August 2022 - Alsa-devel - mailman.alsa-project.org

snd_pcm_sw_params_set_xfer_align() is deprecated
by GitHub issues - opened 11 Aug '22

11 Aug '22

alsa-project/alsa-oss issue #2 was opened from davidebeatrici: ```c pcm.c: In function 'oss_dsp_sw_params': pcm.c:376:17: warning: 'snd_pcm_sw_params_set_xfer_align' is deprecated [-Wdeprecated-declarations] 376 | snd_pcm_sw_params_set_xfer_align(pcm, sw, 1); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ In file included from /usr/include/alsa/asoundlib.h:54, from pcm.c:40: /usr/include/alsa/pcm.h:1368:5: note: declared here 1368 | int snd_pcm_sw_params_set_xfer_align(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val) __attribute__((deprecated)); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ``` Issue URL : https://github.com/alsa-project/alsa-oss/issues/2 Repository URL: https://github.com/alsa-project/alsa-oss

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[PATCH v8 2/2] ASoC: sun50i-dmic: dt-bindings: add DT bindings for DMIC controller
by Ban Tao 11 Aug '22

11 Aug '22

DT binding documentation for this new ASoC driver. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Reviewed-by: Maxime Ripard <maxime(a)cerno.tech> --- v1->v2: 1.Fix some build errors. v2->v3: 1.Fix some build errors. v3->v4: 1.None. v4->v5: 1.Add interrupt. 2.Keep clock and reset index. v5->v6: 1.None. v6->v7: 1.None. v7->v8: 1.Fix some build errors. --- .../bindings/sound/allwinner,sun50i-h6-dmic.yaml | 79 ++++++++++++++++++++++ 1 file changed, 79 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 0000000..0cfc07f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC Device Tree Bindings + +maintainers: + - Ban Tao <fengzheng923(a)gmail.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun50i-h6-dmic + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + #include <dt-bindings/clock/sun50i-h6-ccu.h> + #include <dt-bindings/reset/sun50i-h6-ccu.h> + + dmic: dmic@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu RST_BUS_DMIC>; + }; + +... -- 2.7.4

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snd_pcm_hw_params_current() always returning zero for periods and buffer_time
by GitHub issues - opened 11 Aug '22

11 Aug '22

alsa-project/alsa-lib issue #258 was opened from ossilator: as can be reproduced rather easily using pyalsaaudio's pcm.dumpinfo(), snd_pcm_hw_params_current() seems to always return zero for periods and buffer_time, at least for playback streams (i didn't test capture). the sequence is basically this: ``` snd_pcm_hw_params_any(...); ... unsigned periods = 4; snd_pcm_hw_params_set_periods_near(..., &periods, ...); // periods is still fine here snd_pcm_hw_params(...); snd_pcm_hw_params_current(...); snd_pcm_hw_params_get_periods(..., &periods, ...); // nope, zero periods ``` i tested this with two completely different setups on two different machines (one a raw driver, another one with pipewire), so it doesn't appear related to the driver. from the other returned parameters it can be inferred that the period count is actually fine. one thing i noticed is that snd_pcm_hw_params_current() uses snd_interval_copy() instead of snd_interval_set_value() for exactly these two parameters. but min & max are both zero. i wasn't able to trace it down so far; both the library and the kernel code look fine. Issue URL : https://github.com/alsa-project/alsa-lib/issues/258 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ALSA: hda/cirrus - support for iMac 12,1 model
by Allen Ballway 11 Aug '22

11 Aug '22

The 12,1 model requires the same configuration as the 12,2 model to enable headphones but has a different codec SSID. Adds 12,1 SSID for matching quirk. Signed-off-by: Allen Ballway <ballway(a)chromium.org> --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 678fbcaf2a3b..e1055d7d9be6 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -396,6 +396,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { /* codec SSID */ SND_PCI_QUIRK(0x106b, 0x0600, "iMac 14,1", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), + SND_PCI_QUIRK(0x106b, 0x0900, "iMac 12,1", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81), -- 2.37.1.559.g78731f0fdb-goog

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[PATCH] ALSA: hda: Fix crash due to jack poll in suspend
by Mohan Kumar 11 Aug '22

