October 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 04 Oct '21

04 Oct '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 126 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 33 +++++ 2 files changed, 159 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..4720eac65177 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,126 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + "#sound-dai-cells": + const: 0 + + gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,pll-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_PLL_DONT_SET - leave mode unchanged + - 1 # ADC3XXX_PLL_ENABLE - use the on-chip PLL + - 2 # ADC3XXX_PLL_BYPASS - do not use the on-chip PLL + - 3 # ADC3XXX_PLL_AUTO - use on-chip PLL if possible + default: 0 + description: | + Set on-chip PLL mode. + ADC3XXX_PLL_DONT_SET is intended for use when setting the clock mod + via the clock id in a machine driver and it is not to be changed + from the previous setting. + ADC3XXX_PLL_AUTO will enable on-chip PLL if possible, depending on the + master clock and sample rate combination. + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + ti,pll-mode = <ADC3XXX_PLL_BYPASS>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..8d488a6c3186 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,33 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +/* PLL modes, derived from clk_id of set_sysclk callback, or set in + * devicetree using the ti,pll-mode property. + * + * The default behavior is to take the first matching entry in the clock + * table, which is intended to be the PLL based one if there is more than one. + * + * Setting the clock source using simple-card (clocks or + * system-clock-frequency property) sets clk_id = 0 = ADC3XXX_CLK_DONT_SET, + * which doesn't change whatever clock setting was previously set up. + */ +#define ADC3XXX_PLL_DONT_SET 0 /* Don't change mode */ +#define ADC3XXX_PLL_ENABLE 1 /* Use PLL for clock generation */ +#define ADC3XXX_PLL_BYPASS 2 /* Don't use PLL for clock generation */ +#define ADC3XXX_PLL_AUTO 3 /* Use first available mode */ + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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Re: [PATCH] ASoc: fix ASoC driver to support ops to register get_time_info
by Mark Brown 04 Oct '21

04 Oct '21

On Mon, Oct 04, 2021 at 07:38:22PM +0800, Chien-Yu Lin wrote: > Dear Sir, > > Sorry for bothering you. > > Does it have any status updated, please? Please don't send content free pings and please allow a reasonable time for review. People get busy, go on holiday, attend conferences and so on so unless there is some reason for urgency (like critical bug fixes) please allow at least a couple of weeks for review. If there have been review comments then people may be waiting for those to be addressed. Sending content free pings adds to the mail volume (if they are seen at all) which is often the problem and since they can't be reviewed directly if something has gone wrong you'll have to resend the patches anyway, so sending again is generally a better approach though there are some other maintainers who like them - if in doubt look at how patches for the subsystem are normally handled.

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[PATCH RFC] ASoC: sh: rcar: dma: : use proper DMAENGINE API for termination
by Wolfram Sang 04 Oct '21

04 Oct '21

dmaengine_terminate_all() is deprecated in favor of explicitly saying if it should be sync or async. Here, we want dmaengine_terminate_sync() because there is no other synchronization code in the driver to handle an async case. Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> --- Morimoto-san, can you please double check if this works. I don't know this driver very well. Thank you! sound/soc/sh/rcar/dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 95aa26d62e4f..d1b46057df9e 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -101,7 +101,7 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); if (dmaen->chan) - dmaengine_terminate_all(dmaen->chan); + dmaengine_terminate_sync(dmaen->chan); return 0; } -- 2.30.2

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[PATCH 0/2] Codec driver for TLV320ADC3001 and 3101
by Ricard Wanderlof 04 Oct '21

04 Oct '21

This is a codec driver for the TLV320ADC3001 and TLV320ADC3101 2 channel audio ADC chips from Texas Instruments. Based on an old driver for these chips that I was referred to by Texas Instruments, but which apparently was never upstreamed, I've taken patches from various incarnations, ported them and augmented them with various new features in order to make a working driver. There is currently no support for the on-chip "miniDSP". Although the TLV320ADC3001 and TLV320ADC3101 have similarities with other TI codec chips for which there are existing drivers, upon closer inspection I decided the differences were too great to warrant attempting to modify an existing driver, especially without access to all supported chips and resources for testing them all. There are certainly things that could be improved, among others in the area of PLL register value calculations, but it has been tested on the Axis ARTPEC-8 platform and has been found to work satisfactorily at least with a TLV320ADC3101 operating as an I2S slave at 48 kHz with 32 bit bit depth, 1 or 2 channels, both with the on-chip PLL enabled and disabled. I have not tested this on a TLV320ADC3001 chip, but I've retained the apparent differences from the '3101 (which seem quite small) from the original driver. /Ricard Ricard Wanderlof (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 126 ++ include/dt-bindings/sound/tlv320adc3xxx.h | 33 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1239 +++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 381 +++++ 6 files changed, 1788 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[PATCH] alsaloop: Adding openctl_elem debug log
by Pavel Hofman 04 Oct '21

