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

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 26 Aug '22

26 Aug '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard: ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` My Motherboard is MSI X570S and output audio works fine but: - Motherboard microphone plug is not work. - Frontal microphone plug is not work. - When change output from gnome settings to hdmi audio does not work, the audio continues to play in the plug and not in the monitor. - When I change the input or output from digital to analog or sinput it does not change at all. - The strangest thing is that the microphone input captures the audio that is heard in the system as if it were the output, audacious records the audio from the desktop when it tries to monitor the microphone. - When plugging or unplugging the microphone, the system detects this change and tries to change the audio input automatically, but it does not capture the microphone and the desktop continues to be heard. - When plug headphone in frontal plug the gnome setting change automaticaly to the headphone item, but the sound play in speakers, not in headphone. - On Audacity when capture Alsa + `USB Audio: #2 (hw: 1,2)` works fine!, but can not set on gnome settings. My alsa data: http://alsa-project.org/db/?f=df4d1f6c126260eabc5f159e64cff37efa064a38 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MPG X570S EDGE MAX WIFI compatibility
by GitHub issues - edited 26 Aug '22

26 Aug '22

alsa-project/alsa-ucm-conf issue #172 was edited from yhojann-cl: Please, add `0db0:151f` to `realtek-alc4080` chip compatibility in MPG X570S EDGE MAX WIFI motherboard: ``` $ lsusb | grep 0db0 Bus 001 Device 003: ID 0db0:151f Micro Star International USB Audio ``` In `ucm2/USB-Audio/USB-Audio.conf` add `151f` to regex of `0db0`: ``` Regex "USB((0b05:(1996|1a2[07]))|(0db0:(1feb|419c|a073|151f)))" ``` My Motherboard is MSI X570S and output audio works fine but: - Motherboard microphone plug is not work. - Frontal microphone plug is not work. - When change output from gnome settings to hdmi audio does not work, the audio continues to play in the plug and not in the monitor. - When I change the input or output from digital to analog or sinput it does not change at all. - The strangest thing is that the microphone input captures the audio that is heard in the system as if it were the output, audacious records the audio from the desktop when it tries to monitor the microphone. - When plugging or unplugging the microphone, the system detects this change and tries to change the audio input automatically, but it does not capture the microphone and the desktop continues to be heard. - When plug headphone in frontal plug the gnome setting change automaticaly to the headphone item, but the sound play in speakers, not in headphone. - On Audacity when capture Alsa + "USB Audio: #2 (hw: 1,2)" works fine!, but can not set on gnome settings. My alsa data: http://alsa-project.org/db/?f=df4d1f6c126260eabc5f159e64cff37efa064a38 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/172 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: Fix misspelling of 'system' as 'ststem'.
by Maurycy Z 26 Aug '22

26 Aug '22

Signed-off-by: Maurycy Zalewski <10maurycy10(a)gmail.com> --- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/soc/fsl/Kconfig | 2 +- sound/soc/generic/Kconfig | 4 ++-- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index a5241a287..ff6ce91bf 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2247,7 +2247,7 @@ static int snd_ice1712_read_eeprom(struct snd_ice1712 *ice, (snd_ice1712_read_i2c(ice, dev, 0x03) << 24); if (ice->eeprom.subvendor == 0 || ice->eeprom.subvendor == (unsigned int)-1) { - /* invalid subvendor from EEPROM, try the PCI subststem ID instead */ + /* invalid subvendor from EEPROM, try the PCI subsystem ID instead */ u16 vendor, device; pci_read_config_word(ice->pci, PCI_SUBSYSTEM_VENDOR_ID, &vendor); pci_read_config_word(ice->pci, PCI_SUBSYSTEM_ID, &device); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 6fab2ad85..9e81e6335 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2266,7 +2266,7 @@ static int snd_vt1724_read_eeprom(struct snd_ice1712 *ice, if (ice->eeprom.subvendor == 0 || ice->eeprom.subvendor == (unsigned int)-1) { /* invalid subvendor from EEPROM, try the PCI - * subststem ID instead + * subsystem ID instead */ u16 vendor, device; pci_read_config_word(ice->pci, PCI_SUBSYSTEM_VENDOR_ID, diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 614eceda6..c7fe12fbd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -366,7 +366,7 @@ config SND_SOC_IMX_CARD help This option enables audio sound card support for i.MX boards with OF-graph DT bindings. - It also support DPCM of single CPU multi Codec ststem. + It also support DPCM of single CPU multi Codec system. endif # SND_IMX_SOC diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index b6df4e26b..955ac8d3f 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -7,7 +7,7 @@ config SND_SIMPLE_CARD select SND_SIMPLE_CARD_UTILS help This option enables generic simple sound card support - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD tristate "ASoC Audio Graph sound card support" @@ -16,7 +16,7 @@ config SND_AUDIO_GRAPH_CARD help This option enables generic simple sound card support with OF-graph DT bindings. - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD2 tristate "ASoC Audio Graph sound card2 support" -- 2.37.2

