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

[PATCH v3] soundwire: SDCA: add helper macro to access controls
by Bard Liao 30 Oct '20

30 Oct '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The upcoming SDCA (SoundWire Device Class Audio) specification defines a hierarchical encoding to interface with Class-defined capabilities. The specification is not yet accessible to the general public but this information is released with explicit permission from the MIPI Board to avoid delays with SDCA support on Linux platforms. A block of 64 MBytes of register addresses are allocated to SDCA controls, starting at address 0x40000000. The 26 LSBs which identify individual controls are set based on the following variables: - Function Number. An SCDA device can be split in up to 8 independent Functions. Each of these Functions is described in the SDCA specification, e.g. Smart Amplifier, Smart Microphone, Simple Microphone, Jack codec, HID, etc. - Entity Number. Within each Function, an Entity is an identifiable block. Up to 127 Entities are connected in a pre-defined graph (similar to USB), with Entity0 reserved for Function-level configurations. In contrast to USB, the SDCA spec pre-defines Function Types, topologies, and allowed options, i.e. the degree of freedom is not unlimited to limit the possibility of errors in descriptors leading to software quirks. - Control Selector. Within each Entity, the SDCA specification defines 48 controls such as Mute, Gain, AGC, etc, and 16 implementation defined ones. Some Control Selectors might be used for low-level platform setup, and other exposed to applications and users. Note that the same Control Selector capability, e.g. Latency control, might be located at different offsets in different entities, the Control Selector mapping is Entity-specific. - Control Number. Some Control Selectors allow channel-specific values to be set, with up to 64 channels allowed. This is mostly used for volume control. - Current/Next values. Some Control Selectors are 'Dual-Ranked'. Software may either update the Current value directly for immediate effect. Alternatively, software may write into the 'Next' values and update the SoundWire 1.2 'Commit Groups' register to copy 'Next' values into 'Current' ones in a synchronized manner. This is different from bank switching which is typically used to change the bus configuration only. - MBQ. the Multi-Byte Quantity bit is used to provide atomic updates when accessing more that one byte, for example a 16-bit volume control would be updated consistently, the intermediate values mixing old MSB with new LSB are not applied. These 6 parameters are used to build a 32-bit address to access the desired Controls. Because of address range, paging is required, but the most often used parameter values are placed in the lower 16 bits of the address. This helps to keep the paging registers constant while updating Controls for a specific Device/Function. Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- Changelog: v2: - add SDW_SDCA_MBQ_CTL v3: - add SDW_SDCA_NEXT_CTL --- include/linux/soundwire/sdw_registers.h | 32 +++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/include/linux/soundwire/sdw_registers.h b/include/linux/soundwire/sdw_registers.h index f420e8059779..e14dff9a9c7f 100644 --- a/include/linux/soundwire/sdw_registers.h +++ b/include/linux/soundwire/sdw_registers.h @@ -298,4 +298,36 @@ #define SDW_CASC_PORT_MASK_INTSTAT3 1 #define SDW_CASC_PORT_REG_OFFSET_INTSTAT3 2 +/* + * v1.2 device - SDCA address mapping + * + * Spec definition + * Bits Contents + * 31 0 (required by addressing range) + * 30:26 0b10000 (Control Prefix) + * 25 0 (Reserved) + * 24:22 Function Number [2:0] + * 21 Entity[6] + * 20:19 Control Selector[5:4] + * 18 0 (Reserved) + * 17:15 Control Number[5:3] + * 14 Next + * 13 MBQ + * 12:7 Entity[5:0] + * 6:3 Control Selector[3:0] + * 2:0 Control Number[2:0] + */ + +#define SDW_SDCA_CTL(fun, ent, ctl, ch) (BIT(30) | \ + (((fun) & 0x7) << 22) | \ + (((ent) & 0x40) << 15) | \ + (((ent) & 0x3f) << 7) | \ + (((ctl) & 0x30) << 15) | \ + (((ctl) & 0x0f) << 3) | \ + (((ch) & 0x38) << 12) | \ + ((ch) & 0x07)) + +#define SDW_SDCA_MBQ_CTL(reg) ((reg) | BIT(13)) +#define SDW_SDCA_NEXT_CTL(reg) ((reg) | BIT(14)) + #endif /* __SDW_REGISTERS_H */ -- 2.17.1