04 Oct '21

When debugging what ctl elem was found and opened (if any) a debug log is helpful. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- alsaloop/pcmjob.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 8b72af4..845ab82 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1191,6 +1191,11 @@ static void openctl_elem(struct loopback_handle *lhandle, if (err < 0) { snd_ctl_elem_value_free(*elem); *elem = NULL; + } else { + if (verbose) + snd_output_printf(lhandle->loopback->output, + "Opened PCM element %s of %s, device %d, subdevice %d\n", + name, snd_ctl_name(lhandle->ctl), device, subdevice); } } } -- 2.25.1

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[PATCH] ALSA: usb-audio: Enable rate validation for Scarlett devices
by Takashi Iwai 04 Oct '21

04 Oct '21

The Scarlett device series from Focusrite Novation seem requiring the sample rate validations as we've done for MOTU devices; otherwise the driver probes invalid audioformat entries that contain the sample rates that actually don't work, and this may result in an incomplete setup as reported recently. This patch adds the needed quirk flag for enabling the sample rate validation for Focusrite Novation devices. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214493 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6ee6d24c847f..889c855addfc 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1900,6 +1900,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY | QUIRK_FLAG_IFACE_DELAY), VENDOR_FLG(0x07fd, /* MOTU */ QUIRK_FLAG_VALIDATE_RATES), + VENDOR_FLG(0x1235, /* Focusrite Novation */ + QUIRK_FLAG_VALIDATE_RATES), VENDOR_FLG(0x152a, /* Thesycon devices */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x1de7, /* Phoenix Audio */ -- 2.26.2

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[PATCH 0/2] ALSA: hda/realtek: Fix name and add quirk for new Clevo
by Werner Sembach 04 Oct '21

04 Oct '21

Add quirk for Clevo X170KM-G to fix internal speaker not working and rename previously named "Clevo X170" into the more verbose "Clevo X170SM" to make the devices differentiable. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org>

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[PATCH] ASoC: fsl: add COMPILE_TEST for SND_SOC_FSL_ASRC
by Randy Dunlap 04 Oct '21

04 Oct '21

Geert pointed out that since sound/soc has the soc_dummy_driver for NO_DMA platforms, it is possible (desirable) to have drivers that depend on HAS_DMA to alternately depend on COMPILE_TEST. This means that SND_ATMEL_FSL_ASRC can depend on HAS_DMA || COMPIE_TEST. Fixes: 121a01521b1e ("ASoC: fsl: fix build failure") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Suggested-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> Cc: Shengjiu Wang <shengjiu.wang(a)nxp.com> Cc: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20211001.orig/sound/soc/fsl/Kconfig +++ linux-next-20211001/sound/soc/fsl/Kconfig @@ -5,7 +5,7 @@ comment "Common SoC Audio options for Fr config SND_SOC_FSL_ASRC tristate "Asynchronous Sample Rate Converter (ASRC) module support" - depends on HAS_DMA + depends on HAS_DMA || COMPILE_TEST select REGMAP_MMIO select SND_SOC_GENERIC_DMAENGINE_PCM help

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[PATCH 00/13] Extend AHUB audio support for Tegra210 and later
by Sameer Pujar 04 Oct '21

