Invalid argument Replacing it with "RegexMatch" fixes this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/cht-bsw-rt5672/HiFi.conf | 4 ++-- ucm2/chtnau8824/HiFi.conf | 2 +- ucm2/chtrt5645/HiFi.conf | 6 +++--- ucm2/sof-soundwire/sof-soundwire.conf | 4 ++-- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/ucm2/cht-bsw-rt5672/HiFi.conf b/ucm2/cht-bsw-rt5672/HiFi.conf index 394e45d..dc8ddd5 100644 --- a/ucm2/cht-bsw-rt5672/HiFi.conf +++ b/ucm2/cht-bsw-rt5672/HiFi.conf @@ -9,7 +9,7 @@ Define.HeadsetMic "yes" If.cfg-dmic1 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(SoMeThInK1)" # fixme! } @@ -21,7 +21,7 @@ If.cfg-dmic1 { If.cfg-dmic2 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(LENOVO.*ThinkPad8)" } diff --git a/ucm2/chtnau8824/HiFi.conf b/ucm2/chtnau8824/HiFi.conf index 6b7624d..3ccd6b1 100644 --- a/ucm2/chtnau8824/HiFi.conf +++ b/ucm2/chtnau8824/HiFi.conf @@ -2,7 +2,7 @@ Define.Speaker "Speaker" If.cfg-mspk { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(cube-i1_TF|PIPO-W2S)" } diff --git a/ucm2/chtrt5645/HiFi.conf b/ucm2/chtrt5645/HiFi.conf index fb8636d..5d0e3bf 100644 --- a/ucm2/chtrt5645/HiFi.conf +++ b/ucm2/chtrt5645/HiFi.conf @@ -4,7 +4,7 @@ Define.DigitalMic "" If.cfg-dmic1 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(ASUSTeK.*T100HAN|ASUSTeK.*T101HA)" } @@ -16,7 +16,7 @@ If.cfg-dmic1 { If.cfg-dmic2 { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(LENOVO.*LenovoMIIX320|MEDION.*Wingman)" } @@ -28,7 +28,7 @@ If.cfg-dmic2 { If.cfg-mspk { Condition { - Type Regex + Type RegexMatch String "${CardLongName}" Regex "(gpd-win-pocket-rt5645|TECLAST-X80Pro)" } diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index fe43d5f..2ba44be 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -38,7 +38,7 @@ DefineRegex { If.hs_init { Condition { - Type Regex + Type RegexMatch Regex "(rt5682|rt700|rt711)" String "${var:HeadsetCodec1}" } @@ -47,7 +47,7 @@ If.hs_init { If.mic_init { Condition { - Type Regex + Type RegexMatch Regex "(rt715)" String "${var:MicCodec1}" } -- 2.26.2

2 5

[PATCH] ucm: Allow empty strings in "${var:...}" substitutions
by Hans de Goede 28 Jun '20

