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

[PATCH] ASoC: wcd934x: Fix a incorrect use of kstrndup
by Miaoqian Lin 14 Dec '21

14 Dec '21

In wcd934x_codec_enable_dec(), widget_name is allocated by kstrndup(). However, according to doc: "Note: Use kmemdup_nul() instead if the size is known exactly." So we should use kmemdup_nul() here instead of kstrndup(). It's similar to CVE-2019-12454. Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/codecs/wcd934x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index e63c6b723d76..c6677cfbce59 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -5005,7 +5005,7 @@ static int wcd934x_codec_enable_dec(struct snd_soc_dapm_widget *w, char *dec; u8 hpf_coff_freq; - widget_name = kstrndup(w->name, 15, GFP_KERNEL); + widget_name = kmemdup_nul(w->name, 15, GFP_KERNEL); if (!widget_name) return -ENOMEM; -- 2.17.1

2 2

Re: linux-next: Tree for Dec 13 (SND_AMD_ACP_CONFIG)
by Randy Dunlap 14 Dec '21

14 Dec '21

On 12/13/21 19:02, broonie(a)kernel.org wrote: > Hi all, > > Non-merge commits (relative to Linus' tree): 5960 > 6555 files changed, 277265 insertions(+), 120864 deletions(-) > > ---------------------------------------------------------------------------- > on i386 or x86_64: when # CONFIG_ACPI is not set, so SND_SOC_ACPI is not set: WARNING: unmet direct dependencies detected for SND_AMD_ACP_CONFIG Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_ACPI [=n] Selected by [y]: - SND_SOC_AMD_ACP_COMMON [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && X86 [=y] && PCI [=y] Full randconfig file is attached -- ~Randy

3 5

[PATCH 0/5] ASoC: qcom: apq8016_sbc: Allow routing audio through QDSP6
by Stephan Gerhold 14 Dec '21

14 Dec '21

This series makes it possible to route audio through the combined audio/modem DSP on MSM8916/APQ8016 devices instead of bypassing it using the LPASS drivers. This is necessary to support certain functionality such as voice call audio. See PATCH 4/5 for details. Also, qcom,apq8016-sbc.txt is converted to DT schema by adding it to the existing qcom,sm8250.yaml. The bindings are similar enough that it is easier to share a single schema instead of duplicating everything into multiple ones. Stephan Gerhold (5): ASoC: dt-bindings: qcom: sm8250: Drop redundant MultiMedia routes ASoC: dt-bindings: qcom: sm8250: Document "aux-devs" ASoC: dt-bindings: qcom: apq8016-sbc: Move to qcom,sm8250 DT schema ASoC: dt-bindings: qcom: Document qcom,msm8916-qdsp6-sndcard compatible ASoC: qcom: apq8016_sbc: Allow routing audio through QDSP6 .../bindings/sound/qcom,apq8016-sbc.txt | 96 ------------- .../bindings/sound/qcom,sm8250.yaml | 136 +++++++++++++++++- sound/soc/qcom/apq8016_sbc.c | 134 ++++++++++++++++- 3 files changed, 259 insertions(+), 107 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc.txt -- 2.34.1

4 16

[PATCH] ASoC: SOF: sof-probes: Constify sof_probe_compr_ops
by Rikard Falkeborn 14 Dec '21

14 Dec '21

The only usage of sof_probe_compr_ops is to assign its address to the cops field in the snd_soc_dai_driver struct (in sound/soc/sof/intel/hda-dai.c). Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/sof/sof-probes.c | 2 +- sound/soc/sof/sof-probes.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-probes.c b/sound/soc/sof/sof-probes.c index 5586af9f1a25..c79026cdb8c7 100644 --- a/sound/soc/sof/sof-probes.c +++ b/sound/soc/sof/sof-probes.c @@ -321,7 +321,7 @@ static int sof_probe_compr_pointer(struct snd_compr_stream *cstream, return snd_sof_probe_compr_pointer(sdev, cstream, tstamp, dai); } -struct snd_soc_cdai_ops sof_probe_compr_ops = { +const struct snd_soc_cdai_ops sof_probe_compr_ops = { .startup = sof_probe_compr_startup, .shutdown = sof_probe_compr_shutdown, .set_params = sof_probe_compr_set_params, diff --git a/sound/soc/sof/sof-probes.h b/sound/soc/sof/sof-probes.h index 35e1dd8d9e03..4a1ed2942d28 100644 --- a/sound/soc/sof/sof-probes.h +++ b/sound/soc/sof/sof-probes.h @@ -32,7 +32,7 @@ int sof_ipc_probe_points_add(struct snd_sof_dev *sdev, int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev, unsigned int *buffer_id, size_t num_buffer_id); -extern struct snd_soc_cdai_ops sof_probe_compr_ops; +extern const struct snd_soc_cdai_ops sof_probe_compr_ops; extern const struct snd_compress_ops sof_probe_compressed_ops; #endif -- 2.34.1

