October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: madera: Add support for using MCLK3
by Charles Keepax 01 Oct '19

01 Oct '19

Some Madera CODECs have a third MCLK pin allow this to be sourced for the FLLs and system clocks. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/madera.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/madera.h b/sound/soc/codecs/madera.h index 1f3e8e230cf25..6d8938a3fb647 100644 --- a/sound/soc/codecs/madera.h +++ b/sound/soc/codecs/madera.h @@ -27,6 +27,7 @@ #define MADERA_FLL_SRC_NONE -1 #define MADERA_FLL_SRC_MCLK1 0 #define MADERA_FLL_SRC_MCLK2 1 +#define MADERA_FLL_SRC_MCLK3 2 #define MADERA_FLL_SRC_SLIMCLK 3 #define MADERA_FLL_SRC_FLL1 4 #define MADERA_FLL_SRC_FLL2 5 @@ -51,6 +52,7 @@ #define MADERA_CLK_SRC_MCLK1 0x0 #define MADERA_CLK_SRC_MCLK2 0x1 +#define MADERA_CLK_SRC_MCLK3 0x2 #define MADERA_CLK_SRC_FLL1 0x4 #define MADERA_CLK_SRC_FLL2 0x5 #define MADERA_CLK_SRC_FLL3 0x6 -- 2.11.0

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[alsa-devel] [PATCH] ASoC: wm_adsp: Fix theoretical NULL pointer for alg_region
by Charles Keepax 01 Oct '19

01 Oct '19

From: Li Xu <li.xu(a)cirrus.com> Fix potential NULL pointer dereference for alg_region in wm_adsp_buffer_parse_legacy. In practice this can never happen as loading the firmware should have failed at the wm_adsp2_setup_algs stage, however probably better for the code to be robust against future changes and this is more helpful for static analysis. Signed-off-by: Li Xu <li.xu(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ae28d9907c30f..85396d920e0a9 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3697,11 +3697,16 @@ static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp) u32 xmalg, addr, magic; int i, ret; + alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); + if (!alg_region) { + adsp_err(dsp, "No algorithm region found\n"); + return -EINVAL; + } + buf = wm_adsp_buffer_alloc(dsp); if (!buf) return -ENOMEM; - alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); xmalg = dsp->ops->sys_config_size / sizeof(__be32); addr = alg_region->base + xmalg + ALG_XM_FIELD(magic); -- 2.11.0

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[alsa-devel] [PATCH 00/13] ASoC: SOF: fixes and improvements
by Pierre-Louis Bossart 01 Oct '19