28 Jun '20

Recent ucm-conf changes introduce checks like this one in various places: If.mspk { Condition { Type String Empty "${var:MonoSpeaker}" } True ... False ... } The 'Empty "${var:MonoSpeaker}"' part can only every succeed if we do: Define.MonoSpeaker "" But so far that would result in an error like this one: ALSA lib ucm_subs.c:367:(uc_mgr_get_substituted_value) variable '${var:MonoSpeaker}' is not defined in this context! ALSA lib main.c:983:(snd_use_case_mgr_open) error: failed to import cht-bsw-rt5672 use case configuration -22 alsaucm: error failed to open sound card cht-bsw-rt5672: Invalid argument This commit fixes this by allowing empty values for "${var:...}" substitutions. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- Note besides the mentioned error, this also fixes similar errors I have been seeing on every board since alsa-ucm-conf commit d001c8de287f ("ucm.conf: add support for the kernel module name tree") --- src/ucm/ucm_subs.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/src/ucm/ucm_subs.c b/src/ucm/ucm_subs.c index 293426f2..a154aa51 100644 --- a/src/ucm/ucm_subs.c +++ b/src/ucm/ucm_subs.c @@ -262,9 +262,10 @@ static char *rval_var(snd_use_case_mgr_t *uc_mgr, const char *id) goto __rval; \ } -#define MATCH_VARIABLE2(name, id, fcn) \ +#define MATCH_VARIABLE2(name, id, fcn, empty_ok) \ if (strncmp((name), (id), sizeof(id) - 1) == 0) { \ idsize = sizeof(id) - 1; \ + allow_empty = (empty_ok); \ fcn2 = (fcn); \ goto __match2; \ } @@ -314,11 +315,11 @@ __std: MATCH_VARIABLE(value, "${CardName}", rval_card_name, false); MATCH_VARIABLE(value, "${CardLongName}", rval_card_longname, false); MATCH_VARIABLE(value, "${CardComponents}", rval_card_components, true); - MATCH_VARIABLE2(value, "${env:", rval_env); - MATCH_VARIABLE2(value, "${sys:", rval_sysfs); - MATCH_VARIABLE2(value, "${var:", rval_var); - MATCH_VARIABLE2(value, "${CardNumberByName:", rval_card_number_by_name); - MATCH_VARIABLE2(value, "${CardIdByName:", rval_card_id_by_name); + MATCH_VARIABLE2(value, "${env:", rval_env, false); + MATCH_VARIABLE2(value, "${sys:", rval_sysfs, false); + MATCH_VARIABLE2(value, "${var:", rval_var, true); + MATCH_VARIABLE2(value, "${CardNumberByName:", rval_card_number_by_name, false); + MATCH_VARIABLE2(value, "${CardIdByName:", rval_card_id_by_name, false); __merr: err = -EINVAL; tmp = strchr(value, '}'); -- 2.26.2

3 2

[PATCH 1/6] ASoC: Intel: cht_bsw_rt5672: Change bus format to I2S 2 channel
by Hans de Goede 28 Jun '20

28 Jun '20

The default mode for SSP configuration is TDM 4 slot and so far we were using this for the bus format on cht-bsw-rt56732 boards. One board, the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The second piggy-backed, output-only codec is inside the keyboard-dock (which has extra speakers). Unlike the main rt5672 codec, we cannot configure this codec, it is hard coded to use 2 channel 24 bit I2S. Using 4 channel TDM leads to the dock speakers codec (which listens in on the data send from the SSP to the rt5672 codec) emiting horribly distorted sound. Since we only support 2 channels anyways, there is no need for TDM on any cht-bsw-rt5672 designs. So we can simply use I2S 2ch everywhere. This commit fixes the Lenovo Miix 2 10 dock speakers issue by changing the bus format set in cht_codec_fixup() to I2S 2 channel. This change has been tested on the following devices with a rt5672 codec: Lenovo Miix 2 10 Lenovo Thinkpad 8 Lenovo Thinkpad 10 (gen 1) Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_rt5672.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 7a43c70a1378..22e432768edb 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -253,21 +253,20 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); /* - * Default mode for SSP configuration is TDM 4 slot + * Default mode for SSP configuration is TDM 4 slot. One board/design, + * the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2. The + * second piggy-backed, output-only codec is inside the keyboard-dock + * (which has extra speakers). Unlike the main rt5672 codec, we cannot + * configure this codec, it is hard coded to use 2 channel 24 bit I2S. + * Since we only support 2 channels anyways, there is no need for TDM + * on any cht-bsw-rt5672 designs. So we simply use I2S 2ch everywhere. */ - ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), - SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_IB_NF | + ret = snd_soc_dai_set_fmt(asoc_rtd_to_cpu(rtd, 0), + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) { - dev_err(rtd->dev, "can't set format to TDM %d\n", ret); - return ret; - } - - /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), 0xF, 0xF, 4, 24); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); return ret; } -- 2.26.2

1 2

problem with Jabra EVOLVE 20 headset, and usb quirk update
by Hin-Tak Leung 27 Jun '20

