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

[PATCH 1/1] MAINTAINERS: change SoundWire maintainer
by Sanyog Kale 15 Jul '20

15 Jul '20

Add Bard as SoundWire maintainer from Intel and change Sanyog's role as reviewer. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 1c1527a34d99..9559e947bf67 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -16012,8 +16012,9 @@ F: sound/soc/sof/ SOUNDWIRE SUBSYSTEM M: Vinod Koul <vkoul(a)kernel.org> -M: Sanyog Kale <sanyog.r.kale(a)intel.com> +M: Bard Liao <yung-chuan.liao(a)linux.intel.com> R: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> +R: Sanyog Kale <sanyog.r.kale(a)intel.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: Documentation/driver-api/soundwire/ -- 2.17.1

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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

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[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

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[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

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[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

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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

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[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

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