01 Oct '19

No new features in this patchset but fixes and improvements identified during validation/product hardening/SoundWire development. I realize we are at the end of the merge window so will let Takashi/Mark decide if this series can be provided as fixes for v5.4 or queued for later. Kai Vehmanen (2): ASoC: SOF: pcm: fix resource leak in hw_free ASoC: SOF: Intel: hda: fix warnings during FW load Keyon Jie (1): ASoC: SOF: topology: fix parse fail issue for byte/bool tuple types Liam Girdwood (3): ASoC: SOF: ipc: retain DSP context after FW exception. ASoC: SOF: Intel: initialise and verify FW crash dump data. ASoC: SOF: Intel: Add context data to any IPC timeout. Marcin Rajwa (2): ASoC: SOF: ipc: introduce no_stream_position in sof_ipc_stream_params struct ASoC: SOF: Intel: hda: fix reset of host_period_bytes Pan Xiuli (1): ASoC: SOF: pcm: harden PCM STOP sequence Pierre-Louis Bossart (2): ASoC: SOF: loader: fix kernel oops on firmware boot failure ASoC: SOF: trace: move to opt-in with Kconfig and module parameter Ranjani Sridharan (2): ASoC: SOF: core: check for mandatory fw_ready op during SOF probe ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture include/sound/sof/stream.h | 4 +-- sound/soc/sof/Kconfig | 16 ++++++++++++ sound/soc/sof/core.c | 28 +++++++++++++++----- sound/soc/sof/debug.c | 16 ++++++++++++ sound/soc/sof/intel/Kconfig | 10 +++++++ sound/soc/sof/intel/bdw.c | 29 +++++++++++++++++++- sound/soc/sof/intel/byt.c | 29 +++++++++++++++++++- sound/soc/sof/intel/hda-ctrl.c | 12 +++------ sound/soc/sof/intel/hda-loader.c | 1 + sound/soc/sof/intel/hda-pcm.c | 15 ++++++++--- sound/soc/sof/intel/hda-stream.c | 45 +++++++++++++++++++++++++++----- sound/soc/sof/intel/hda.c | 7 +++++ sound/soc/sof/intel/hda.h | 5 +++- sound/soc/sof/ipc.c | 10 +------ sound/soc/sof/loader.c | 4 +-- sound/soc/sof/pcm.c | 35 +++++++++++++++++-------- sound/soc/sof/sof-priv.h | 17 ++++++++---- sound/soc/sof/topology.c | 4 ++- sound/soc/sof/trace.c | 17 +++++++++++- 19 files changed, 245 insertions(+), 59 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH 0/4] ASoC: Intel: boards: improvements for RT5651/82-based platforms
by Pierre-Louis Bossart 01 Oct '19

01 Oct '19

RT5651/RT5682 boards are used for SOF validation. This patchset corrects problems identified during module load/reload tests and an error for DMIC support in some configurations. These fixes are not critical and can be queued for v5.5 Bard Liao (1): ASoC: intel: sof_rt5682: use separate route map for dmic Jaska Uimonen (3): ASoC: intel: bytcr_rt5651: add null check to support_button_press ASoC: rt5682: add NULL handler to set_jack function ASoC: intel: sof_rt5682: add remove function to disable jack sound/soc/codecs/rt5651.c | 3 ++ sound/soc/codecs/rt5682.c | 12 +++++-- sound/soc/intel/boards/sof_rt5682.c | 51 ++++++++++++++++++++++++++--- 3 files changed, 60 insertions(+), 6 deletions(-) -- 2.20.1

2 8

[alsa-devel] [PATCH] ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency
by Arnd Bergmann 01 Oct '19

01 Oct '19

CONFIG_SND_SOC_SOF_IMX depends on CONFIG_SND_SOC_SOF, but is in turn referenced by the sof-of-dev driver. This creates a reverse dependency that manifests in a link error when CONFIG_SND_SOC_SOF_OF is built-in but CONFIG_SND_SOC_SOF_IMX=m: sound/soc/sof/sof-of-dev.o:(.data+0x118): undefined reference to `sof_imx8_ops' Make the latter a 'bool' symbol and change the Makefile so the imx8 driver is compiled the same way as the driver using it. A nicer way would be to reverse the layering and move all the imx specific bits of sof-of-dev.c into the imx driver itself, which can then call into the common code. Doing this would need more testing and can be done if we add another driver like the first one. Fixes: f4df4e4042b0 ("ASoC: SOF: imx8: Fix COMPILE_TEST error") Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sof/imx/Kconfig | 2 +- sound/soc/sof/imx/Makefile | 4 +++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5acae75f5750..a3891654a1fc 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -12,7 +12,7 @@ config SND_SOC_SOF_IMX_TOPLEVEL if SND_SOC_SOF_IMX_TOPLEVEL config SND_SOC_SOF_IMX8 - tristate "SOF support for i.MX8" + bool "SOF support for i.MX8" depends on IMX_SCU depends on IMX_DSP help diff --git a/sound/soc/sof/imx/Makefile b/sound/soc/sof/imx/Makefile index 6ef908e8c807..9e8f35df0ff2 100644 --- a/sound/soc/sof/imx/Makefile +++ b/sound/soc/sof/imx/Makefile @@ -1,4 +1,6 @@ # SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) snd-sof-imx8-objs := imx8.o -obj-$(CONFIG_SND_SOC_SOF_IMX8) += snd-sof-imx8.o +ifdef CONFIG_SND_SOC_SOF_IMX8 +obj-$(CONFIG_SND_SOC_SOF_OF) += snd-sof-imx8.o +endif -- 2.20.0