2 1

[PATCH] ALSA: control: Re-order bounds checking in get_ctl_id_hash()
by Dan Carpenter 26 Aug '22

26 Aug '22

These two checks are in the reverse order so it might read one element beyond the end of the array. First check if the "i" is within bounds before using it. Fixes: 6ab55ec0a938 ("ALSA: control: Fix an out-of-bounds bug in get_ctl_id_hash()") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index e8fc4c511e5f..a7271927d875 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -391,7 +391,7 @@ static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) h = id->iface; h = MULTIPLIER * h + id->device; h = MULTIPLIER * h + id->subdevice; - for (i = 0; id->name[i] && i < SNDRV_CTL_ELEM_ID_NAME_MAXLEN; i++) + for (i = 0; i < SNDRV_CTL_ELEM_ID_NAME_MAXLEN && id->name[i]; i++) h = MULTIPLIER * h + id->name[i]; h = MULTIPLIER * h + id->index; h &= LONG_MAX; -- 2.35.1

2 1

Add interactive pause/resume functionality
by GitHub pull_request - edited 26 Aug '22

26 Aug '22

alsa-project/tinycompress pull request #14 was edited from LaurentiuM1234: We want to be able to interactively pause and resume the stream. Initially, when trying to play an MP3 song, cplay would hang during the polling on compress_write. The cause for this was the fact that, after calling `get_codec_mp3`, the file pointer would point to the first MP3 data, leaving out the first MP3 header. I'm assuming that this would cause the codec to be unable to process the received frames since it wouldn't be able to find the first MP3 header. In order to be able to use the pause/resume functionality on MP3 streams, this problem had to be addressed before adding the actual functionality. Request URL : https://github.com/alsa-project/tinycompress/pull/14 Patch URL : https://github.com/alsa-project/tinycompress/pull/14.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

Add interactive pause/resume functionality
by GitHub pull_request - edited 26 Aug '22

26 Aug '22

alsa-project/tinycompress pull request #14 was edited from LaurentiuM1234: We want to be able to interactively pause and resume the stream. Initially, when trying to play an MP3 song, cplay would hang during the polling on compress_write. The cause for this was probably the fact that, after calling `get_codec_mp3`, the file pointer would point to the first MP3 data, leaving out the first MP3 header. I'm assuming that this would cause the codec to hang since it wouldn't be able to find the first MP3 header. In order to be able to use the pause/resume functionality on MP3 streams, this problem had to be addressed before adding the actual functionality. Request URL : https://github.com/alsa-project/tinycompress/pull/14 Patch URL : https://github.com/alsa-project/tinycompress/pull/14.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

Add interactive pause/resume functionality
by GitHub pull_request - edited 26 Aug '22

26 Aug '22

alsa-project/tinycompress pull request #14 was edited from LaurentiuM1234: We want to be able to interactively pause and resume the stream Request URL : https://github.com/alsa-project/tinycompress/pull/14 Patch URL : https://github.com/alsa-project/tinycompress/pull/14.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH 1/2] sound: realtek: Add pincfg for ASUS G533Z
by Luke D. Jones 26 Aug '22

