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

[PATCH AUTOSEL 5.15 38/82] ALSA: usb-audio: disable implicit feedback sync for Behringer UFX1204 and UFX1604
by Sasha Levin 09 Nov '21

09 Nov '21

From: Geraldo Nascimento <geraldogabriel(a)gmail.com> [ Upstream commit 28c369e60827f706cef4604a3e2848198f25bd26 ] Behringer UFX1204 and UFX1604 have Synchronous endpoints to which current ALSA code applies implicit feedback sync as if they were Asynchronous endpoints. This breaks UAC compliance and is unneeded. The commit 5e35dc0338d85ccebacf3f77eca1e5dea73155e8 and subsequent 1a15718b41df026cffd0e42cfdc38a1384ce19f9 were meant to clear up noise. Unfortunately, noise persisted for those using higher sample rates and this was only solved by commit d2e8f641257d0d3af6e45d6ac2d6f9d56b8ea964 Since there are no more reports of noise, let's get rid of the implicit-fb quirks breaking UAC compliance. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> Link: https://lore.kernel.org/r/YVYSnoQ7nxLXT0Dq@geday Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/implicit.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 23767a14d1266..70319c822c10b 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -54,8 +54,6 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* Fixed EP */ /* FIXME: check the availability of generic matching */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0001, 0x81, 1), /* Behringer UFX1604 */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0002, 0x81, 1), /* Behringer UFX1204 */ IMPLICIT_FB_FIXED_DEV(0x2466, 0x8010, 0x81, 2), /* Fractal Audio Axe-Fx III */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0001, 0x81, 2), /* Solid State Logic SSL2 */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0002, 0x81, 2), /* Solid State Logic SSL2+ */ -- 2.33.0

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[PATCH AUTOSEL 5.15 36/82] ASoC: SOF: Intel: hda-dai: fix potential locking issue
by Sasha Levin 09 Nov '21

09 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit a20f3b10de61add5e14b6ce4df982f4df2a4cbbc ] The initial hdac_stream code was adapted a third time with the same locking issues. Move the spin_lock outside the loops and make sure the fields are protected on read/write. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Acked-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210924192417.169243-5-pierre-louis.bossart@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-dai.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index c1f9f0f584647..6704dbcd101cd 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -68,6 +68,7 @@ static struct hdac_ext_stream * return NULL; } + spin_lock_irq(&bus->reg_lock); list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = stream_to_hdac_ext_stream(stream); @@ -107,12 +108,12 @@ static struct hdac_ext_stream * * is updated in snd_hdac_ext_stream_decouple(). */ if (!res->decoupled) - snd_hdac_ext_stream_decouple(bus, res, true); - spin_lock_irq(&bus->reg_lock); + snd_hdac_ext_stream_decouple_locked(bus, res, true); + res->link_locked = 1; res->link_substream = substream; - spin_unlock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); return res; } -- 2.33.0

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[PATCH AUTOSEL 5.15 14/82] ASoC: mediatek: mt8195: Add missing of_node_put()
by Sasha Levin 09 Nov '21

09 Nov '21

From: Bixuan Cui <cuibixuan(a)huawei.com> [ Upstream commit b2fc2c92d2fd34d93268f677e514936f50dd6b5c ] The platform_node is returned by of_parse_phandle() should have of_node_put() before return. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Bixuan Cui <cuibixuan(a)huawei.com> Link: https://lore.kernel.org/r/20210911081246.33867-1-cuibixuan@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index de09f67c04502..a3fa8efc8f81c 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -1040,8 +1040,10 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) } priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) + if (!priv) { + of_node_put(platform_node); return -ENOMEM; + } snd_soc_card_set_drvdata(card, priv); @@ -1049,6 +1051,8 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) if (ret) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + + of_node_put(platform_node); return ret; } -- 2.33.0

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aaf plugin not working raspberry pi cm4
by GitHub issues - edited 09 Nov '21