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[alsa-devel] [PATCH] ASoC: max98373: check for device node before parsing
by Sathyanarayana Nujella 01 Oct '19

01 Oct '19

Below Oops is caused in a system which uses ACPI instead of device node: of_get_named_gpiod_flags: can't parse 'maxim,reset-gpio' property of node '(null)[0]' BUG: kernel NULL pointer dereference, address: 0000000000000010 This patch avoids NULL pointer deferencing by adding a check before parsing and initializes to make reset-gpio pin as invalid. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> --- sound/soc/codecs/max98373.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index e609abcf3220..eb709d528259 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -901,16 +901,20 @@ static void max98373_slot_config(struct i2c_client *i2c, max98373->i_slot = value & 0xF; else max98373->i_slot = 1; - - max98373->reset_gpio = of_get_named_gpio(dev->of_node, + if (dev->of_node) { + max98373->reset_gpio = of_get_named_gpio(dev->of_node, "maxim,reset-gpio", 0); - if (!gpio_is_valid(max98373->reset_gpio)) { - dev_err(dev, "Looking up %s property in node %s failed %d\n", - "maxim,reset-gpio", dev->of_node->full_name, - max98373->reset_gpio); + if (!gpio_is_valid(max98373->reset_gpio)) { + dev_err(dev, "Looking up %s property in node %s failed %d\n", + "maxim,reset-gpio", dev->of_node->full_name, + max98373->reset_gpio); + } else { + dev_dbg(dev, "maxim,reset-gpio=%d", + max98373->reset_gpio); + } } else { - dev_dbg(dev, "maxim,reset-gpio=%d", - max98373->reset_gpio); + /* this makes reset_gpio as invalid */ + max98373->reset_gpio = -1; } if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value)) -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: Intel: cht_bsw_rt5645: Add quirk for boards using pmc_plt_clk_0
by Sam McNally 01 Oct '19

01 Oct '19

As of commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL"), the cht_bsw_rt5645 driver needs to enable the clock it's using for the codec's mclk. It does this from commit 7735bce05a9c ("ASoC: Intel: boards: use devm_clk_get() unconditionally"), enabling pmc_plt_clk_3. However, Strago family Chromebooks use pmc_plt_clk_0 for the codec mclk, resulting in white noise with some digital microphones. Add a DMI-based quirk for Strago family Chromebooks to use pmc_plt_clk_0 instead - mirroring the changes made to cht_bsw_max98090_ti in commit a182ecd3809c ("ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0") and making use of the existing dmi_check_system() call and related infrastructure added in commit 22af29114eb4 ("ASoC: Intel: cht-bsw-rt5645: add quirks for SSP0/AIF1/AIF2 routing"). Signed-off-by: Sam McNally <sammc(a)chromium.org> --- Changes in v2: - Added to the description references to more related commits (similar cht_bsw_max98090_ti clock quirks inspiring this change and and the cht_bsw_rt5645 quirks setup) sound/soc/intel/boards/cht_bsw_rt5645.c | 26 +++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 8879c3be29d5..c68a5b85a4a0 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -48,6 +48,7 @@ struct cht_mc_private { #define CHT_RT5645_SSP2_AIF2 BIT(16) /* default is using AIF1 */ #define CHT_RT5645_SSP0_AIF1 BIT(17) #define CHT_RT5645_SSP0_AIF2 BIT(18) +#define CHT_RT5645_PMC_PLT_CLK_0 BIT(19) static unsigned long cht_rt5645_quirk = 0; @@ -59,6 +60,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0_AIF1 enabled"); if (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2) dev_info(dev, "quirk SSP0_AIF2 enabled"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + dev_info(dev, "quirk PMC_PLT_CLK_0 enabled"); } static int platform_clock_control(struct snd_soc_dapm_widget *w, @@ -226,15 +229,21 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, return 0; } -/* uncomment when we have a real quirk static int cht_rt5645_quirk_cb(const struct dmi_system_id *id) { cht_rt5645_quirk = (unsigned long)id->driver_data; return 1; } -*/ static const struct dmi_system_id cht_rt5645_quirk_table[] = { + { + /* Strago family Chromebooks */ + .callback = cht_rt5645_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_Strago"), + }, + .driver_data = (void *)CHT_RT5645_PMC_PLT_CLK_0, + }, { }, }; @@ -526,6 +535,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) int dai_index = 0; int ret_val = 0; int i; + const char *mclk_name; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -662,11 +672,15 @@ static int snd_cht_mc_probe(struct platform_device *pdev) if (ret_val) return ret_val; - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { - dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + dev_err(&pdev->dev, "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } -- 2.23.0.237.gc6a4ce50a0-goog