26 Aug '22

Fixes up the pincfg for ASUS ROG Strix G15 (G533Z) laptop. Signed-off-by: Luke D. Jones <luke(a)ljones.dev> --- sound/pci/hda/patch_realtek.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fd630d62b5a0..e9cad49ee2a5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5828,7 +5828,7 @@ static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, struct alc_spec *spec = codec->spec; spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; alc255_set_default_jack_type(codec); - } + } else alc_fixup_headset_mode(codec, fix, action); } @@ -7022,6 +7022,7 @@ enum { ALC294_FIXUP_ASUS_GX502_HP, ALC294_FIXUP_ASUS_GX502_PINS, ALC294_FIXUP_ASUS_GX502_VERBS, + ALC285_FIXUP_ASUS_G533Z_PINS, ALC294_FIXUP_ASUS_GU502_HP, ALC294_FIXUP_ASUS_GU502_PINS, ALC294_FIXUP_ASUS_GU502_VERBS, @@ -8363,6 +8364,17 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc294_fixup_gu502_hp, }, + [ALC285_FIXUP_ASUS_G533Z_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170120 }, + { 0x19, 0x03A11050 }, /* front HP mic */ + { 0x1B, 0x03A11C30 }, /* rear external mic */ + { 0x21, 0x03211420 }, /* front HP out */ + { } + }, + .chained = false, + }, [ALC294_FIXUP_ASUS_COEF_1B] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -9294,6 +9306,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), + SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE), -- 2.37.2

2 2

[PATCH v8 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 26 Aug '22

26 Aug '22

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.Fix some compilation errors. 2.Modify some code styles. v2->v3: None. v3->v4: 1.add sig_bits. v4->v5: None. v5->v6: 1.Modify RXFIFO_CTL_MODE to mode 1. v6->v7: 1.Modify dmic_rate_s to be a global variable. 2.Changed some macro names to make more sense. 3.Align code format. 4.Add a depends on PM to Kconfig entry. v7->v8: None. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index e9d5b05..839f625 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -796,6 +796,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa9..8543234 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && PM && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be34..4483fe9 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 0000000..76b3378 --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMPLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMPLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_MASK BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_LSB (0 << 9) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_MSB (1 << 9) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK BIT(8) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_16 (0 << 8) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_24 (1 << 8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static struct dmic_rate dmic_rate_s[] = { + {48000, 0x0}, + {44100, 0x0}, + {32000, 0x1}, + {24000, 0x2}, + {22050, 0x2}, + {16000, 0x3}, + {12000, 0x4}, + {11025, 0x4}, + {8000, 0x5}, +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + unsigned int chan_en = (1 << channels) - 1; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, chan_en); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(chan_en)); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_16); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_24); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE_MASK, + SUN50I_DMIC_RXFIFO_CTL_MODE_MSB); + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMPLE_RATE_MASK, + SUN50I_DMIC_SR_SAMPLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_DMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_DMIC_FORMATS, + .sig_bits = 21, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + + ret = clk_prepare_enable(host->bus_clk); + if (ret) { + clk_disable_unprepare(host->dmic_clk); + return ret; + } + + return 0; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, &sun50i_dmic_component, + &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_disable_runtime_pm; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_disable_runtime_pm: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.7.4

3 3

[PATCH] ASoC: wm_adsp: Silent parsing error on loading speaker protection fw
by Cristian Ciocaltea 26 Aug '22

26 Aug '22

The tracing capabilities for the speaker protection fw enabled via commit c55b3e46cb99 ("ASoC: wm_adsp: Add trace caps to speaker protection FW") are not be available on all platforms, such as the Valve's Steam Deck which is based on the Halo Core DSP. As a consequence, whenever the firmware is loaded, a rather misleading 'Failed to parse legacy: -19' error message is written to the kernel ring buffer: [ 288.977412] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Firmware version: 3 [ 288.978002] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: cs35l41-dsp1-spk-prot.wmfw: Fri 02 Apr 2021 21:03:50 W. Europe Daylight Time [ 289.094065] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Firmware: 400a4 vendor: 0x2 v0.33.0, 2 algorithms [ 289.095073] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: 0: ID cd v29.53.0 XM@94 YM@e [ 289.095665] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: 1: ID f20b v0.0.1 XM@170 YM@0 [ 289.096275] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Protection: C:\Users\ocanavan\Desktop\cirrusTune_july2021.bin [ 291.172383] steamdeck kernel: cs35l41 spi-VLV1776:01: DSP1: Failed to parse legacy: -19 Update wm_adsp_buffer_init() to *not* report the ENODEV error when the firmware type is WM_ADSP_FW_SPK_PROT. Fixes: c55b3e46cb99 ("ASoC: wm_adsp: Add trace caps to speaker protection FW") Signed-off-by: Cristian Ciocaltea <cristian.ciocaltea(a)collabora.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index cfaa45ede916..7514fc03b468 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1602,7 +1602,7 @@ static int wm_adsp_buffer_init(struct wm_adsp *dsp) if (list_empty(&dsp->buffer_list)) { /* Fall back to legacy support */ ret = wm_adsp_buffer_parse_legacy(dsp); - if (ret) + if (ret && (dsp->fw != WM_ADSP_FW_SPK_PROT || ret != -ENODEV)) adsp_warn(dsp, "Failed to parse legacy: %d\n", ret); } base-commit: 568035b01cfb107af8d2e4bd2fb9aea22cf5b868 -- 2.37.2