27 Jun '20

Hi, This is a headset i.e. with both a ear set, and a microphone - the playback seems fine, but recording only accept one setting - and also I was having problem with zoom using it ("cannot get freq" see the tail end of this e-mail). switching via the gnome setting -> sound (from built-in hda) seems sluggish, and occasionally the volume is frozen to mute and unresponsive to unmute. $ arecord -f S16_LE -D front:CARD=J20,DEV=0 -d 10 wonder.wav Recording WAVE 'wonder.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono Warning: rate is not accurate (requested = 8000Hz, got = 16000Hz) please, try the plug plugin and, also zoom seems to have problem getting at the microphone. I read that it is something to do with rate querying? Attached is "lsusb -vvvv -d 0b0e:0301 > usb-audio-lsusb-vvvv-0b0e:0301". I have tried just letting udev doing its thing, or preloading the module with with modprobe snd_usb_audio quirk_alias=0b0e0301:0b0e0349 or modprobe snd_usb_audio quirk_alias=0b0e0301:0b0e030b as below. But the "cannot get freq" message stays. This is based on similar vendor ids - sound/usb/format.c: if (chip->usb_id == USB_ID(0x0b0e, 0x030b)) { sound/usb/quirks.c: chip->usb_id == USB_ID(0x0b0e, 0x0349) || The device does not seems to be under quirks.c:snd_usb_ctl_msg_quirk() ( I don't really get delays, I think), format.c:formatparse_audio_format_rates_v1() seems to be somewhat related, but I don't have issue with playback, just capture. Should I insert this vid/pid into bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) as suggested elsewhere? I am just uncomfortable doing so as there isn't a quirk for this vendor there, yet. Also, related question: the format rate quirk routine and usb_ctl_msg_quirk routine does not seem to be affected by the alias_quirk list, as far as I understand and read the code? Here is the dmesg, under plain udev autoprobe, and the two preloads. I hope this is enough info. Please feel free to ask for more. (I also maintain https://github.com/HinTak/sound-usb-dkms/ as part of my previous job, but this device/usage is slightly complicated as I need to sign modified modules due to UEFI ; so I'd rather try something that has a chance of working, then just blindly try any modification) [25475.773095] usbcore: registered new interface driver snd-usb-audio [25482.559878] usb 2-2: new full-speed USB device number 6 using xhci_hcd [25482.764666] usb 2-2: New USB device found, idVendor=0b0e, idProduct=0301, bcdDevice= 3.00 [25482.764669] usb 2-2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [25482.764671] usb 2-2: Product: Jabra EVOLVE 20 [25482.764672] usb 2-2: Manufacturer: GN Netcom A/S [25482.764674] usb 2-2: SerialNumber: 0001B7518B0E07 [25482.770770] usb 2-2: device (0b0e:0301): applying quirk alias 0b0e:0349 [25482.794845] usb 2-2: 1:1: cannot get freq at ep 0x83 [25482.808838] usb 2-2: 2:1: cannot get freq at ep 0x4 [25483.133618] input: GN Netcom A/S Jabra EVOLVE 20 as /devices/pci0000:00/0000:00:10.0/usb2/2-2/2-2:1.3/0003:0B0E:0301.0004/input/input16 [25483.187155] jabra 0003:0B0E:0301.0004: input,hiddev96,hidraw0: USB HID v1.00 Device [GN Netcom A/S Jabra EVOLVE 20] on usb-0000:00:10.0-2/input3 [25483.444718] usb 2-2: 2:1: cannot get freq at ep 0x4 [25865.406518] usb 2-2: USB disconnect, device number 6 [25883.242957] usbcore: deregistering interface driver snd-usb-audio [25895.379533] usb 2-2: new full-speed USB device number 7 using xhci_hcd [25895.584862] usb 2-2: New USB device found, idVendor=0b0e, idProduct=0301, bcdDevice= 3.00 [25895.584868] usb 2-2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [25895.584871] usb 2-2: Product: Jabra EVOLVE 20 [25895.584874] usb 2-2: Manufacturer: GN Netcom A/S [25895.584876] usb 2-2: SerialNumber: 0001B7518B0E07 [25895.640239] input: GN Netcom A/S Jabra EVOLVE 20 as /devices/pci0000:00/0000:00:10.0/usb2/2-2/2-2:1.3/0003:0B0E:0301.0005/input/input17 [25895.692123] jabra 0003:0B0E:0301.0005: input,hiddev96,hidraw0: USB HID v1.00 Device [GN Netcom A/S Jabra EVOLVE 20] on usb-0000:00:10.0-2/input3 [25895.875421] usb 2-2: 1:1: cannot get freq at ep 0x83 [25895.885726] usb 2-2: 2:1: cannot get freq at ep 0x4 [25896.060320] usbcore: registered new interface driver snd-usb-audio [25896.149752] usb 2-2: 2:1: cannot get freq at ep 0x4 [25899.451644] usb 2-2: USB disconnect, device number 7 [25919.315839] usbcore: registered new interface driver snd-usb-audio [25924.595119] usb 2-2: new full-speed USB device number 8 using xhci_hcd [25924.797720] usb 2-2: New USB device found, idVendor=0b0e, idProduct=0301, bcdDevice= 3.00 [25924.797722] usb 2-2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [25924.797724] usb 2-2: Product: Jabra EVOLVE 20 [25924.797725] usb 2-2: Manufacturer: GN Netcom A/S [25924.797726] usb 2-2: SerialNumber: 0001B7518B0E07 [25924.802782] usb 2-2: device (0b0e:0301): applying quirk alias 0b0e:030b [25924.822715] usb 2-2: 1:1: cannot get freq at ep 0x83 [25924.832718] usb 2-2: 2:1: cannot get freq at ep 0x4 [25925.052655] input: GN Netcom A/S Jabra EVOLVE 20 as /devices/pci0000:00/0000:00:10.0/usb2/2-2/2-2:1.3/0003:0B0E:0301.0006/input/input18 [25925.104757] jabra 0003:0B0E:0301.0006: input,hiddev96,hidraw0: USB HID v1.00 Device [GN Netcom A/S Jabra EVOLVE 20] on usb-0000:00:10.0-2/input3 [25925.338094] usb 2-2: 2:1: cannot get freq at ep 0x4 [26253.731814] usb 2-2: USB disconnect, device number 8