3 2

[alsa-devel] [PATCH V2 1/2] ASoC: fsl_mqs: add DT binding documentation
by Shengjiu Wang 01 Oct '19

01 Oct '19

Add the DT binding documentation for NXP MQS driver Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v2 -refine the comments for properties .../devicetree/bindings/sound/fsl,mqs.txt | 36 +++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt new file mode 100644 index 000000000000..40353fc30255 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt @@ -0,0 +1,36 @@ +fsl,mqs audio CODEC + +Required properties: + - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" + "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs". + - clocks : A list of phandles + clock-specifiers, one for each entry in + clock-names + - clock-names : "mclk" - must required. + "core" - required if compatible is "fsl,imx8qm-mqs", it + is for register access. + - gpr : A phandle of General Purpose Registers in IOMUX Controller. + Required if compatible is "fsl,imx6sx-mqs". + +Required if compatible is "fsl,imx8qm-mqs": + - power-domains: A phandle of PM domain provider node. + - reg: Offset and length of the register set for the device. + +Example: + +mqs: mqs { + compatible = "fsl,imx6sx-mqs"; + gpr = <&gpr>; + clocks = <&clks IMX6SX_CLK_SAI1>; + clock-names = "mclk"; + status = "disabled"; +}; + +mqs: mqs@59850000 { + compatible = "fsl,imx8qm-mqs"; + reg = <0x59850000 0x10000>; + clocks = <&clk IMX8QM_AUD_MQS_IPG>, + <&clk IMX8QM_AUD_MQS_HMCLK>; + clock-names = "core", "mclk"; + power-domains = <&pd_mqs0>; + status = "disabled"; +}; -- 2.21.0

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[alsa-devel] [PATCH] ASoC: dt-bindings: mt8183: add missing update
by Tzung-Bi Shih 01 Oct '19

01 Oct '19

Headset codec is optional. Add missing update to DT binding document. Fixes: a962a809e5e4 ("ASoC: mediatek: mt8183: make headset codec optional") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../bindings/sound/mt8183-mt6358-ts3a227-max98357.txt | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt index d6d5207fa996..17ff3892f439 100644 --- a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt +++ b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt @@ -2,9 +2,11 @@ MT8183 with MT6358, TS3A227 and MAX98357 CODECS Required properties: - compatible : "mediatek,mt8183_mt6358_ts3a227_max98357" -- mediatek,headset-codec: the phandles of ts3a227 codecs - mediatek,platform: the phandle of MT8183 ASoC platform +Optional properties: +- mediatek,headset-codec: the phandles of ts3a227 codecs + Example: sound { -- 2.23.0.351.gc4317032e6-goog