2 2

[PATCH v2 1/2] ASoC: dt-bindings: max98396: Document data monitor properties
by Daniel Mack 25 Aug '22

25 Aug '22

This device features a data monitor that puts the device in software reset upon a configurable set of events. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- v1 -> v2: fix a typo and remove a stray blank line .../bindings/sound/adi,max98396.yaml | 37 +++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml index 8d2ef991db40..f4eefe10c494 100644 --- a/Documentation/devicetree/bindings/sound/adi,max98396.yaml +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -78,6 +78,43 @@ properties: interleaved on a single output channel. type: boolean + adi,dmon-stuck-enable: + description: + Enables the "data monitor stuck" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-stuck-threshold-bits: + description: + Sets the threshold for the "data monitor stuck" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [9, 11, 13, 15] + default: 15 + + adi,dmon-magnitude-enable: + description: + Enables the "data monitor magnitude" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-magnitude-threshold-bits: + description: + Sets the threshold for the "data monitor magnitude" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [2, 3, 4, 5] + default: 5 + + adi,dmon-duration-msecs: + description: + Sets the duration for the "data monitor" feature, in milliseconds. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [64, 256, 1024, 4096] + default: 64 + reset-gpios: maxItems: 1 -- 2.37.2

2 2

[PATCH 1/2] lib/string_helpers: Introduce strsplit_u32()
by Cezary Rojewski 25 Aug '22