09 Nov '21

alsa-project/alsa-plugins issue #39 was edited from steeley: trying to send to the ALSA device aaf0. pI cm4 with kernel setup to support avb/tsn correctly. (Linux cm432 5.10.76-v7l-AVB_KERNEL+ #3 SMP Sat Nov 6 22:50:41 GMT 2021 armv7l GNU/Linux). PCIe i210 network card is the interface set up correctly according to: https://tsn.readthedocs.io/avb.html Error is: `sudo speaker-test -p 2500 -F S16_BE -c 2 -r 48000 -D aaf0 speaker-test 1.1.9 Playback device is aaf0 Stream parameters are 48000Hz, S16_BE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 1 to 206158430 Period size range from 0 to 206158431 Requested period time 2500 us Periods = 4 was set period_size = 120 was set buffer_size = 480 0 - Front Left ALSA lib pcm_aaf.c:670:(aaf_tx_pdu) Failed to send AAF PDU Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Input/output error ` Issue URL : https://github.com/alsa-project/alsa-plugins/issues/39 Repository URL: https://github.com/alsa-project/alsa-plugins

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aaf plugin not working raspberry pi cm4
by GitHub issues - opened 09 Nov '21

09 Nov '21

alsa-project/alsa-plugins issue #39 was opened from steeley: trying to send to the ALSA device aaf0. pI cm4 with kernel setup to support avb/tsn correctly. PCIe i210 network card is the interface set up correctly according to: https://tsn.readthedocs.io/avb.html Error is: `sudo speaker-test -p 2500 -F S16_BE -c 2 -r 48000 -D aaf0 speaker-test 1.1.9 Playback device is aaf0 Stream parameters are 48000Hz, S16_BE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 1 to 206158430 Period size range from 0 to 206158431 Requested period time 2500 us Periods = 4 was set period_size = 120 was set buffer_size = 480 0 - Front Left ALSA lib pcm_aaf.c:670:(aaf_tx_pdu) Failed to send AAF PDU Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Input/output error ` Issue URL : https://github.com/alsa-project/alsa-plugins/issues/39 Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH v4 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 09 Nov '21

09 Nov '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 | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 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..c4fed6335230 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# 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 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + 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. + + 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. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +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>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + 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..59ed510e6cf1 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#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 */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#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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[PATCH] ALSA: ctl: Fix copy of updated id with element read/write
by Alan Young 09 Nov '21

09 Nov '21

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[PATCH v4 0/2] Codec driver for TLV320ADC3001 and 3101
by Ricard Wanderlof 09 Nov '21

09 Nov '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; in the original driver the only differences were related to the now-removed initialization code, so I've just retained the compatible strings for both chips. Limitations: - GPIO pins on chip can only be set up as outputs. - Using BCLK as an input to clock chain is supported by the hw but not by driver. Changes v3->v4: More fixes for issues reported by kernel test robot (lkp(a)intel.com). Don't know why these weren't reported the first time. Changes v2->v3: Fix a couple of issues reported by the kernel test robot (lkp(a)intel.com). Changes v1->v2: Large portions of the driver have been reworked from v1 based on remarks from Mark. More specifically: - Remove an unneeded #ifdef and EXPORT_SYMBOL_GPL. - Add ADC3XXX_ prefix to all macro names to avoid namespace contention. - Clean up certain macro names with mixed lower case - upper case names. - Do away with the D0..D7 macros representing bit values. - Fix general formatting: spaces before and after {, }, replace multi spaces with tabs, maximum 2 blank lines anywhere, and other minor issues. - Page register does not need to be volatile. - Rather than use 0 / -6dB enum for input attenuator, use standard TLV. - The dither offset control contains reserved values so use VALUE_ENUM instead of straight ENUM. - Replace vector of SOC_ENUMs with individual variables. - Fix ALSA control names: - Use Volume at the end of all gain controls, and remove "Gain" where applicable. - Use "Switch" at the end of all switch controls, either adding it or changing "switch" into "Switch" as applicable. - Switches and Volumes related to capture should be set up as such, so add "Capture" to widget names where appropriate. This is mostly important for the PGA Capture and Switch widgets, not just for consistency, but also because amixer simple Switch controls expect different names (cap/nocap instead of on/off) for control values for capture. - Make widget names have capital first letters for all consitutent words. - When resuming, the registers will not have changed, so no need to call snd_soc_component_cache_sync(). - Move reset gpio init to I2C probe - Remove needless _priv at end of private struct name. - Make hex constants lower case - _CBM_CFM/_CBS_CFS -> _CBP_CFP/_CBC_CFC - Instead of using device specific ALSA controls to control the two GPIO pins, use GPIOLIB. GPIO pin functions still can be set up in the devicetree as the pins can be used for other functions which are not GPIO related. - Limitation: The GPIO pins can only be used as outputs, even though the hardware alternatively supports using them as inputs. - Handle GPIO configs in a vector rather than separately. - Use more appropriate module_i2c_driver instead of generic module_init/module_exit. - Remove useless initialization data. - As the ADC mute function had no control, add it. - Clean up GPIO initialization which was done both in the I2C probe and in the component probe. - Remove probe, remove, suspend and resume callbacks, which either do not do anything at all anymore, or uselessly just do set_bias_level. - The PLL should not be configured in the devicetree, so move the corresponding definitions to tlv320adc3xxx.h where they can be picked up by a machine driver if needed. - Default to PLL AUTO mode. - Use "reset-gpios" instead of too-generic "gpios" for dt property for configuring codec reset pin. - Use function for printing PLL mode rather than macro. - dev_info is too verbose for most of the printouts, change them to dev_dbg. - Add support for configuring micbias voltage in DT. - Remove ALSA control for mic bias. - Move PLL and NADC, MADC divisor enable to DAPM, including 10 ms post PLL start up delay via POST_PMU event. - ADC is powered on using DAPM, so needless to do it in set_bias_level, thus remove it. - Let DAPM handle BCLK divisor enable. With this, we can remove set_bias_level as it doesn't do anything anymore. - Fix incorrect argument order to a snd_soc_component_update_bits call. - Although the chip itself can operate using BCLK as the clock source, either directly to the divider chain or via the PLL, this is not yet supported in the driver, so require the clocks property for the MCLK for now. - Move PLL rate structure out of .h file where it does not belong. /Ricard Ricard Wanderlof (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 +++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1071 +++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 379 ++++++ 6 files changed, 1624 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 0/2] ASoC: rt9120: Fix settings and add rt9120s compatibility
by cy_huang 09 Nov '21