2 1

[alsa-devel] [PATCH v2] ASoC: soc-component.h: remove GPL explanation from header
by Kuninori Morimoto 01 Oct '19

01 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc-component.h already has SPDX License, thus, GPL explanation is not needed. This patch removes it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark - this is v2 of "[PATCH] ASoC: soc-component.h: tidyup for soc-component.h" - Subject was renamed - keep "soc.h" - do nothing to header-check include/sound/soc-component.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5d80b2e..1a832695 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -3,10 +3,6 @@ * soc-component.h * * Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SOC_COMPONENT_H #define __SOC_COMPONENT_H -- 2.7.4

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[alsa-devel] [PATCH] ASoC: xlnx: Use devm_platform_ioremap_resource() in xlnx_formatter_pcm_probe()
by Markus Elfring 01 Oct '19

01 Oct '19

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 19 Sep 2019 17:27:57 +0200 Simplify this function implementation by using a known wrapper function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/xilinx/xlnx_formatter_pcm.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 48970efe7838..fb652e73abeb 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -564,7 +564,6 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) int ret; u32 val; struct xlnx_pcm_drv_data *aud_drv_data; - struct resource *res; struct device *dev = &pdev->dev; aud_drv_data = devm_kzalloc(dev, sizeof(*aud_drv_data), GFP_KERNEL); @@ -584,13 +583,7 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) return ret; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "audio formatter node:addr to resource failed\n"); - ret = -ENXIO; - goto clk_err; - } - aud_drv_data->mmio = devm_ioremap_resource(dev, res); + aud_drv_data->mmio = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(aud_drv_data->mmio)) { dev_err(dev, "audio formatter ioremap failed\n"); ret = PTR_ERR(aud_drv_data->mmio); -- 2.23.0

2 1

[alsa-devel] [PATCH v3 1/2] ASoC: dt-bindings: Convert Allwinner A10 codec to a schema
by Maxime Ripard 01 Oct '19