11 Aug '22

With jackpoll_in_suspend flag set, there is a possibility that jack poll worker thread will run even after system suspend was completed. Any register access after system pm callback flow will result in kernel crash as still jack poll worker thread tries to access registers. To fix the crash issue during system flow, cancel the jack poll worker thread during system pm prepare callback and cancel the worker thread at start of runtime suspend callback and re-schedule at last to avoid any unwarranted access of register by worker thread during suspend flow. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> --- sound/pci/hda/hda_codec.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7b2e62fa82d5..384426d7e9dd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2940,8 +2940,7 @@ static int hda_codec_runtime_suspend(struct device *dev) if (!codec->card) return 0; - if (!codec->bus->jackpoll_in_suspend) - cancel_delayed_work_sync(&codec->jackpoll_work); + cancel_delayed_work_sync(&codec->jackpoll_work); state = hda_call_codec_suspend(codec); if (codec->link_down_at_suspend || @@ -2949,6 +2948,11 @@ static int hda_codec_runtime_suspend(struct device *dev) (state & AC_PWRST_CLK_STOP_OK))) snd_hdac_codec_link_down(&codec->core); snd_hda_codec_display_power(codec, false); + + if (codec->bus->jackpoll_in_suspend && + (dev->power.power_state.event != PM_EVENT_SUSPEND)) + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); return 0; } @@ -2972,6 +2976,9 @@ static int hda_codec_runtime_resume(struct device *dev) #ifdef CONFIG_PM_SLEEP static int hda_codec_pm_prepare(struct device *dev) { + struct hda_codec *codec = dev_to_hda_codec(dev); + + cancel_delayed_work_sync(&codec->jackpoll_work); dev->power.power_state = PMSG_SUSPEND; return pm_runtime_suspended(dev); } @@ -2991,9 +2998,6 @@ static void hda_codec_pm_complete(struct device *dev) static int hda_codec_pm_suspend(struct device *dev) { - struct hda_codec *codec = dev_to_hda_codec(dev); - - cancel_delayed_work_sync(&codec->jackpoll_work); dev->power.power_state = PMSG_SUSPEND; return pm_runtime_force_suspend(dev); } -- 2.17.1

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[PATCH] ASoC: Intel: kbl_rt5663_max98927: Simplify clk_get() usage
by Christophe JAILLET 10 Aug '22

10 Aug '22

If clk_get() returns -ENOENT, there is no need to defer the driver, -ENOENT will be returned the same for each retries. So, return the error code directly instead of -EPROBE_DEFER. Remove this special case and use dev_err_probe() to simplify code. It will also be less verbose if the clk is really deferred. Fixes: f7f61e08fe58 ("ASoC: Intel: kbl: Enable mclk and ssp sclk early") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- This is based on my understanding of clk_get(). Review with care. Not sure the Fixes tag is needed. The patch does not fix anything. If devm_clk_get() returns -ENOENT, it will just loop several time until the framework gives up. If it returns -EPROBE_DEFER, this case is already handled by the "return ret;" So this patch should be a no-op, just a clean-up. --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 31 ++++---------------- 1 file changed, 6 insertions(+), 25 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 2d4224c5b152..07b00af2fa3c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -989,7 +989,6 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_rt5663_private *ctx; struct snd_soc_acpi_mach *mach; - int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -1009,32 +1008,14 @@ static int kabylake_audio_probe(struct platform_device *pdev) &constraints_dmic_2ch : &constraints_dmic_channels; ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); - return ret; - } + if (IS_ERR(ctx->mclk)) + return dev_err_probe(&pdev->dev, PTR_ERR(ctx->mclk), + "Failed to get ssp1_mclk\n"); ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); - return ret; - } + if (IS_ERR(ctx->sclk)) + return dev_err_probe(&pdev->dev, PTR_ERR(ctx->sclk), + "Failed to get ssp1_sclk\n"); return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); } -- 2.34.1

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[PATCH] ASoC: dt-bindings: fsl,sai: Convert format to json-schema
by Shengjiu Wang 10 Aug '22