04 Oct '21

Earlier as part of series [0], support for ADMAIF and I/O modules (such as I2S, DMIC and DSPK) was added. This series aims at exposing some of the AHUB internal modules (listed below), which can be used for audio pre or post processing. * SFC (Sampling Frequency Converter) * MVC (Master Volume Control) * AMX (Audio Multiplexer) * ADX (Audio Demultiplexer) * Mixer These modules can be plugged into audio paths and relevant processing can be done. The MUX routes are extended to allow add or remove above modules in the path via mixer controls. This is similar to how specific ADMAIF channels are connected to relevant I/O module instances at the moment. Some of these modules can alter PCM parameters. Consider example of resampler (44.1 -> 48 kHz) in the path. aplay(44.1 kHz) -> ADMAIF -> SFC -> (48 kHz) I2S -> (48kHz) Codec The modules following SFC should be using converted sample rate and DAIs need to be configured accordingly. The audio-graph driver provides a mechanism to fixup the new parameters which can be specified in DT for a given DAI. Then core uses these new values via fixup callback and then pass it to respective DAIs hw_param() callback. The "convert-rate", described in [1], property can be used when there is rate conversion in the audio path. Similarly "convert-channels" can be used when there is channel conversion in the path. There is no "convert-xxx" property for sample size conversions. It can be added if necessary. [0] https://www.lkml.org/lkml/2020/7/21/1357 [1] Documentation/devicetree/bindings/sound/audio-graph-port.yaml Sameer Pujar (13): ASoC: soc-pcm: Don't reconnect an already active BE ASoC: simple-card-utils: Increase maximum DAI links limit to 512 ASoC: audio-graph: Fixup CPU endpoint hw_params in a BE<->BE link ASoC: dt-bindings: tegra: Few more Tegra210 AHUB modules ASoC: tegra: Add routes for few AHUB modules ASoC: tegra: Add Tegra210 based MVC driver ASoC: tegra: Add Tegra210 based SFC driver ASoC: tegra: Add Tegra210 based AMX driver ASoC: tegra: Add Tegra210 based ADX driver ASoC: tegra: Add Tegra210 based Mixer driver arm64: defconfig: Enable few Tegra210 based AHUB drivers arm64: tegra: Add few AHUB devices for Tegra210 and later arm64: tegra: Extend APE audio support on Jetson platforms .../bindings/sound/nvidia,tegra210-adx.yaml | 74 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 20 + .../bindings/sound/nvidia,tegra210-amx.yaml | 72 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 67 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 79 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 76 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 1554 ++++++++- arch/arm64/boot/dts/nvidia/tegra186.dtsi | 120 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 1493 ++++++++- .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 1520 ++++++++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 116 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 77 + arch/arm64/configs/defconfig | 5 + include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 4 +- sound/soc/soc-pcm.c | 4 + sound/soc/tegra/Kconfig | 48 + sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra210_adx.c | 527 +++ sound/soc/tegra/tegra210_adx.h | 72 + sound/soc/tegra/tegra210_ahub.c | 511 ++- sound/soc/tegra/tegra210_amx.c | 595 ++++ sound/soc/tegra/tegra210_amx.h | 93 + sound/soc/tegra/tegra210_mixer.c | 667 ++++ sound/soc/tegra/tegra210_mixer.h | 100 + sound/soc/tegra/tegra210_mvc.c | 629 ++++ sound/soc/tegra/tegra210_mvc.h | 117 + sound/soc/tegra/tegra210_sfc.c | 3542 ++++++++++++++++++++ sound/soc/tegra/tegra210_sfc.h | 78 + 31 files changed, 13601 insertions(+), 423 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml create mode 100644 sound/soc/tegra/tegra210_adx.c create mode 100644 sound/soc/tegra/tegra210_adx.h create mode 100644 sound/soc/tegra/tegra210_amx.c create mode 100644 sound/soc/tegra/tegra210_amx.h create mode 100644 sound/soc/tegra/tegra210_mixer.c create mode 100644 sound/soc/tegra/tegra210_mixer.h create mode 100644 sound/soc/tegra/tegra210_mvc.c create mode 100644 sound/soc/tegra/tegra210_mvc.h create mode 100644 sound/soc/tegra/tegra210_sfc.c create mode 100644 sound/soc/tegra/tegra210_sfc.h -- 2.7.4

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mpv segfaults in a52 plugin
by GitHub issues - opened 02 Oct '21

02 Oct '21

alsa-project/alsa-plugins issue #35 was opened from ykurlaev: When using mpv with the option `audio-device=alsa/<a52 device>` it gets a segfault on this line https://github.com/alsa-project/alsa-plugins/blob/da157e978d73f1947bb77637c… because it calls `dump` before `prepare` has been called and `rec->avctx` is NULL. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/35 Repository URL: https://github.com/alsa-project/alsa-plugins

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