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[PATCH] ALSA: ac97: fix reference leak in ac97_bus_remove
by Zhang Qilong 30 Oct '20

30 Oct '20

pm_runtime_get_sync() will increase pm usage counter even it failed. We should call pm_runtime_put_noidle to decrease it to prevent reference leak. Signed-off-by: Zhang Qilong <zhangqilong3(a)huawei.com> --- sound/ac97/bus.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 7985dd8198b6..2c4779d851ad 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -521,8 +521,11 @@ static int ac97_bus_remove(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + dev_err(dev, "Failed to enable device: %d\n", ret); + pm_runtime_put_noidle(dev); return ret; + } ret = adrv->remove(adev); pm_runtime_put_noidle(dev); -- 2.17.1

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[PATCH v4 00/15] Audio graph card updates and usage with Tegra210 audio
by Sameer Pujar 30 Oct '20

30 Oct '20

Summary of changes: * Support multiple instances of a component. For example there can be multiple I2S devices which can use the same component driver. * Support open platforms with empty Codec endpoint. Customers can plug their own HW and can populate codec endpoint. * In a component model there can be many components which can be connected togethe. In such cases Identify no-pcm DPCM DAI links which can be used in BE<->BE connections. * Add Tegra audio graph driver which is based on generic audio graph driver and specific customizations are done in Tegra driver. * This pushes DT support for Tegra210 based platforms which uses audio-graph card and above enhancements. The series is based on following references where DPCM usgae for Tegra Audio and simple-card driver proposal were discussed. * https://lkml.org/lkml/2020/4/30/519 (DPCM for Tegra) * https://lkml.org/lkml/2020/6/27/4 (simple-card driver) Changelog ========= v3 -> v4 -------- * Added new patches to convert graph.txt and audio-graph-card.txt to corresponding json-schema files. Later these references are used in Tegra audio graph schema. * AHUB component binding docs are updated to reflect the usage of ports/port/endpoint * More common stuff is moved into graph_parse_of() and this is used by both generic and Tegra audio graph. * DT binding for Tegra audio graph is updated to included "ports { }" * As per the suggestion 'void *data' member is dropped from 'asoc_simple_priv' and instead container method is used to maintain required custom data internal to Tegra audio graph. v2 -> v3 -------- * Dropped new compatible addition in generic graph driver after reviewing it with Morimoto-san. Instead added Tegra audio graph driver and new compatibles are added in the same. * Added new patches to expose new members for customization in audio graph driver. * Added new patch for Tegra audio graph driver and related documentation. * Minor change in below commit where mutex version of helper is used "ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM" * DT binding is updated to use the newly exposed compatibles * No changes in other patches v1 -> v2 -------- * Re-organized ports/endpoints description for ADMAIF and XBAR. Updated DT patches accordingly. * After above change, multiple Codec endpoint support is not required and hence dropped for now. This will be considered separately if at all required in future. * Re-ordered patches in the series. Sameer Pujar (15): ASoC: soc-core: Fix component name_prefix parsing ASoC: soc-pcm: Get all BEs along DAPM path ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM ASoC: audio-graph: Support empty Codec endpoint ASoC: audio-graph: Expose new members for asoc_simple_priv ASoC: audio-graph: Expose helpers from audio graph Documentation: of: Convert graph bindings to json-schema ASoC: dt-bindings: audio-graph: Convert bindings to json-schema ASoC: dt-bindings: tegra: Add graph bindings ASoC: dt-bindings: tegra: Add json-schema for Tegra audio graph card ASoC: tegra: Add audio graph based card driver arm64: defconfig: Enable Tegra audio graph card driver arm64: tegra: Audio graph header for Tegra210 arm64: tegra: Audio graph sound card for Jetson Nano and TX1 Documentation/devicetree/bindings/graph.txt | 128 ----- Documentation/devicetree/bindings/graph.yaml | 170 +++++++ .../devicetree/bindings/sound/audio-graph-card.txt | 337 ------------- .../bindings/sound/audio-graph-card.yaml | 548 +++++++++++++++++++++ .../sound/nvidia,tegra-audio-graph-card.yaml | 158 ++++++ .../bindings/sound/nvidia,tegra186-dspk.yaml | 7 + .../bindings/sound/nvidia,tegra210-admaif.yaml | 7 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 7 + .../bindings/sound/nvidia,tegra210-dmic.yaml | 7 + .../bindings/sound/nvidia,tegra210-i2s.yaml | 7 + .../boot/dts/nvidia/tegra210-audio-graph.dtsi | 153 ++++++ arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 262 ++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 146 ++++++ arch/arm64/configs/defconfig | 1 + include/sound/graph_card.h | 16 + include/sound/simple_card_utils.h | 3 + include/sound/soc.h | 1 + sound/soc/generic/audio-graph-card.c | 175 ++++--- sound/soc/soc-core.c | 3 +- sound/soc/soc-pcm.c | 3 +- sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_audio_graph_card.c | 255 ++++++++++ 23 files changed, 1882 insertions(+), 523 deletions(-) delete mode 100644 Documentation/devicetree/bindings/graph.txt create mode 100644 Documentation/devicetree/bindings/graph.yaml delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi create mode 100644 include/sound/graph_card.h create mode 100644 sound/soc/tegra/tegra_audio_graph_card.c -- 2.7.4