10 Aug '22

Convert the NXP SAI binding to DT schema format using json-schema. The Synchronous Audio Interface (SAI) provides an interface that supports full-duplex serial interfaces with frame synchronization formats such as I2S, AC97, TDM, and codec/DSP interfaces. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- .../devicetree/bindings/sound/fsl,sai.yaml | 175 ++++++++++++++++++ .../devicetree/bindings/sound/fsl-sai.txt | 95 ---------- 2 files changed, 175 insertions(+), 95 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,sai.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..adcd77531eba --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,175 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Audio Interface (SAI). + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The SAI is based on I2S module that used communicating with audio codecs, + which provides a synchronous audio interface that supports fullduplex + serial interfaces with frame synchronization such as I2S, AC97, TDM, and + codec/DSP interfaces. + +properties: + compatible: + minItems: 1 + maxItems: 2 + items: + enum: + - fsl,vf610-sai + - fsl,imx6sx-sai + - fsl,imx6ul-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - fsl,imx8ulp-sai + + reg: + maxItems: 1 + + interrupts: + items: + - description: receive and transmit interrupt + + dmas: + minItems: 2 + maxItems: 2 + description: + Must contain a list of pairs of references to DMA specifiers, one for + transmission, and one for reception. + + dma-names: + minItems: 2 + maxItems: 2 + items: + enum: + - tx + - rx + + clocks: + minItems: 4 + items: + - description: The ipg clock for register access + - description: master clock source 0 (obsoleted, compatible for old dts) + - description: master clock source 1 + - description: master clock source 2 + - description: master clock source 3 + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + + clock-names: + minItems: 4 + maxItems: 7 + items: + enum: + - bus + - mclk0 + - mclk1 + - mclk2 + - mclk3 + - pll8k + - pll11k + + lsb-first: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Configures whether the LSB or the MSB is transmitted + first for the fifo data. If this property is absent, + the MSB is transmitted first as default, or the LSB + is transmitted first. + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Boolean property, required if all the SAI + registers are big-endian rather than little-endian. + + fsl,sai-synchronous-rx: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, indicating + that SAI will work in the synchronous mode (sync Tx + with Rx) which means both the transmitter and the + receiver will send and receive data by following + receiver's bit clocks and frame sync clocks. + fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. + + fsl,sai-asynchronous: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, indicating + that SAI will work in the asynchronous mode, which + means both transmitter and receiver will send and + receive data by following their own bit clocks and + frame sync clocks separately. + If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transmitter and receiver will send and receive data by following clocks + of transmitter. + fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + description: | + configure the dataline. it has 3 value for each configuration + first one means the type: I2S(1) or PDM(2) + second one is dataline mask for 'rx' + third one is dataline mask for 'tx'. + for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; + it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type + rx mask is 0xff, tx mask is 0x11 (dataline 1 and 5 enabled). + + fsl,sai-mclk-direction-output: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, + indicates that SAI will output the SAI MCLK clock. + + fsl,shared-interrupt: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, + indicates that interrupt is shared with other modules. + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/vf610-clock.h> + sai2: sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + interrupts = <86 IRQ_TYPE_LEVEL_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "rx", "tx"; + dmas = <&edma0 0 20>, + <&edma0 0 21>; + big-endian; + lsb-first; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt deleted file mode 100644 index fbdefc3fade7..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ /dev/null @@ -1,95 +0,0 @@ -Freescale Synchronous Audio Interface (SAI). - -The SAI is based on I2S module that used communicating with audio codecs, -which provides a synchronous audio interface that supports fullduplex -serial interfaces with frame synchronization such as I2S, AC97, TDM, and -codec/DSP interfaces. - -Required properties: - - - compatible : Compatible list, contains "fsl,vf610-sai", - "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai", - "fsl,imx8qm-sai", "fsl,imx8mm-sai", - "fsl,imx8mn-sai", "fsl,imx8mp-sai", or - "fsl,imx8ulp-sai". - - - reg : Offset and length of the register set for the device. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "bus" for register access and - "mclk1", "mclk2", "mclk3" for bit clock and frame - clock providing. - "pll8k", "pll11k" are optional, they are the clock - source for root clock, one is for 8kHz series rates - another one is for 11kHz series rates. - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - pinctrl-names : Must contain a "default" entry. - - - pinctrl-NNN : One property must exist for each entry in - pinctrl-names. See ../pinctrl/pinctrl-bindings.txt - for details of the property values. - - - lsb-first : Configures whether the LSB or the MSB is transmitted - first for the fifo data. If this property is absent, - the MSB is transmitted first as default, or the LSB - is transmitted first. - - - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating - that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transmitter and the - receiver will send and receive data by following - receiver's bit clocks and frame sync clocks. - - - fsl,sai-asynchronous: This is a boolean property. If present, indicating - that SAI will work in the asynchronous mode, which - means both transmitter and receiver will send and - receive data by following their own bit clocks and - frame sync clocks separately. - - - fsl,dataline : configure the dataline. it has 3 value for each configuration - first one means the type: I2S(1) or PDM(2) - second one is dataline mask for 'rx' - third one is dataline mask for 'tx'. - for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; - it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type - rx mask is 0xff, tx mask is 0x11 (dataline 1 and 4 enabled). - -Optional properties: - - - big-endian : Boolean property, required if all the SAI - registers are big-endian rather than little-endian. - -Optional properties (for mx6ul): - - - fsl,sai-mclk-direction-output: This is a boolean property. If present, - indicates that SAI will output the SAI MCLK clock. - -Note: -- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the - default synchronous mode (sync Rx with Tx) will be used, which means both - transmitter and receiver will send and receive data by following clocks - of transmitter. -- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. - -Example: -sai2: sai@40031000 { - compatible = "fsl,vf610-sai"; - reg = <0x40031000 0x1000>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_PLATFORM_BUS>, - <&clks VF610_CLK_SAI2>, - <&clks 0>, <&clks 0>; - clock-names = "bus", "mclk1", "mclk2", "mclk3"; - dma-names = "tx", "rx"; - dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, - <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; - big-endian; - lsb-first; -}; -- 2.34.1