25 Aug '22

Add strsplit_u32() and its __user variant to allow for splitting specified string into array of u32 tokens. Originally this functionality was added for the SOF sound driver. As more users are on the horizon, relocate it so it becomes a common good. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/linux/string_helpers.h | 3 + lib/string_helpers.c | 96 +++++++++++++++++++++++++++++++ sound/soc/sof/sof-client-probes.c | 51 +--------------- 3 files changed, 100 insertions(+), 50 deletions(-) diff --git a/include/linux/string_helpers.h b/include/linux/string_helpers.h index 4d72258d42fd..a4630ddfca27 100644 --- a/include/linux/string_helpers.h +++ b/include/linux/string_helpers.h @@ -126,4 +126,7 @@ static inline const char *str_enabled_disabled(bool v) return v ? "enabled" : "disabled"; } +int strsplit_u32(const char *str, const char *delim, u32 **tkns, size_t *num_tkns); +int strsplit_u32_user(const char __user *from, size_t count, loff_t *ppos, const char *delim, + u32 **tkns, size_t *num_tkns); #endif diff --git a/lib/string_helpers.c b/lib/string_helpers.c index 5ed3beb066e6..bb24f0c62539 100644 --- a/lib/string_helpers.c +++ b/lib/string_helpers.c @@ -984,3 +984,99 @@ void fortify_panic(const char *name) } EXPORT_SYMBOL(fortify_panic); #endif /* CONFIG_FORTIFY_SOURCE */ + +/** + * strsplit_u32 - Split string into sequence of u32 tokens + * @str: The string to split into tokens. + * @delim: The string containing delimiter characters. + * @tkns: Returned u32 sequence pointer. + * @num_tkns: Returned number of tokens obtained. + * + * On success @num_tkns and @tkns are assigned the number of tokens extracted + * and the array itself respectively. + * Caller takes responsibility for freeing @tkns when no longer needed. + */ +int strsplit_u32(const char *str, const char *delim, u32 **tkns, size_t *num_tkns) +{ + size_t max_count = 32; + size_t count = 0; + char *s, **p; + u32 *buf, *tmp; + int ret = 0; + + p = (char **)&str; + *tkns = NULL; + *num_tkns = 0; + + buf = kcalloc(max_count, sizeof(*buf), GFP_KERNEL); + if (!buf) + return -ENOMEM; + + while ((s = strsep(p, delim)) != NULL) { + ret = kstrtouint(s, 0, buf + count); + if (ret) + goto free_buf; + + if (++count > max_count) { + max_count *= 2; + tmp = krealloc(buf, max_count * sizeof(*buf), GFP_KERNEL); + if (!tmp) { + ret = -ENOMEM; + goto free_buf; + } + buf = tmp; + } + } + + if (!count) + goto free_buf; + *tkns = kmemdup(buf, count * sizeof(*buf), GFP_KERNEL); + if (*tkns == NULL) { + ret = -ENOMEM; + goto free_buf; + } + *num_tkns = count; + +free_buf: + kfree(buf); + return ret; +} +EXPORT_SYMBOL(strsplit_u32); + +/** + * strsplit_u32_user - Split string into sequence of u32 tokens + * @from: The user space buffer to read from + * @ppos: The current position in the buffer + * @count: The maximum number of bytes to read + * @delim: The string containing delimiter characters. + * @tkns: Returned u32 sequence pointer. + * @num_tkns: Returned number of tokens obtained. + * + * On success @num_tkns and @tkns are assigned the number of tokens extracted + * and the array itself respectively. + * Caller takes responsibility for freeing @tkns when no longer needed. + */ +int strsplit_u32_user(const char __user *from, size_t count, loff_t *ppos, const char *delim, + u32 **tkns, size_t *num_tkns) +{ + char *buf; + int ret; + + buf = kmalloc(count + 1, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + ret = simple_write_to_buffer(buf, count, ppos, from, count); + if (ret != count) { + ret = (ret < 0) ? ret : -EIO; + goto free_buf; + } + + buf[count] = '\0'; + ret = strsplit_u32(buf, delim, tkns, num_tkns); + +free_buf: + kfree(buf); + return ret; +} +EXPORT_SYMBOL(strsplit_u32_user); diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index 1f1ea93a7fbf..48ebbe58e2b9 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -12,6 +12,7 @@ #include <linux/debugfs.h> #include <linux/module.h> #include <linux/pm_runtime.h> +#include <linux/string_helpers.h> #include <sound/soc.h> #include <sound/sof/header.h> #include "sof-client.h" @@ -410,56 +411,6 @@ static const struct snd_compress_ops sof_probes_compressed_ops = { .copy = sof_probes_compr_copy, }; -/** - * strsplit_u32 - Split string into sequence of u32 tokens - * @buf: String to split into tokens. - * @delim: String containing delimiter characters. - * @tkns: Returned u32 sequence pointer. - * @num_tkns: Returned number of tokens obtained. - */ -static int strsplit_u32(char *buf, const char *delim, u32 **tkns, size_t *num_tkns) -{ - char *s; - u32 *data, *tmp; - size_t count = 0; - size_t cap = 32; - int ret = 0; - - *tkns = NULL; - *num_tkns = 0; - data = kcalloc(cap, sizeof(*data), GFP_KERNEL); - if (!data) - return -ENOMEM; - - while ((s = strsep(&buf, delim)) != NULL) { - ret = kstrtouint(s, 0, data + count); - if (ret) - goto exit; - if (++count >= cap) { - cap *= 2; - tmp = krealloc(data, cap * sizeof(*data), GFP_KERNEL); - if (!tmp) { - ret = -ENOMEM; - goto exit; - } - data = tmp; - } - } - - if (!count) - goto exit; - *tkns = kmemdup(data, count * sizeof(*data), GFP_KERNEL); - if (!(*tkns)) { - ret = -ENOMEM; - goto exit; - } - *num_tkns = count; - -exit: - kfree(data); - return ret; -} - static int tokenize_input(const char __user *from, size_t count, loff_t *ppos, u32 **tkns, size_t *num_tkns) { -- 2.25.1

5 29

[PATCH -next] ASoC: sigmadsp: switch to use kmemdup_nul() helper
by Yang Yingliang 25 Aug '22