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[PATCH v3 1/2] ASoC: tegra20-spdif: remove "default m"
by Michał Mirosław 30 Oct '20

30 Oct '20

Make tegra20-spdif default to N as all other drivers do. Signed-off-by: Michał Mirosław <mirq-linux(a)rere.qmqm.pl> Fixes: 774fec338bfc ("ASoC: Tegra: Implement SPDIF CPU DAI") --- v3: split-off the defconfig changes v2: add the symbol to defconfig as suggested by Thierry Reding --- sound/soc/tegra/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 3d91bd3e59cd..a62cc87551ac 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -39,7 +39,6 @@ config SND_SOC_TEGRA20_I2S config SND_SOC_TEGRA20_SPDIF tristate "Tegra20 SPDIF interface" depends on SND_SOC_TEGRA - default m help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support -- 2.20.1

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[PATCH v9 00/14] Add Allwinner H3/H5/H6/A64 HDMI audio
by Clément Péron 30 Oct '20

30 Oct '20

Hi, This series add H6 I2S support and the I2S node missing to support HDMI audio in different Allwinner SoC. As we first use some TDM property to make the I2S working we the simple soundcard. We have now drop this simple sound card and a proper dedicated soundcard will be introduce later. This make the title of this series wrong now :/. Regards, Clement Change since v8: - move the comment near the function prototype - collect Maxime Ripard tags Change since v7: - rebase on next-20201026 - comment about slots and slot_width Change since v6: - move set_channel_cfg() in first position - convert return value to decimal Change since v5: - Drop HDMI simple soundcard - Collect Chen-Yu Tsai tags - Configure channels from 9 to 15. - Remove DMA RX for H3/H5 - Fix Documentation for H3/H5 Change since v4: - add more comment on get_wss() and set_channel_cfg() patch - merge soundcard and DAI HDMI patches Change since v3: - add Samuel Holland patch to reconfigure FIFO_TX_REG when suspend is enabled - readd inversion to H6 LRCK sun50i_h6_i2s_set_soc_fmt() - Fix get_wss() for sun4i - Add a commit to fix checkpatch warning Change since v2: - rebase on next-20200918 - drop revert LRCK polarity patch - readd simple-audio-card,frame-inversion in dts - Add patch for changing set_chan_cfg params Change since v1: - rebase on next-20200828 - add revert LRCK polarity - remove all simple-audio-card,frame-inversion in dts - add Ondrej patches for Orange Pi board - Add arm64 defconfig patch Clément Péron (6): ASoC: sun4i-i2s: Change set_chan_cfg() params ASoC: sun4i-i2s: Change get_sr() and get_wss() to be more explicit ASoC: sun4i-i2s: Fix sun8i volatile regs ASoC: sun4i-i2s: fix coding-style for callback definition arm64: defconfig: Enable Allwinner i2s driver dt-bindings: sound: sun4i-i2s: Document H3 with missing RX channel possibility Jernej Skrabec (3): ASoC: sun4i-i2s: Add support for H6 I2S dt-bindings: ASoC: sun4i-i2s: Add H6 compatible arm64: dts: allwinner: h6: Add I2S1 node Marcus Cooper (4): ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support arm64: dts: allwinner: a64: Add I2S2 node arm: dts: sunxi: h3/h5: Add I2S2 node Samuel Holland (1): ASoC: sun4i-i2s: Fix setting of FIFO modes .../sound/allwinner,sun4i-a10-i2s.yaml | 6 +- arch/arm/boot/dts/sunxi-h3-h5.dtsi | 13 + arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 14 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 13 + arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-i2s.c | 384 +++++++++++++++--- 6 files changed, 376 insertions(+), 55 deletions(-) -- 2.25.1