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Re: [PATCH] ASoC: codec: tlv320aic32x4: fix mono playback via I2S
by Mark Brown 10 Aug '22

10 Aug '22

On Wed, 10 Aug 2022 12:41:56 +0200, Philipp Zabel wrote: > The two commits referenced below break mono playback via I2S DAI because > they set BCLK to half the required speed. For PCM transport over I2S, the > number of transmitted channels is always 2, even for mono playback. > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: codec: tlv320aic32x4: fix mono playback via I2S commit: b4b5f29a076e52181f63e45a2ad1bc88593072e3 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

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ALSA: Setting FIFO type as the slave for dsnoop
by GitHub issues - opened 10 Aug '22

10 Aug '22

alsa-project/alsa-lib issue #257 was opened from cnkoder: **Problem** I am facing problems in using "dsnoop" to share/divide the audio stream into two. When dsnoop plugins' slave is set to be a FIFO it throws an error. executed: sudo arecord -D default -f cd defRecording.wav -c 1 -r 32000 error: ALSA lib pcm_direct.c:1809:(_snd_pcm_direct_get_slave_ipc_offset) Invalid type 'fifo' for slave PCM arecord: main:828: audio open error: Invalid argument This is the current asound.conf settings asound.conf   pcm.!default { type asym playback.pcm "playback" capture.pcm "capture1 " } pcm.playback { type plug slave.pcm "eci" } # Stream Output 1: Final pcm.capture1 { type plug slave.pcm "array" } # Stream Output 2: Final pcm.capture2 { type plug slave.pcm "array" } # Used to share the record device pcm.array { type dsnoop slave { pcm "eco" channels 1 } ipc_key 666666 } # Writes audio coming from any sort of player to ec.input, this is read by the echo # cancellation software. pcm.eci { type plug slave { format S16_LE rate 32000 channels 1 pcm { type file slave.pcm null file "/tmp/ec.input" format "raw" } } } # Read FIFO output which contains echo cancelled audio pcm.eco { type plug slave.pcm { type fifo infile "/tmp/ec.output" rate 32000 format S16_LE channels 1 } #ipc_key 666666 }  Note: eco is used to read the FIFO file which contains the echo canceled audio coming in from cancellation software. This software's input is hw:0 and records audio directly from the microphone, and then processes and passes this over to ec.output Dsnoop works well when the slave.pcm is a hardware device but as soon as I point to something else it fails. Is there a workaround or any other solution to tackle this problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/257 Repository URL: https://github.com/alsa-project/alsa-lib

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Re: [PATCH v2] ASoC: amd: acp: Add TDM support for acp i2s stream
by Venkata Prasad Potturu 10 Aug '22