2 1

[PATCH v2] ASoC: SOF: Remove pm_runtime_put_autosuspend() for SOF OF device
by Daniel Baluta 14 Dec '21

14 Dec '21

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> In SOF OF device, pm_runtime_put_autosuspend() is not matching any pm_runtime_get_sync(). This is imbalanced for PM runtime. Also, for consistency we call pm_runtime_mark_last_busy() before enabling PM runtime. 1. Remove pm_runtime_put_autosuspend() in probe_complete 2. Reorder PM runtime calls int probe_complete Signed-off-by: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Changes since v1: - Added my own signed-off-by sound/soc/sof/sof-of-dev.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index 412cbb824b84..b0089698eecb 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -74,11 +74,9 @@ static void sof_of_probe_complete(struct device *dev) /* allow runtime_pm */ pm_runtime_set_autosuspend_delay(dev, SND_SOF_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(dev); + pm_runtime_mark_last_busy(dev); pm_runtime_set_active(dev); pm_runtime_enable(dev); - - pm_runtime_mark_last_busy(dev); - pm_runtime_put_autosuspend(dev); } static int sof_of_probe(struct platform_device *pdev) -- 2.27.0

2 1

[PATCH] ASoC: mediatek: assign correct type to argument
by Jiaxin Yu 14 Dec '21

14 Dec '21

Fix the following sparse warning: (new ones prefixed by >>) >> sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c:370:33: sparse: sparse: incorrect type in argument 3 (different base types) sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c:370:33: sparse: expected unsigned int to sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c:370:33: sparse: got restricted snd_pcm_format_t [usertype] Correct discription of format, use S32_LE and S24_LE to distinguish the different 32bit. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> Reported-by: kernel test robot <lkp(a)intel.com> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 8 ++++---- .../mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 12 ++++++------ .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 4 ++-- .../mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c | 4 ++-- .../mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 4 ++-- 5 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index a4d26a6fc849..f8a72a5102ad 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -155,9 +155,9 @@ static const struct snd_soc_ops mt8183_da7219_rt1015_i2s_ops = { static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S32_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), - 0, SNDRV_PCM_FORMAT_LAST); + 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); params_set_format(params, SNDRV_PCM_FORMAT_S32_LE); @@ -167,9 +167,9 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int mt8183_rt1015_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), - 0, SNDRV_PCM_FORMAT_LAST); + 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index aeb1af86047e..d5fc86132b49 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -94,11 +94,11 @@ static const struct snd_soc_ops mt8183_mt6358_rt1015_i2s_ops = { static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - dev_dbg(rtd->dev, "%s(), fix format to 32bit\n", __func__); + dev_dbg(rtd->dev, "%s(), fix format to S32_LE\n", __func__); - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S32_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), - 0, SNDRV_PCM_FORMAT_LAST); + 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); params_set_format(params, SNDRV_PCM_FORMAT_S32_LE); return 0; @@ -107,11 +107,11 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int mt8183_rt1015_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - dev_dbg(rtd->dev, "%s(), fix format to 32bit\n", __func__); + dev_dbg(rtd->dev, "%s(), fix format to S24_LE\n", __func__); - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), - 0, SNDRV_PCM_FORMAT_LAST); + 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); return 0; diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index a606133951b7..1d16939f80e3 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -350,9 +350,9 @@ static int mt8192_mt6359_hdmi_init(struct snd_soc_pcm_runtime *rtd) static int mt8192_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), - 0, SNDRV_PCM_FORMAT_LAST); + 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c index cca1c739e690..5cdbfaafd479 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c @@ -359,7 +359,7 @@ static int mt8195_rt5682_init(struct snd_soc_pcm_runtime *rtd) static int mt8195_etdm_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); @@ -464,7 +464,7 @@ static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index 11a185da0d96..fa50a31e9718 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -355,7 +355,7 @@ static int mt8195_rt5682_init(struct snd_soc_pcm_runtime *rtd) static int mt8195_etdm_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); @@ -463,7 +463,7 @@ static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { - /* fix BE i2s format to 32bit, clean param mask first */ + /* fix BE i2s format to S24_LE, clean param mask first */ snd_mask_reset_range(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), 0, (__force unsigned int)SNDRV_PCM_FORMAT_LAST); -- 2.25.1