25 Aug '22

Use kmemdup_nul() helper instead of open-coding to simplify the code. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/sigmadsp.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index b992216aee55..3047a6fbb380 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -227,13 +227,11 @@ static int sigma_fw_load_control(struct sigmadsp *sigmadsp, if (!ctrl) return -ENOMEM; - name = kzalloc(name_len + 1, GFP_KERNEL); + name = kmemdup_nul(ctrl_chunk->name, name_len, GFP_KERNEL); if (!name) { ret = -ENOMEM; goto err_free_ctrl; } - memcpy(name, ctrl_chunk->name, name_len); - name[name_len] = '\0'; ctrl->name = name; /* -- 2.25.1

2 1

[PATCH v2 0/2] ASoC: mediatek: dt-bindings: modify machine
by Chunxu Li 25 Aug '22

25 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> Changes since V1: - remove unnecessary quotes Add SOF related field. 1. Add a property "mediatek,adsp", Only when adsp phandle could be retrieved, from DTS, the SOF related part of machine driver is executed. 2. Add a property "mediatek,dai-link" to support dai-links could be specified from DTS chunxu.li (2): ASoC: mediatek: dt-bindings: modify machine bindings for SOF ASoC: mediatek: dt-bindings: modify machine bindings for SOF .../bindings/sound/mt8186-mt6366-da7219-max98357.yaml | 10 ++++++++++ .../bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml | 10 ++++++++++ 2 files changed, 20 insertions(+) -- 2.25.1

3 7

[PATCH 1/2] ASoC: fsl_aud2htx: register platform component before registering cpu dai
by Shengjiu Wang 25 Aug '22

25 Aug '22

There is no defer probe when adding platform component to snd_soc_pcm_runtime(rtd), the code is in snd_soc_add_pcm_runtime() snd_soc_register_card() -> snd_soc_bind_card() -> snd_soc_add_pcm_runtime() -> adding cpu dai -> adding codec dai -> adding platform component. So if the platform component is not ready at that time, then the sound card still registered successfully, but platform component is empty, the sound card can't be used. As there is defer probe checking for cpu dai component, then register platform component before cpu dai to avoid such issue. And the behavior of imx_pcm_dma_init() is same as common devm_snd_dmaengine_pcm_register(), so use devm_snd_dmaengine_pcm_register() instead Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_aud2htx.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 873295f59ad7..bc1b9c6df95c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -234,6 +234,16 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) regcache_cache_only(aud2htx->regmap, true); + /* + * Register platform component before registering cpu dai for there + * is not defer probe for platform component in snd_soc_add_pcm_runtime(). + */ + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "failed to pcm register\n"); + return ret; + } + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_aud2htx_component, &fsl_aud2htx_dai, 1); @@ -242,10 +252,6 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) return ret; } - ret = imx_pcm_dma_init(pdev); - if (ret) - dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); - return ret; } -- 2.34.1

2 2

[PATCH 1/2] ASoC: dt-bindings: max98396: Document data monitor properties
by Daniel Mack 25 Aug '22

25 Aug '22

This device features a data monitor that puts the device in software reset upon a configurable set of events. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- .../bindings/sound/adi,max98396.yaml | 37 +++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml index 8d2ef991db40..f4eefe10c494 100644 --- a/Documentation/devicetree/bindings/sound/adi,max98396.yaml +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -78,6 +78,43 @@ properties: interleaved on a single output channel. type: boolean + adi,dmon-stuck-enable: + description: + Enables the "data monitor stuck" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-stuck-threshold-bits: + description: + Sets the threshold for the "data monitor stuck" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [9, 11, 13, 15] + default: 15 + + adi,dmon-magnitude-enable: + description: + Enables the "data monitor magnitude" feature. Once the data monitor is + enabled, it actively monitors the selected input data (from DIN) to the + speaker amplifier. Once a data error is detected, the data monitor + automatically places the device into software shutdown. + type: boolean + + adi,dmon-magnitude-threshold-bits: + description: + Sets the threshold for the "data monitor magnitude" feature, in bits. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [2, 3, 4, 5] + default: 5 + + adi,dmon-duration-msecs: + description: + Sets the duration for the "data monitor" feature, in milliseconds. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [64, 256, 1024, 4096] + default: 64 + reset-gpios: maxItems: 1 -- 2.37.2

