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

Question about buildroot vs aplay/arecord with -d
by Kuninori Morimoto 09 Oct '20

09 Oct '20

Hi ALSA ML I'm now using buildroot, and want to use aplay/arecord in the same time with -d option. This works for me aplay -d 3 xxx And this works for me, too arecord -d 3 yyy But, this doesn't work, and system will hung up aplay -d 3 xxx & arecord -d 3 yyy Sometimes it will indicates this message rcu: INFO: rcu_preempt detected stalls on CPUs/tasks: rcu: 0-....: (1 GPs behind) idle=496/0/0x1 softirq=228/228 fqs=2172 (detected by 1, t=6502 jiffies, g=-103, q=17) Task dump for CPU 0: task:swapper/0 state:R running task stack: 0 pid: 0 ppid: 0 flags:0x0000002a Call trace: __switch_to+0x144/0x1a0 init_task+0x0/0x1780 ... Do you know why this happen ? Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH][next] ASoC: mchp-spdifrx: fix spelling mistake "overrrun" -> "overrun"
by Colin King 09 Oct '20

09 Oct '20

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_warn message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/atmel/mchp-spdifrx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/mchp-spdifrx.c b/sound/soc/atmel/mchp-spdifrx.c index 726e4951d9a5..e6ded6f8453f 100644 --- a/sound/soc/atmel/mchp-spdifrx.c +++ b/sound/soc/atmel/mchp-spdifrx.c @@ -338,7 +338,7 @@ static irqreturn_t mchp_spdif_interrupt(int irq, void *dev_id) } if (pending & SPDIFRX_IR_OVERRUN) { - dev_warn(dev->dev, "Overrrun detected\n"); + dev_warn(dev->dev, "Overrun detected\n"); ret = IRQ_HANDLED; } -- 2.27.0

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[PATCH 0/2] ASoC: stm32: dfsdm: change rate limits
by Olivier Moysan 09 Oct '20

09 Oct '20

Widening of the supported rate range in the STM32 DFSDM driver. The rates were previously limited to 8kHz, 16kHz and 32kHz. Allow rate capture in the whole range 8kHz-48kHz as there is no hardware limitation to support it. Actual sample resolution is dependent on audio rate and DFSDM configuration. Add a trace to allow simple check of sample resolution. Olivier Moysan (2): ASoC: stm32: dfsdm: change rate limits ASoC: stm32: dfsdm: add actual resolution trace drivers/iio/adc/stm32-dfsdm-adc.c | 4 ++++ drivers/iio/adc/stm32-dfsdm.h | 2 ++ sound/soc/stm/stm32_adfsdm.c | 8 +++----- 3 files changed, 9 insertions(+), 5 deletions(-) -- 2.17.1

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[PATCH v2 1/2] dt-bindings: tas2764: Add the TAS2764 binding doc
by Dan Murphy 09 Oct '20

09 Oct '20

Add the binding for the TAS2764 Smart Amplifier. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2764.yaml | 74 +++++++++++++++++++ 1 file changed, 74 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2764.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2764.yaml b/Documentation/devicetree/bindings/sound/tas2764.yaml new file mode 100644 index 000000000000..d2c90c82b13f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2764.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2764.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2764 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2764 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. + +properties: + compatible: + enum: + - ti,tas2764 + + reg: + maxItems: 1 + description: | + I2C address of the device can be between 0x38 to 0x45. + + reset-gpios: + description: GPIO used to reset the device. + + shutdown-gpios: + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@38 { + compatible = "ti,tas2764"; + reg = <0x38>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpio = <&gpio1 15 0>; + shutdown-gpios = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + +... -- 2.28.0.585.ge1cfff676549

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[PATCH v3 1/2] dt-bindings: tas2764: Add the TAS2764 binding doc
by Dan Murphy 09 Oct '20

09 Oct '20

Add the binding for the TAS2764 Smart Amplifier. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2764.yaml | 76 +++++++++++++++++++ 1 file changed, 76 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2764.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2764.yaml b/Documentation/devicetree/bindings/sound/tas2764.yaml new file mode 100644 index 000000000000..5bf8c76ecda1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2764.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2764.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2764 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2764 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. + +properties: + compatible: + enum: + - ti,tas2764 + + reg: + maxItems: 1 + description: | + I2C address of the device can be between 0x38 to 0x45. + + reset-gpios: + maxItems: 1 + description: GPIO used to reset the device. + + shutdown-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@38 { + compatible = "ti,tas2764"; + reg = <0x38>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpios = <&gpio1 15 0>; + shutdown-gpios = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + +... -- 2.28.0.585.ge1cfff676549