4 3

Re: [PATCH] ALSA: hda/realtek: Add new alc285-hp-amp-init model
by Takashi Iwai 14 Dec '21

14 Dec '21

On Mon, 13 Dec 2021 17:22:47 +0100, Bradley Scott wrote: > > Adds a new "alc285-hp-amp-init" model that can be used to apply the ALC285 > HP speaker amplifier initialization fixup to devices that are not already > known by passing "hda_model=alc285-hp-amp-init" to the > snd-sof-intel-hda-common module or "model=alc285-hp-amp-init" to the > snd-hda-intel module, depending on which is being used. > > Signed-off-by: Bradley Scott <bscott(a)teksavvy.com> Thanks, applied now. Takashi

1 0

[PATCH v2] ALSA: hda/realtek: Amp init fixup for HP ZBook 15 G6
by Bradley Scott 14 Dec '21

14 Dec '21

HP ZBook 15 G6 (SSID 103c:860f) needs the same speaker amplifier initialization as used on several other HP laptops using ALC285. Signed-off-by: Bradley Scott <Bradley.Scott(a)zebra.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 3599f4c85ebf..d162662fe684 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8660,6 +8660,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), + SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), -- 2.25.1

2 1

[PATCH v4] ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema
by Jayesh Choudhary 14 Dec '21