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[PATCH] ASoC: nau8315: add codec driver
by David Lin 30 Oct '20

30 Oct '20

Add codec driver for Nuvoton NAU8315. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8315.txt | 18 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8315.c | 166 ++++++++++++++++++ 4 files changed, 191 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8315.txt create mode 100644 sound/soc/codecs/nau8315.c diff --git a/Documentation/devicetree/bindings/sound/nau8315.txt b/Documentation/devicetree/bindings/sound/nau8315.txt new file mode 100644 index 000000000000..6eaec46f384c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8315.txt @@ -0,0 +1,18 @@ +Nuvoton NAU8315 Mono Class-D Amplifier + +Required properties: +- compatible : "nuvoton,nau8315" + +Optional properties: +- enable-gpios : GPIO specifier for the chip's device enable input(EN) pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). + +Example: + +#include <dt-bindings/gpio/gpio.h> + +nau8315 { + compatible = "nuvoton,nau8315"; + enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5791b7056af6..2b6eceb3573c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -130,6 +130,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_MT6358 imply SND_SOC_MT6359 imply SND_SOC_MT6660 + imply SND_SOC_NAU8315 imply SND_SOC_NAU8540 imply SND_SOC_NAU8810 imply SND_SOC_NAU8822 @@ -1760,6 +1761,10 @@ config SND_SOC_MT6660 Select N if you don't have MT6660 on board. Select M to build this as module. +config SND_SOC_NAU8315 + tristate "Nuvoton Technology Corporation NAU8315 CODEC" + depends on GPIOLIB + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 11ce98c25d6c..144d9256d4d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -129,6 +129,7 @@ snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o snd-soc-mt6359-objs := mt6359.o snd-soc-mt6660-objs := mt6660.o +snd-soc-nau8315-objs := nau8315.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -438,6 +439,7 @@ obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o obj-$(CONFIG_SND_SOC_MT6359) += snd-soc-mt6359.o obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o +obj-$(CONFIG_SND_SOC_NAU8315) += snd-soc-nau8315.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/nau8315.c b/sound/soc/codecs/nau8315.c new file mode 100644 index 000000000000..e6bc5c0a5036 --- /dev/null +++ b/sound/soc/codecs/nau8315.c @@ -0,0 +1,166 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// nau8315.c -- NAU8315 ALSA SoC Audio Amplifier Driver +// +// Copyright 2020 Nuvoton Technology Crop. +// +// Author: David Lin <ctlin0(a)nuvoton.com> +// +// Based on MAX98357A.c + +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/err.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> + +struct nau8315_priv { + struct gpio_desc *enable; + int enpin_switch; +}; + +static int nau8315_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8315_priv *nau8315 = + snd_soc_component_get_drvdata(component); + + if (!nau8315->enable) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (nau8315->enpin_switch) { + gpiod_set_value(nau8315->enable, 1); + dev_dbg(component->dev, "set enable to 1"); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + gpiod_set_value(nau8315->enable, 0); + dev_dbg(component->dev, "set enable to 0"); + break; + } + + return 0; +} + +static int nau8315_enpin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8315_priv *nau8315 = + snd_soc_component_get_drvdata(component); + + if (event & SND_SOC_DAPM_POST_PMU) + nau8315->enpin_switch = 1; + else if (event & SND_SOC_DAPM_POST_PMD) + nau8315->enpin_switch = 0; + + return 0; +} + +static const struct snd_soc_dapm_widget nau8315_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("Speaker"), + SND_SOC_DAPM_OUT_DRV_E("EN_Pin", SND_SOC_NOPM, 0, 0, NULL, 0, + nau8315_enpin_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route nau8315_dapm_routes[] = { + {"EN_Pin", NULL, "HiFi Playback"}, + {"Speaker", NULL, "EN_Pin"}, +}; + +static const struct snd_soc_component_driver nau8315_component_driver = { + .dapm_widgets = nau8315_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8315_dapm_widgets), + .dapm_routes = nau8315_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8315_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct snd_soc_dai_ops nau8315_dai_ops = { + .trigger = nau8315_daiops_trigger, +}; + +#define NAU8315_RATES SNDRV_PCM_RATE_8000_96000 +#define NAU8315_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE) + +static struct snd_soc_dai_driver nau8315_dai_driver = { + .name = "nau8315-hifi", + .playback = { + .stream_name = "HiFi Playback", + .formats = NAU8315_FORMATS, + .rates = NAU8315_RATES, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &nau8315_dai_ops, +}; + +static int nau8315_platform_probe(struct platform_device *pdev) +{ + struct nau8315_priv *nau8315; + + nau8315 = devm_kzalloc(&pdev->dev, sizeof(*nau8315), GFP_KERNEL); + if (!nau8315) + return -ENOMEM; + + nau8315->enable = devm_gpiod_get_optional(&pdev->dev, + "enable", GPIOD_OUT_LOW); + if (IS_ERR(nau8315->enable)) + return PTR_ERR(nau8315->enable); + + dev_set_drvdata(&pdev->dev, nau8315); + + return devm_snd_soc_register_component(&pdev->dev, + &nau8315_component_driver, + &nau8315_dai_driver, 1); +} + +#ifdef CONFIG_OF +static const struct of_device_id nau8315_device_id[] = { + { .compatible = "nuvoton,nau8315" }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8315_device_id); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id nau8315_acpi_match[] = { + { "NVTN2010", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8315_acpi_match); +#endif + +static struct platform_driver nau8315_platform_driver = { + .driver = { + .name = "nau8315", + .of_match_table = of_match_ptr(nau8315_device_id), + .acpi_match_table = ACPI_PTR(nau8315_acpi_match), + }, + .probe = nau8315_platform_probe, +}; +module_platform_driver(nau8315_platform_driver); + +MODULE_DESCRIPTION("ASoC NAU8315 Mono Class-D Amplifier Driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); -- 2.25.1 =========================================================================================== The privileged confidential information contained in this email is intended for use only by the addressees as indicated by the original sender of this email. If you are not the addressee indicated in this email or are not responsible for delivery of the email to such a person, please kindly reply to the sender indicating this fact and delete all copies of it from your computer and network server immediately. Your cooperation is highly appreciated. It is advised that any unauthorized use of confidential information of Nuvoton is strictly prohibited; and any information in this email irrelevant to the official business of Nuvoton shall be deemed as neither given nor endorsed by Nuvoton.