1 1

[PATCH] ASoC: dapm: Export new 'graph.dot' file in debugfs
by Martin Povišer 25 Aug '22

25 Aug '22

Provide a DOT summary of the DAPM graph in a newly added 'graph.dot' file in debugfs, placed in the card's DAPM directory. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- Sample output: https://cutebit.org/macaudio-j274.svg (With unupstreamed sound drivers on Mac mini (2020)) The helper bufprintf macro triggers checkpath.pl: ERROR: Macros with complex values should be enclosed in parentheses #47: FILE: sound/soc/soc-dapm.c:2235: +#define bufprintf(...) \ + ret += scnprintf(buf + ret, bufsize - ret, __VA_ARGS__) but adding in {} to the macro body interferes with the if/else constructions later, so I left it as-is. sound/soc/soc-dapm.c | 141 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 141 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 73b8bd452ca7..86524908c3fd 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2210,6 +2210,143 @@ static const struct file_operations dapm_bias_fops = { .llseek = default_llseek, }; +static ssize_t dapm_graph_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + struct snd_soc_card *card = file->private_data; + struct snd_soc_dapm_context *dapm; + struct snd_soc_dapm_path *p; + struct snd_soc_dapm_widget *w; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dapm_widget *wdone[16]; + struct snd_soc_dai *dai; + int i, num_wdone = 0, cluster = 0; + char *buf; + ssize_t bufsize; + ssize_t ret = 0; + + bufsize = 1024 * card->num_dapm_widgets; + buf = kmalloc(bufsize, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + mutex_lock(&card->dapm_mutex); + +#define bufprintf(...) \ + ret += scnprintf(buf + ret, bufsize - ret, __VA_ARGS__) + + bufprintf("digraph dapm {\n"); + bufprintf("label=\"%s\";\n", card->name); + + /* + * Print the user-visible PCM devices of the card. + */ + bufprintf("subgraph cluster_%d {\n", cluster++); + bufprintf("label=\"PCM devices\";style=filled;fillcolor=lightgray;\n"); + for_each_card_rtds(card, rtd) { + if (rtd->dai_link->no_pcm) + continue; + + bufprintf("w%pK [label=\"%d: %s\"];\n", rtd, + rtd->pcm->device, rtd->dai_link->name); + } + bufprintf("};\n"); + + /* + * Print the playback/capture widgets of CPU-side DAIs, and link + * them to the PCM devices. Keep a list of already printed + * widgets in 'wdone', so they will be skipped later. Do not put + * these widgets in a component cluster like we will do with + * the other widgets later, since that just clutters the graph. + */ + for_each_card_rtds(card, rtd) { + for_each_rtd_cpu_dais(rtd, i, dai) { + if (dai->playback_widget) { + w = dai->playback_widget; + bufprintf("w%pK [label=\"%s\"];\n", w, w->name); + if (!rtd->dai_link->no_pcm) + bufprintf("w%pK -> w%pK;\n", rtd, w); + wdone[num_wdone] = w; + if (num_wdone < ARRAY_SIZE(wdone)) + num_wdone++; + } + + if (dai->capture_widget) { + w = dai->capture_widget; + bufprintf("w%pK [label=\"%s\"];\n", w, w->name); + if (!rtd->dai_link->no_pcm) + bufprintf("w%pK -> w%pK;\n", w, rtd); + wdone[num_wdone] = w; + if (num_wdone < ARRAY_SIZE(wdone)) + num_wdone++; + } + } + } + + for_each_card_dapms(card, dapm) { + const char *prefix = soc_dapm_prefix(dapm); + + if (dapm != &card->dapm) { + bufprintf("subgraph cluster_%d {\n", cluster++); + if (prefix && dapm->component) + bufprintf("label=\"%s (%s)\";\n", prefix, + dapm->component->name); + else if (dapm->component) + bufprintf("label=\"%s\";\n", + dapm->component->name); + } + + for_each_card_widgets(dapm->card, w) { + const char *name = w->name; + bool skip = false; + + if (w->dapm != dapm) + continue; + + if (list_empty(&w->edges[0]) && list_empty(&w->edges[1])) + continue; + + for (i = 0; i < num_wdone; i++) + if (wdone[i] == w) + skip = true; + if (skip) + continue; + + if (prefix && strlen(name) > strlen(prefix) + 1) + name += strlen(prefix) + 1; + + bufprintf("w%pK [label=\"%s\"];\n", w, name); + } + + if (dapm != &card->dapm) + bufprintf("}\n"); + } + + list_for_each_entry(p, &card->paths, list) { + if (p->name) + bufprintf("w%pK -> w%pK [label=\"%s\"];\n", + p->source, p->sink, p->name); + else + bufprintf("w%pK -> w%pK;\n", p->source, p->sink); + } + + bufprintf("}\n"); +#undef bufprintf + + mutex_unlock(&card->dapm_mutex); + + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + + kfree(buf); + return ret; +} + +static const struct file_operations dapm_graph_fops = { + .open = simple_open, + .read = dapm_graph_read_file, + .llseek = default_llseek, +}; + void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent) { @@ -2220,6 +2357,10 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, debugfs_create_file("bias_level", 0444, dapm->debugfs_dapm, dapm, &dapm_bias_fops); + + if (dapm == &dapm->card->dapm) + debugfs_create_file("graph.dot", 0444, dapm->debugfs_dapm, + dapm->card, &dapm_graph_fops); } static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) -- 2.33.0