14 Dec '21

Convert the bindings for McASP controllers for TI SOCs from txt to YAML schema. Adds additional properties 'clocks', 'clock-names', 'power-domains', '#sound-dai-cells' and 'port' which were missing from txt file. Removes properties 'sram-size-playback' and 'sram-size-capture' since they are not used. Adds 'dmas' and 'dma-names' in the example which were missing from the txt file. Changes 'interrupts' and 'interrupt-names' from optional to required properties. Changes 'op-mode', 'serial-dir' and 'tdm-slots' to optional properties as they are not needed if the McASP is used only as GPIO. Adds the yaml file in the 'MAINTAINERS' under the heading 'TEXAS INSTRUMENTS ASoC DRIVERS' Signed-off-by: Jayesh Choudhary <j-choudhary(a)ti.com> --- Changelog: v4: - changes the commit message - adds type and description to relevant properties - changes maxItems for 'serial-dir' - removes properties 'sram-size-playback' and 'sram-size-capture' as they are not used - removes 'function-gpios' - removes 'num-serializer' - marks 'tdm-slots', 'serial-dir' and 'op-mode' as optional properties - adds the yaml file in MAINTAINERS v3: - removes maxItems from 'clock-names' v2: - changes the commit message - modifies the properties 'clocks', 'clock-names', 'dma-names', 'dmas', 'interrupts' and 'interrupt-names' according to the arm SOCs - adds 'ports' and 'num-serializer' as node properties .../bindings/sound/davinci-mcasp-audio.txt | 86 -------- .../bindings/sound/davinci-mcasp-audio.yaml | 185 ++++++++++++++++++ MAINTAINERS | 1 + 3 files changed, 186 insertions(+), 86 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt deleted file mode 100644 index bd863bd69501..000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ /dev/null @@ -1,86 +0,0 @@ -Texas Instruments McASP controller - -Required properties: -- compatible : - "ti,dm646x-mcasp-audio" : for DM646x platforms - "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - "ti,dra7-mcasp-audio" : for DRA7xx platforms - "ti,omap4-mcasp-audio" : for OMAP4 - -- reg : Should contain reg specifiers for the entries in the reg-names property. -- reg-names : Should contain: - * "mpu" for the main registers (required). For compatibility with - existing software, it is recommended this is the first entry. - * "dat" for separate data port register access (optional). -- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, - IEC60958-1, and AES-3 formats. -- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted - or received over one serializer. -- serial-dir : A list of serializer configuration. Each entry is a number - indication for serializer pin direction. - (0 - INACTIVE, 1 - TX, 2 - RX) -- dmas: two element list of DMA controller phandles and DMA request line - ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. The dma - identifiers must be "rx" and "tx". - -Optional properties: - -- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0 -- tx-num-evt : FIFO levels. -- rx-num-evt : FIFO levels. -- dismod : Specify the drive on TX pin during inactive slots - 0 : 3-state - 2 : logic low - 3 : logic high - Defaults to 'logic low' when the property is not present -- sram-size-playback : size of sram to be allocated during playback -- sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP -- interrupt-names : Known interrupt names are "tx" and "rx" -- pinctrl-0: Should specify pin control group used for this controller. -- pinctrl-names: Should contain only one value - "default", for more details - please refer to pinctrl-bindings.txt -- fck_parent : Should contain a valid clock name which will be used as parent - for the McASP fck -- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK - and FS rate if applicable: - AUCLK rate = auxclk-fs-ratio * FS rate - -Optional GPIO support: -If any McASP pin need to be used as GPIO then the McASP node must have: -... - gpio-controller - #gpio-cells = <2>; -... - -When requesting a GPIO, the first parameter is the PIN index in McASP_P* -registers. -For example to request the AXR2 pin of mcasp8: -function-gpios = <&mcasp8 2 0>; - -Or to request the ACLKR pin of mcasp8: -function-gpios = <&mcasp8 29 0>; - -For generic gpio information, please refer to bindings/gpio/gpio.txt - -Example: - -mcasp0: mcasp0@1d00000 { - compatible = "ti,da830-mcasp-audio"; - reg = <0x100000 0x3000>; - reg-names "mpu"; - interrupts = <82>, <83>; - interrupt-names = "tx", "rx"; - op-mode = <0>; /* MCASP_IIS_MODE */ - tdm-slots = <2>; - serial-dir = < - 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ - 0 0 0 0 - 0 0 0 1 - 2 0 0 0 >; - tx-num-evt = <1>; - rx-num-evt = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml new file mode 100644 index 000000000000..5b553c90771a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,185 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McASP Controller for TI SoCs + +maintainers: + - Jayesh Choudhary <j-choudhary(a)ti.com> + +properties: + compatible: + enum: + - ti,dm646x-mcasp-audio + - ti,da830-mcasp-audio + - ti,am33xx-mcasp-audio + - ti,dra7-mcasp-audio + - ti,omap4-mcasp-audio + + reg: + minItems: 1 + items: + - description: main registers + - description: data port register + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + op-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + description: I2S - 0 or DIT - 1 operation mode + enum: + - 0 + - 1 + + tdm-slots: + $ref: /schemas/types.yaml#/definitions/uint32 + description: number of channels over one serializer + maxItems: 1 + + serial-dir: + description: + A list of serializer configuration + Entry is indication for serializer pin direction + 0 - Inactive, 1 - TX, 2 - RX + All AXR pins should be present in the array even if inactive + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 25 + items: + minimum: 0 + maximum: 2 + default: 0 + + dmas: + minItems: 1 + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + minItems: 1 + items: + - const: tx + - const: rx + + ti,hwmods: + $ref: /schemas/types.yaml#/definitions/string + description: Name of hwmod associated with McASP + maxItems: 1 + deprecated: true + + tx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: configures WFIFO threshold + maxItems: 1 + + rx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: configures RFIFO threshold + maxItems: 1 + + dismod: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + specify the drive on TX pin during inactive time slots + 0 - 3-state, 2 - logic low, 3 - logic high + enum: + - 0 + - 2 + - 3 + default: 2 + + interrupts: + anyOf: + - minItems: 1 + items: + - description: TX FIFO interrupt + - description: RX FIFO interrupt + - items: + - description: common FIFO interrupt + + interrupt-names: + oneOf: + - minItems: 1 + items: + - const: tx + - const: rx + - const: common + + fck_parent: + $ref: /schemas/types.yaml#/definitions/string + description: parent clock name for McASP fck + maxItems: 1 + + auxclk-fs-ratio: + $ref: /schemas/types.yaml#/definitions/uint32 + description: ratio of AUCLK and FS rate if applicable + maxItems: 1 + + gpio-controller: true + + "#gpio-cells": + const: 2 + + clocks: + minItems: 1 + items: + - description: functional clock + - description: module specific optional ahclkx clock + - description: module specific optional ahclkr clock + + clock-names: + minItems: 1 + items: + - const: fck + - const: ahclkx + - const: ahclkr + + power-domains: + description: phandle to the corresponding power-domain + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + port: + description: connection for when McASP is used via graph card + type: object + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - interrupts + - interrupt-names + +additionalProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index 00ad0cb5cb05..3fdf1e23f7d7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -18855,6 +18855,7 @@ TEXAS INSTRUMENTS ASoC DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained +F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ TEXAS INSTRUMENTS' DAC7612 DAC DRIVER -- 2.17.1

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Does alsa-lib call this kernel api?
by GitHub issues - opened 14 Dec '21

14 Dec '21

alsa-project/alsa-lib issue #204 was opened from Nick0412: Where in the alsa library does it reference the alsa kernel api: https://www.kernel.org/doc/html/latest/sound/kernel-api/alsa-driver-api.htm… Do these two apis interact at all? Issue URL : https://github.com/alsa-project/alsa-lib/issues/204 Repository URL: https://github.com/alsa-project/alsa-lib

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