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[PATCH v2 1/2] ASoC: dt-bindings: fsl_aud2htx: Add binding doc for aud2htx module
by Shengjiu Wang 30 Oct '20

30 Oct '20

AUD2HTX (Audio Subsystem TO HDMI TX Subsystem) is a new IP module found on i.MX8MP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - fix indentation issue - remove nodename .../bindings/sound/fsl,aud2htx.yaml | 64 +++++++++++++++++++ 1 file changed, 64 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml new file mode 100644 index 000000000000..6d9ba2946bfb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml @@ -0,0 +1,64 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,aud2htx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Subsystem to HDMI RTX Subsystem Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + compatible: + const: fsl,imx8mp-aud2htx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: bus + + dmas: + items: + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: tx + + power-domains: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mp-clock.h> + + aud2htx: aud2htx@30cb0000 { + compatible = "fsl,imx8mp-aud2htx"; + reg = <0x30cb0000 0x10000>; + interrupts = <GIC_SPI 130 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUD2HTX_IPG>; + clock-names = "bus"; + dmas = <&sdma2 26 2 0>; + dma-names = "tx"; + power-domains = <&audiomix_pd>; + }; -- 2.27.0

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[PATCH v2 0/2] ASoC: soc-compress: tidyup STREAM vs COMPRESS
by Kuninori Morimoto 30 Oct '20