10 Aug '22

On 8/7/22 12:35, kernel test robot wrote: > Hi Venkata, > > Thank you for the patch! Yet something to improve: > > [auto build test ERROR on broonie-sound/for-next] > [also build test ERROR on linus/master next-20220805] > [cannot apply to v5.19] > [If your patch is applied to the wrong git tree, kindly drop us a note. > And when submitting patch, we suggest to use '--base' as documented in > https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgit-scm.c…] > > url:https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgithu… > base:https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgit.… for-next > config: x86_64-allmodconfig (https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fdownload.…) > compiler: gcc-11 (Debian 11.3.0-3) 11.3.0 > reproduce (this is a W=1 build): > #https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgithub.com%2Fintel-lab-lkp%2Flinux%2Fcommit%2Fc603f9570621784501d7ce7f5099be37983cb810&data=05%7C01%7Cvenkataprasad.potturu%40amd.com%7C06397243ad964e430c5a08da784359bd%7C3dd8961fe4884e608e11a82d994e183d%7C0%7C0%7C637954527920715155%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=QaOUzq7KDkNDmDtTbawTnpB46weD9fXj8rnZmxUC9kI%3D&reserved=0 > git remote add linux-reviewhttps://nam11.safelinks.protection.outlook.com/?url=https%3A%2F… > git fetch --no-tags linux-review Venkata-Prasad-Potturu/ASoC-amd-acp-Add-TDM-support-for-acp-i2s-stream/20220805-140248 > git checkout c603f9570621784501d7ce7f5099be37983cb810 > # save the config file > mkdir build_dir && cp config build_dir/.config > make W=1 O=build_dir ARCH=x86_64 SHELL=/bin/bash sound/soc/ > > If you fix the issue, kindly add following tag where applicable > Reported-by: kernel test robot<lkp(a)intel.com> > > All errors (new ones prefixed by >>): > > sound/soc/amd/acp/acp-i2s.c: In function 'acp_i2s_set_tdm_slot': >>> sound/soc/amd/acp/acp-i2s.c:74:29: error: 'struct acp_dev_data' has no member named 'acp_lock' > 74 | spin_lock_irq(&adata->acp_lock); > | ^~ > In file included from include/linux/kernel.h:21, > from arch/x86/include/asm/percpu.h:27, > from arch/x86/include/asm/current.h:6, > from include/linux/sched.h:12, > from include/linux/ratelimit.h:6, > from include/linux/dev_printk.h:16, > from include/linux/device.h:15, > from include/linux/platform_device.h:13, > from sound/soc/amd/acp/acp-i2s.c:15: > sound/soc/amd/acp/acp-i2s.c:75:43: error: 'struct acp_dev_data' has no member named 'stream_list' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~ > include/linux/container_of.h:18:33: note: in definition of macro 'container_of' > 18 | void *__mptr = (void *)(ptr); \ > | ^~~ > include/linux/list.h:531:9: note: in expansion of macro 'list_entry' > 531 | list_entry((ptr)->next, type, member) > | ^~~~~~~~~~ > include/linux/list.h:674:20: note: in expansion of macro 'list_first_entry' > 674 | for (pos = list_first_entry(head, typeof(*pos), member); \ > | ^~~~~~~~~~~~~~~~ > sound/soc/amd/acp/acp-i2s.c:75:9: note: in expansion of macro 'list_for_each_entry' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~~~~~~~~~~~~~~~~~~ > In file included from include/linux/bits.h:22, > from include/linux/ratelimit_types.h:5, > from include/linux/ratelimit.h:5, > from include/linux/dev_printk.h:16, > from include/linux/device.h:15, > from include/linux/platform_device.h:13, > from sound/soc/amd/acp/acp-i2s.c:15: > sound/soc/amd/acp/acp-i2s.c:75:43: error: 'struct acp_dev_data' has no member named 'stream_list' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~ > include/linux/build_bug.h:78:56: note: in definition of macro '__static_assert' > 78 | #define __static_assert(expr, msg, ...) _Static_assert(expr, msg) > | ^~~~ > include/linux/container_of.h:19:9: note: in expansion of macro 'static_assert' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~~~~~~~~~~~~ > include/linux/container_of.h:19:23: note: in expansion of macro '__same_type' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~~~~~~~~~~ > include/linux/list.h:520:9: note: in expansion of macro 'container_of' > 