01 Oct '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> The Allwinner SoCs have an embedded audio codec that is supported in Linux, with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that controller over to a YAML schemas. Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Changes from v2: - Change the audio-routing values to an enum, and enforce boundaries on the size - Add restrictions to the possible values of audio-routing Changes from v1: - Fix subject prefix --- .../sound/allwinner,sun4i-a10-codec.yaml | 262 ++++++++++++++++++ .../devicetree/bindings/sound/sun4i-codec.txt | 94 ------- 2 files changed, 262 insertions(+), 94 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml new file mode 100644 index 000000000000..faa75b91c072 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -0,0 +1,262 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - allwinner,sun4i-a10-codec + - allwinner,sun6i-a31-codec + - allwinner,sun7i-a20-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: codec + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + allwinner,audio-routing: + description: |- + A list of the connections between audio components. Each entry + is a pair of strings, the first being the connection's sink, the + second being the connection's source. + allOf: + - $ref: /schemas/types.yaml#definitions/non-unique-string-array + - minItems: 2 + maxItems: 18 + enum: + # Audio Pins on the SoC + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + + # Microphone Biases from the SoC + - HBIAS + - MBIAS + + # Board Connectors + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + allwinner,codec-analog-controls: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the codec analog controls in the PRCM + + allwinner,pa-gpios: + description: GPIO to enable the external amplifier + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + then: + if: + properties: + compatible: + const: allwinner,sun6i-a31-codec + + then: + required: + - resets + - allwinner,audio-routing + + else: + required: + - resets + - allwinner,audio-routing + - allwinner,codec-analog-controls + + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-a23-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-h3-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-v3s-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - MIC1 + - HBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + +additionalProperties: false + +examples: + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; + }; + + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = <0 29 4>; + clocks = <&ccu 61>, <&ccu 135>; + clock-names = "apb", "codec"; + resets = <&ccu 42>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt deleted file mode 100644 index 66579bbd3294..000000000000 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ /dev/null @@ -1,94 +0,0 @@ -* Allwinner A10 Codec - -Required properties: -- compatible: must be one of the following compatibles: - - "allwinner,sun4i-a10-codec" - - "allwinner,sun6i-a31-codec" - - "allwinner,sun7i-a20-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- reg: must contain the registers location and length -- interrupts: must contain the codec interrupt -- dmas: DMA channels for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry - in clock-names. -- clock-names: should contain the following: - - "apb": the parent APB clock for this controller - - "codec": the parent module clock - -Optional properties: -- allwinner,pa-gpios: gpio to enable external amplifier - -Required properties for the following compatibles: - - "allwinner,sun6i-a31-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- resets: phandle to the reset control for this device -- allwinner,audio-routing: A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. Valid names include: - - Audio pins on the SoC: - "HP" - "HPCOM" - "LINEIN" (not on sun8i-v3s) - "LINEOUT" (not on sun8i-a23 or sun8i-v3s) - "MIC1" - "MIC2" (not on sun8i-v3s) - "MIC3" (sun6i-a31 only) - - Microphone biases from the SoC: - "HBIAS" - "MBIAS" (not on sun8i-v3s) - - Board connectors: - "Headphone" - "Headset Mic" - "Line In" - "Line Out" - "Mic" - "Speaker" - -Required properties for the following compatibles: - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- allwinner,codec-analog-controls: A phandle to the codec analog controls - block in the PRCM. - -Example: -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun7i-a20-codec"; - reg = <0x01c22c00 0x40>; - interrupts = <0 30 4>; - clocks = <&apb0_gates 0>, <&codec_clk>; - clock-names = "apb", "codec"; - dmas = <&dma 0 19>, <&dma 0 19>; - dma-names = "rx", "tx"; -}; - -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun6i-a31-codec"; - reg = <0x01c22c00 0x98>; - interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>; - clock-names = "apb", "codec"; - resets = <&ccu RST_APB1_CODEC>; - dmas = <&dma 15>, <&dma 15>; - dma-names = "rx", "tx"; - allwinner,audio-routing = - "Headphone", "HP", - "Speaker", "LINEOUT", - "LINEIN", "Line In", - "MIC1", "MBIAS", - "MIC1", "Mic", - "MIC2", "HBIAS", - "MIC2", "Headset Mic"; -}; -- 2.21.0

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[alsa-devel] [PATCH v2 00/14] ASoC: soc-core cleanup - step 3
by Kuninori Morimoto 01 Oct '19

01 Oct '19

Hi Mark These are v2 of soc-core.c cleanup (maybe) step 3. These try to clenaup around kmalloc/kfree related code. Basically these do nothing from technical point of view. I added +1 extra patch as [00/14] Kuninori Morimoto (14): ASoC: soc-core: move soc_free_pcm_runtime() ASoC: soc-core: merge soc_add_pcm_runtime() into soc_new_pcm_runtime() ASoC: soc-core: call list_del(&rtd->list) at soc_free_pcm_runtime() ASoC: soc-core: create rtd->codec_dais first ASoC: soc-core: merge soc_new_pcm_runtime() and soc_rtd_init() ASoC: soc-core: merge soc_free_pcm_runtime() and soc_rtd_free() ASoC: soc-core: tidyup soc_new_pcm_runtime() alloc order ASoC: soc-core: remove snd_soc_rtdcom_del_all() ASoC: soc-core: use devm_kzalloc() for rtd ASoC: soc-core: remove soc_remove_dai_links() ASoC: soc-core: add soc_setup_card_name() ASoC: soc-core: use devm_xxx for component related resource ASoC: soc-core: setup card->rtd_num at snd_soc_instantiate_card() ASoC: soc-ops: use snd_soc_card_get_kcontrol() at snd_soc_limit_volume() include/sound/soc.h | 1 - sound/soc/soc-core.c | 287 ++++++++++++++++++++++++++------------------------- sound/soc/soc-ops.c | 11 +- 3 files changed, 151 insertions(+), 148 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v7] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 01 Oct '19