30 Oct '20

Hi Mark These are v2 of tidyup patch-set for SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx. soc-compress is using both SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx, but mixed use. This is confusable, but no problem. Because these are defined as UAPI and are using same value. This patch-set make sure these are same value. Kuninori Morimoto (2): ASoC: soc-compress: tidyup STREAM vs COMPRESS ASoC: soc-compress: assume SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx are same sound/soc/soc-compress.c | 63 +++++++++++++++------------------------- 1 file changed, 23 insertions(+), 40 deletions(-) -- 2.25.1

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[PATCH 0/2] ASoC: soc-compress: tidyup STREAM vs COMPRESS
by Kuninori Morimoto 30 Oct '20

30 Oct '20

Hi Mark soc-compress is using both SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx. It needs to be converted, but some of them didn't. This is bug, but this bug does nothing because these are using same value. This patch-set cleanup these. Kuninori Morimoto (2): ASoC: soc-compress: tidyup STREAM vs COMPRESS ASoC: soc-compress: add soc_compr_cstream_to_stream() sound/soc/soc-compress.c | 65 ++++++++++++++++------------------------ 1 file changed, 25 insertions(+), 40 deletions(-) -- 2.25.1

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[PATCH v2] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 30 Oct '20

30 Oct '20

With a stream with low bitrate, user can't pause or resume the stream near the end of the stream because current ALSA doesn't allow it. If the stream has very low bitrate enough to store whole stream into the buffer, user can't do anything except stop the stream and then restart it from the first because most of applications call draining after sending last frame to the kernel. If pause, resume are allowed during draining, user experience can be enhanced. To prevent malfunction in HW drivers which don't support pause during draining, pause during draining will only work if HW driver enable this feature explicitly by calling snd_compr_use_pause_in_draining(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- include/sound/compress_driver.h | 17 +++++++++++++ sound/core/compress_offload.c | 44 +++++++++++++++++++++++++++------ 2 files changed, 53 insertions(+), 8 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 70cbc5095e72..5a0d6737de5e 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -67,6 +67,7 @@ struct snd_compr_runtime { * @metadata_set: metadata set flag, true when set * @next_track: has userspace signal next track transition, true when set * @partial_drain: undergoing partial_drain for stream, true when set + * @pause_in_draining: paused during draining state, true when set * @private_data: pointer to DSP private data * @dma_buffer: allocated buffer if any */ @@ -80,6 +81,7 @@ struct snd_compr_stream { bool metadata_set; bool next_track; bool partial_drain; + bool pause_in_draining; void *private_data; struct snd_dma_buffer dma_buffer; }; @@ -142,6 +144,7 @@ struct snd_compr_ops { * @direction: Playback or capture direction * @lock: device lock * @device: device id + * @use_pause_in_draining: allow pause in draining, true when set */ struct snd_compr { const char *name; @@ -152,6 +155,7 @@ struct snd_compr { unsigned int direction; struct mutex lock; int device; + bool use_pause_in_draining; #ifdef CONFIG_SND_VERBOSE_PROCFS /* private: */ char id[64]; @@ -166,6 +170,19 @@ int snd_compress_deregister(struct snd_compr *device); int snd_compress_new(struct snd_card *card, int device, int type, const char *id, struct snd_compr *compr); +/** + * snd_compr_use_pause_in_draining - Allow pause and resume in draining state + * @substream: compress substream to set + * + * Allow pause and resume in draining state. + * Only HW driver supports this transition can call this API. + */ +static inline void snd_compr_use_pause_in_draining( + struct snd_compr_stream *substream) +{ + substream->device->use_pause_in_draining = true; +} + /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the * framework know that a fragment has been consumed from the ring buffer diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 0e53f6f31916..a071485383ed 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -708,11 +708,24 @@ static int snd_compr_pause(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->pause_in_draining = true; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + } return retval; } @@ -720,11 +733,25 @@ static int snd_compr_resume(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_PAUSED: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining || + !stream->pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->pause_in_draining = false; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + } return retval; } @@ -767,6 +794,7 @@ static int snd_compr_stop(struct snd_compr_stream *stream) /* clear flags and stop any drain wait */ stream->partial_drain = false; stream->metadata_set = false; + stream->pause_in_draining = false; snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; -- 2.21.0

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