1 0

[PATCH 16/17] ASoC: Intel: sdw_max98373: add card_late_probe support
by Pierre-Louis Bossart 26 Jun '20

26 Jun '20

From: randerwang <rander.wang(a)linux.intel.com> Disable Left and Right Spk pin after boot so that sof can get suspended. Please check patch: ASoC: Intel: Boards: tgl_max98373: add dai_trigger function Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: randerwang <rander.wang(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_sdw.c | 19 ++++++++++++++++++- sound/soc/intel/boards/sof_sdw_common.h | 6 ++++++ sound/soc/intel/boards/sof_sdw_max98373.c | 12 ++++++++++++ 3 files changed, 36 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 45be9ec6d4ef..be8eccb50450 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -237,6 +237,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { .direction = {true, true}, .dai_name = "max98373-aif1", .init = sof_sdw_mx8373_init, + .codec_card_late_probe = sof_sdw_mx8373_late_probe, }, { .id = 0x5682, @@ -927,13 +928,29 @@ static int sof_card_dai_links_create(struct device *dev, return 0; } +static int sof_sdw_card_late_probe(struct snd_soc_card *card) +{ + int i, ret; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + if (!codec_info_list[i].late_probe) + continue; + + ret = codec_info_list[i].codec_card_late_probe(card); + if (ret < 0) + return ret; + } + + return sof_sdw_hdmi_card_late_probe(card); +} + /* SoC card */ static const char sdw_card_long_name[] = "Intel Soundwire SOF"; static struct snd_soc_card card_sof_sdw = { .name = "soundwire", .owner = THIS_MODULE, - .late_probe = sof_sdw_hdmi_card_late_probe, + .late_probe = sof_sdw_card_late_probe, .codec_conf = codec_conf, .num_configs = ARRAY_SIZE(codec_conf), }; diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 3f820cf99a89..426017626b16 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -11,6 +11,7 @@ #include <linux/bits.h> #include <linux/types.h> +#include <sound/soc.h> #define MAX_NO_PROPS 2 #define MAX_HDMI_NUM 4 @@ -61,6 +62,9 @@ struct sof_sdw_codec_info { struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback); + + bool late_probe; + int (*codec_card_late_probe)(struct snd_soc_card *card); }; struct mc_private { @@ -114,6 +118,8 @@ int sof_sdw_mx8373_init(const struct snd_soc_acpi_link_adr *link, struct sof_sdw_codec_info *info, bool playback); +int sof_sdw_mx8373_late_probe(struct snd_soc_card *card); + /* RT5682 support */ int sof_sdw_rt5682_init(const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, diff --git a/sound/soc/intel/boards/sof_sdw_max98373.c b/sound/soc/intel/boards/sof_sdw_max98373.c index a38ddc099a95..6437872a9b3d 100644 --- a/sound/soc/intel/boards/sof_sdw_max98373.c +++ b/sound/soc/intel/boards/sof_sdw_max98373.c @@ -68,7 +68,19 @@ int sof_sdw_mx8373_init(const struct snd_soc_acpi_link_adr *link, if (info->amp_num == 2) dai_links->init = spk_init; + info->late_probe = true; + dai_links->ops = &max_98373_sdw_ops; return 0; } + +int sof_sdw_mx8373_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dapm_context *dapm = &card->dapm; + + /* Disable Left and Right Spk pin after boot */ + snd_soc_dapm_disable_pin(dapm, "Left Spk"); + snd_soc_dapm_disable_pin(dapm, "Right Spk"); + return snd_soc_dapm_sync(dapm); +} -- 2.20.1

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Fwd: [PATCH] snd/soc: correct trivial kernel-doc inconsistency
by Colton Lewis 26 Jun '20

26 Jun '20

Can some please accept or comment on this patch? ---------- Forwarded Message ---------- Subject: [PATCH] snd/soc: correct trivial kernel-doc inconsistency Date: Saturday, June 13, 2020, 9:29:08 PM CDT From: Colton Lewis <colton.w.lewis(a)protonmail.com> To: lgirdwood(a)gmail.com CC: trivial(a)kernel.org, Colton Lewis <colton.w.lewis(a)protonmail.com> Silence documentation build warning by correcting kernel-doc comment for snd_soc_runtime_action. ./sound/soc/soc-pcm.c:220: warning: Function parameter or member 'action' not described in 'snd_soc_runtime_action' Signed-off-by: Colton Lewis <colton.w.lewis(a)protonmail.com> --- sound/soc/soc-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c114b4542ce..9ab376844ac1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -208,6 +208,7 @@ static inline void dpcm_remove_debugfs_state(struct snd_soc_dpcm *dpcm) * PCM runtime components * @rtd: ASoC PCM runtime that is activated * @stream: Direction of the PCM stream + * @action: Activate stream if 1. Deactivate if -1. * * Increments/Decrements the active count for all the DAIs and components * attached to a PCM runtime. -- 2.26.2 -----------------------------------------

3 4

Cannot use HyperX Cloud Flight S Microphone
by GitHub issues - opened 25 Jun '20

25 Jun '20

alsa-project/alsa-lib issue #65 was opened from DirtyCajunRice: I have a New USB headset that was added/fixed in kernel patch 5.7rc3 (https://lkml.org/lkml/2020/4/4/197). It now loads all the output options but no input options. Is there something i can do to force it? also have this if it helps: http://alsa-project.org/db/?f=7cf962fe3df280a9a81f5cd1e82633d8cba7b02e Issue URL : https://github.com/alsa-project/alsa-lib/issues/65 Repository URL: https://github.com/alsa-project/alsa-lib

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