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[PATCH 1/2] ASoC: Intel: catpt: Fix compilation when CONFIG_MODULES is disabled
by Cezary Rojewski 09 Oct '20

09 Oct '20

module_is_live() is available only when CONFIG_MODULES is enabled. Replace its usage with try_module_get() which is present regardless of said config's status. Fixes: 7a10b66a5df9 ("ASoC: Intel: catpt: Device driver lifecycle") Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/catpt/device.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/catpt/device.c b/sound/soc/intel/catpt/device.c index 390ffb203de0..a70179959795 100644 --- a/sound/soc/intel/catpt/device.c +++ b/sound/soc/intel/catpt/device.c @@ -81,10 +81,11 @@ static int __maybe_unused catpt_resume(struct device *dev) if (ret) return ret; - if (!module_is_live(dev->driver->owner)) { + if (!try_module_get(dev->driver->owner)) { dev_info(dev, "module unloading, skipping fw boot\n"); return 0; } + module_put(dev->driver->owner); ret = catpt_boot_firmware(cdev, true); if (ret) { @@ -107,10 +108,12 @@ static int __maybe_unused catpt_resume(struct device *dev) static int __maybe_unused catpt_runtime_suspend(struct device *dev) { - if (!module_is_live(dev->driver->owner)) { + if (!try_module_get(dev->driver->owner)) { dev_info(dev, "module unloading, skipping suspend\n"); return 0; } + module_put(dev->driver->owner); + return catpt_suspend(dev); } -- 2.17.1

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[PATCH] ASoC: ti: davinci-mcasp: Use &pdev->dev for early dev_warn
by Peter Ujfalusi 09 Oct '20

09 Oct '20

At this point mcasp->dev is not initialized and we would have NULL pointer dereference if we would have failed to get the mem memory resource by name. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 3ffdd0f6292a..a6b72ad53b43 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2155,7 +2155,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (!mem) { - dev_warn(mcasp->dev, + dev_warn(&pdev->dev, "\"mpu\" mem resource not found, using index 0\n"); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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(Optional?) DMA vs. PIO
by Andy Shevchenko 08 Oct '20

08 Oct '20

Hi! During internal review of one patch I have been puzzled with the following code and Pierre suggested to ask mailing list for help. My main concern is what was the idea behind? Does it mean we support optional DMA in such case? If now, why not to return an error code directly? ---8<---8<---8<--- > Why ASoC core has the following code in the first place: > > 387 chan = dma_request_chan(dev, name); > 388 if (IS_ERR(chan)) { > 389 if (PTR_ERR(chan) == -EPROBE_DEFER) > 390 return -EPROBE_DEFER; > 391 pcm->chan[i] = NULL; > 392 } else { > 393 pcm->chan[i] = chan; > 394 } > > (note lines 389-391). > If PIO fallback is not okay, why not to return an error there? no idea, the code has been this way since 2013 (5eda87b890f867b098e5566b5543642851e8b9c3) It's worth asking the question on the mailing list, I don't know if this is a bug or a feature. -- With Best Regards, Andy Shevchenko

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[PATCH] ALSA: hda/realtek - The front Mic on a HP machine doesn't work
by Jeremy Szu 08 Oct '20

08 Oct '20

On a HP ZCentral, the front Mic could not be detected. The codec of the HP ZCentrol is alc671 and it needs to override the pin configuration to enable the headset mic. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d4f17b465892..add79ed08bdb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9622,6 +9622,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), -- 2.28.0

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[PATCH 4/6] ALSA: hdspm: Fix typo arbitary
by Naoki Hayama 08 Oct '20

08 Oct '20

Fix comment typo. s/arbitary/arbitrary/ Signed-off-by: Naoki Hayama <naoki.hayama(a)lineo.co.jp> --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 572350aaf18d..a808b445afd8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1217,7 +1217,7 @@ static int snd_hdspm_use_is_exclusive(struct hdspm *hdspm) return ret; } -/* round arbitary sample rates to commonly known rates */ +/* round arbitrary sample rates to commonly known rates */ static int hdspm_round_frequency(int rate) { if (rate < 38050) -- 2.17.1

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