01 Oct '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: stm32: sai: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/stm/stm32_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index ef4273361d0d..e20267504b16 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -100,7 +100,7 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); - stm32_sai_pclk_disable(&sai->pdev->dev); + stm32_sai_pclk_disable(&sai->pdev->dev); return -EINVAL; } -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: wcd9335: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is an if statement that is indented one level too deeply, remove the extraneous tabs. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wcd9335.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f318403133e9..f11ffa28683b 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2837,11 +2837,11 @@ static int wcd9335_codec_enable_dec(struct snd_soc_dapm_widget *w, TX_HPF_CUT_OFF_FREQ_MASK) >> 5; snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x10); snd_soc_component_update_bits(comp, dec_cfg_reg, 0x08, 0x00); - if (hpf_coff_freq != CF_MIN_3DB_150HZ) { - snd_soc_component_update_bits(comp, dec_cfg_reg, - TX_HPF_CUT_OFF_FREQ_MASK, - hpf_coff_freq << 5); - } + if (hpf_coff_freq != CF_MIN_3DB_150HZ) { + snd_soc_component_update_bits(comp, dec_cfg_reg, + TX_HPF_CUT_OFF_FREQ_MASK, + hpf_coff_freq << 5); + } break; case SND_SOC_DAPM_POST_PMD: snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x00); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: rt5663: clean up indentation issues
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There are two break statements that are indented one level too deeply, remove the extraneous tabs. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5663.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 2943692f66ed..e6c1ec6c426e 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3644,7 +3644,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_PWR_ANLG_1, RT5663_LDO1_DVO_MASK | RT5663_AMP_HP_MASK, RT5663_LDO1_DVO_0_9V | RT5663_AMP_HP_3X); - break; + break; case CODEC_VER_0: regmap_update_bits(rt5663->regmap, RT5663_DIG_MISC, RT5663_DIG_GATE_CTRL_MASK, RT5663_DIG_GATE_CTRL_EN); @@ -3663,7 +3663,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_TDM_2, RT5663_DATA_SWAP_ADCDAT1_MASK, RT5663_DATA_SWAP_ADCDAT1_LL); - break; + break; default: dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); } -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: amd: acp3x: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.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 bc4dfafdfcd1..ea57088d50ce 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -631,7 +631,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); - return -ENODEV; + return -ENODEV; } adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: core: Clarify usage of ignore_machine
by Daniel Baluta 01 Oct '19

01 Oct '19

For a sound card ignore_machine means that existing FEs links should be ignored and existing BEs links should be overridden with some information from the matching component driver. Current code make some confusions about this so fix it! Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 88978a3036c4..e32a45f6bd88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1853,7 +1853,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) for_each_component(component) { - /* does this component override FEs ? */ + /* does this component override BEs ? */ if (!component->driver->ignore_machine) continue; @@ -1874,7 +1874,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) continue; } - dev_info(card->dev, "info: override FE DAI link %s\n", + dev_info(card->dev, "info: override BE DAI link %s\n", card->dai_link[i].name); /* override platform component */ -- 2.17.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: adau1761: Add PGA Slew time control
by Alexandru Ardelean 01 Oct '19