520 | container_of(ptr, type, member) > | ^~~~~~~~~~~~ > include/linux/list.h:531:9: note: in expansion of macro 'list_entry' > 531 | list_entry((ptr)->next, type, member) > | ^~~~~~~~~~ > include/linux/list.h:674:20: note: in expansion of macro 'list_first_entry' > 674 | for (pos = list_first_entry(head, typeof(*pos), member); \ > | ^~~~~~~~~~~~~~~~ > sound/soc/amd/acp/acp-i2s.c:75:9: note: in expansion of macro 'list_for_each_entry' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~~~~~~~~~~~~~~~~~~ > include/linux/container_of.h:19:54: error: 'struct acp_stream' has no member named 'list' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~ > include/linux/build_bug.h:78:56: note: in definition of macro '__static_assert' > 78 | #define __static_assert(expr, msg, ...) _Static_assert(expr, msg) > | ^~~~ > include/linux/container_of.h:19:9: note: in expansion of macro 'static_assert' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~~~~~~~~~~~~ > include/linux/container_of.h:19:23: note: in expansion of macro '__same_type' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~~~~~~~~~~ > include/linux/list.h:520:9: note: in expansion of macro 'container_of' > 520 | container_of(ptr, type, member) > | ^~~~~~~~~~~~ > include/linux/list.h:531:9: note: in expansion of macro 'list_entry' > 531 | list_entry((ptr)->next, type, member) > | ^~~~~~~~~~ > include/linux/list.h:674:20: note: in expansion of macro 'list_first_entry' > 674 | for (pos = list_first_entry(head, typeof(*pos), member); \ > | ^~~~~~~~~~~~~~~~ > sound/soc/amd/acp/acp-i2s.c:75:9: note: in expansion of macro 'list_for_each_entry' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~~~~~~~~~~~~~~~~~~ > sound/soc/amd/acp/acp-i2s.c:75:43: error: 'struct acp_dev_data' has no member named 'stream_list' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~ > include/linux/build_bug.h:78:56: note: in definition of macro '__static_assert' > 78 | #define __static_assert(expr, msg, ...) _Static_assert(expr, msg) > | ^~~~ > include/linux/container_of.h:19:9: note: in expansion of macro 'static_assert' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~~~~~~~~~~~~ > include/linux/container_of.h:20:23: note: in expansion of macro '__same_type' > 20 | __same_type(*(ptr), void), \ > | ^~~~~~~~~~~ > include/linux/list.h:520:9: note: in expansion of macro 'container_of' > 520 | container_of(ptr, type, member) > | ^~~~~~~~~~~~ > include/linux/list.h:531:9: note: in expansion of macro 'list_entry' > 531 | list_entry((ptr)->next, type, member) > | ^~~~~~~~~~ > include/linux/list.h:674:20: note: in expansion of macro 'list_first_entry' This patch is dependent on below patch, we will send these two patches in one series. depends on :- https://patchwork.kernel.org/project/alsa-devel/patch/20220804072556.601396… > > > vim +74 sound/soc/amd/acp/acp-i2s.c > > 47 > 48 static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mask, > 49 int slots, int slot_width) > 50 { > 51 struct device *dev = dai->component->dev; > 52 struct acp_dev_data *adata = snd_soc_dai_get_drvdata(dai); > 53 struct acp_stream *stream; > 54 int slot_len; > 55 > 56 switch (slot_width) { > 57 case SLOT_WIDTH_8: > 58 slot_len = 8; > 59 break; > 60 case SLOT_WIDTH_16: > 61 slot_len = 16; > 62 break; > 63 case SLOT_WIDTH_24: > 64 slot_len = 24; > 65 break; > 66 case SLOT_WIDTH_32: > 67 slot_len = 0; > 68 break; > 69 default: > 70 dev_err(dev, "Unsupported bitdepth %d\n", slot_width); > 71 return -EINVAL; > 72 } > 73 > > 74 spin_lock_irq(&adata->acp_lock); > 75 list_for_each_entry(stream, &adata->stream_list, list) { > 76 if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) > 77 adata->tdm_tx_fmt[stream->dai_id - 1] = > 78 FRM_LEN | (slots << 15) | (slot_len << 18); > 79 else if (rx_mask && stream->dir == SNDRV_PCM_STREAM_CAPTURE) > 80 adata->tdm_rx_fmt[stream->dai_id - 1] = > 81 FRM_LEN | (slots << 15) | (slot_len << 18); > 82 } > 83 spin_unlock_irq(&adata->acp_lock); > 84 return 0; > 85 } > 86 > > --

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