09 Nov '21

From: ChiYuan Huang <cy_huang(a)richtek.com> This patch series is to fix settings and add rt9120s compatibility ChiYuan Huang (2): ASoC: rt9120: Update internal setting and fix auto sync problem ASoC: rt9120: Add the compatibility with rt9120s sound/soc/codecs/rt9120.c | 59 +++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 49 insertions(+), 10 deletions(-) -- 2.7.4

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[PATCH v2] dt-bindings: sound: gtm601: convert to YAML
by David Heidelberg 09 Nov '21

09 Nov '21

Convert GTM601 binding to the YAML format. Signed-off-by: David Heidelberg <david(a)ixit.cz> --- v2: - add #sound-dai-cells - put kernel(a)puri.sm into maintainers .../devicetree/bindings/sound/gtm601.txt | 19 ---------- .../devicetree/bindings/sound/gtm601.yaml | 36 +++++++++++++++++++ 2 files changed, 36 insertions(+), 19 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/gtm601.txt create mode 100644 Documentation/devicetree/bindings/sound/gtm601.yaml diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt deleted file mode 100644 index efa32a486c4a..000000000000 --- a/Documentation/devicetree/bindings/sound/gtm601.txt +++ /dev/null @@ -1,19 +0,0 @@ -GTM601 UMTS modem audio interface CODEC - -This device has no configuration interface. The sample rate and channels are -based on the compatible string - "option,gtm601" = 8kHz mono - "broadmobi,bm818" = 48KHz stereo - -Required properties: - - - compatible : one of - "option,gtm601" - "broadmobi,bm818" - - -Example: - -codec: gtm601_codec { - compatible = "option,gtm601"; -}; diff --git a/Documentation/devicetree/bindings/sound/gtm601.yaml b/Documentation/devicetree/bindings/sound/gtm601.yaml new file mode 100644 index 000000000000..e81a6aa75522 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/gtm601.yaml @@ -0,0 +1,36 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/gtm601.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: GTM601 UMTS modem audio interface CODEC + +maintainers: + - kernel(a)puri.sm + +description: > + This device has no configuration interface. The sample rate and channels are + based on the compatible string + +properties: + compatible: + oneOf: + - items: + - const: broadmobi,bm818 + - const: option,gtm601 + - items: + - enum: + - broadmobi,bm818 # 48 kHz stereo + - option,gtm601 # 8 kHz mono + + '#sound-dai-cells': + const: 0 + +additionalProperties: false + +examples: + - | + gtm601_codec { + compatible = "option,gtm601"; + }; -- 2.33.0

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