01 Oct '19

From: Lars-Peter Clausen <lars(a)metafoo.de> The PGA Slew Time control allows to configure the rate with which the PGA gain control ramps up/down to the target setting. The PGA slew control is done via the ALC Control 0 register. There are 2 bits on that reg, that control PGA slew time, while the other bits control parts of the ALC (automatic level control) block. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- sound/soc/codecs/adau1761.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..428bf95ea648 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -29,6 +29,7 @@ #define ADAU1761_REC_MIXER_RIGHT1 0x400d #define ADAU1761_LEFT_DIFF_INPUT_VOL 0x400e #define ADAU1761_RIGHT_DIFF_INPUT_VOL 0x400f +#define ADAU1761_ALC_CTRL0 0x4011 #define ADAU1761_PLAY_LR_MIXER_LEFT 0x4020 #define ADAU1761_PLAY_MIXER_LEFT0 0x401c #define ADAU1761_PLAY_MIXER_LEFT1 0x401d @@ -72,6 +73,7 @@ static const struct reg_default adau1761_reg_defaults[] = { { ADAU1761_REC_MIXER_RIGHT0, 0x00 }, { ADAU1761_REC_MIXER_RIGHT1, 0x00 }, { ADAU1761_LEFT_DIFF_INPUT_VOL, 0x00 }, + { ADAU1761_ALC_CTRL0, 0x00 }, { ADAU1761_RIGHT_DIFF_INPUT_VOL, 0x00 }, { ADAU1761_PLAY_LR_MIXER_LEFT, 0x00 }, { ADAU1761_PLAY_MIXER_LEFT0, 0x00 }, @@ -148,6 +150,21 @@ static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_capture_bias_enum, ADAU17X1_REC_POWER_MGMT, 1, 0x3, adau1761_bias_select_text, adau1761_bias_select_values); +static const unsigned int adau1761_pga_slew_time_values[] = { + 3, 0, 1, 2, +}; + +static const char * const adau1761_pga_slew_time_text[] = { + "Off", + "24 ms", + "48 ms", + "96 ms", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_pga_slew_time_enum, + ADAU1761_ALC_CTRL0, 6, 0x3, adau1761_pga_slew_time_text, + adau1761_pga_slew_time_values); + static const struct snd_kcontrol_new adau1761_jack_detect_controls[] = { SOC_SINGLE("Speaker Auto-mute Switch", ADAU1761_DIGMIC_JACKDETECT, 4, 1, 0), @@ -162,6 +179,8 @@ static const struct snd_kcontrol_new adau1761_differential_mode_controls[] = { SOC_DOUBLE_R_TLV("PGA Boost Capture Volume", ADAU1761_REC_MIXER_LEFT1, ADAU1761_REC_MIXER_RIGHT1, 3, 2, 0, adau1761_pga_boost_tlv), + + SOC_ENUM("PGA Capture Slew Time", adau1761_pga_slew_time_enum), }; static const struct snd_kcontrol_new adau1761_single_mode_controls[] = { @@ -633,6 +652,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg) case ADAU1761_DEJITTER: case ADAU1761_CLK_ENABLE0: case ADAU1761_CLK_ENABLE1: + case ADAU1761_ALC_CTRL0: return true; default: break; -- 2.20.1

2 7

[alsa-devel] [PATCH v2] ASoC: soc-pcm: Use different sequence for start/stop trigger
by Peter Ujfalusi 01 Oct '19

01 Oct '19

On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - Rebased on top 'ASoC: soc-component: fix a couple missing error assignments' Changes since rfc: - Fixed missing error assignments (copy pasted the error) - Document the trigger order in the commit message Regards, Peter sound/soc/soc-pcm.c | 64 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 61 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a1b99ac57d9e..66910500e3b6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,8 +1056,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } @@ -1074,6 +1074,42 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 01 Oct '19

01 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index aa7efd4da1c8..c60e951122c9 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -1982,6 +1982,15 @@ static void hdmi_disable_overflow_interrupts(struct dw_hdmi *hdmi) HDMI_IH_MUTE_FC_STAT2); } +static void dw_hdmi_audio_restore(struct dw_hdmi *hdmi) +{ + mutex_lock(&hdmi->audio_mutex); + + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); + + mutex_unlock(&hdmi->audio_mutex); +} + static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) { int ret; @@ -2045,7 +2054,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + dw_hdmi_audio_restore(hdmi); } /* not for DVI mode */ -- 2.23.0.444.g18eeb5a265-goog

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