5 10

[PATCH] ASoC: max98088: add support for noise gate reg
by Tommaso Merciai 25 Aug '22

25 Aug '22

Add support for Noise Gate Threshold reg (ANTH reg 0x40 bit 4 to 7) References: - https://datasheets.maximintegrated.com/en/ds/MAX98089.pdf, p75 Signed-off-by: Tommaso Merciai <tommaso.merciai(a)amarulasolutions.com> --- sound/soc/codecs/max98088.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 5ef2e1279ee7..76ef71ab9c83 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -474,6 +474,9 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = { max98088_mic2pre_get, max98088_mic2pre_set, max98088_micboost_tlv), + SOC_SINGLE("Noise Gate Threshold", M98088_REG_40_MICAGC_THRESH, + 4, 15, 0), + SOC_SINGLE("INA Volume", M98088_REG_37_LVL_INA, 0, 7, 1), SOC_SINGLE("INB Volume", M98088_REG_38_LVL_INB, 0, 7, 1), -- 2.25.1

2 1

[PATCH 0/3] ASoC: SOF: mediatek: update SOF driver for mt8186
by Chunxu Li 25 Aug '22

25 Aug '22

In these patches, we update SOF driver for mt8186 Chunxu Li (3): ASoC: SOF: mediatek: Add dai driver for mt8186 ASoC: SOF: mediatek: add snd_sof_dsp_ops callbacks for pcm and mail box ASoC: SOF: mediatek: Use generic implementation for .ipc_msg_data field sound/soc/sof/mediatek/mt8186/mt8186.c | 49 ++++++++++++++++++++++---- 1 file changed, 42 insertions(+), 7 deletions(-) -- 2.25.1

2 4

[PATCH] ASoC: codecs: add suspend and resume for ES8316
by Zhu Ning 25 Aug '22

25 Aug '22

The registers may be lost after suspend due to powerdown. regcache_sync solves this issue. Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/codecs/es8316.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index de7185f73e1e..8643014472ae 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -767,9 +767,31 @@ static void es8316_remove(struct snd_soc_component *component) clk_disable_unprepare(es8316->mclk); } +static int es8316_resume(struct snd_soc_component *component) +{ + struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8316->regmap, false); + regcache_sync(es8316->regmap); + + return 0; +} + +static int es8316_suspend(struct snd_soc_component *component) +{ + struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8316->regmap, true); + regcache_mark_dirty(es8316->regmap); + + return 0; +} + static const struct snd_soc_component_driver soc_component_dev_es8316 = { .probe = es8316_probe, .remove = es8316_remove, + .resume = es8316_resume, + .suspend = es8316_suspend, .set_jack = es8316_set_jack, .controls = es8316_snd_controls, .num_controls = ARRAY_SIZE(es8316_snd_controls), -- 2.36.1

2 1

[PATCH v2 1/6] sysfs: do not create empty directories if no attributes are present
by Greg Kroah-Hartman 25 Aug '22

25 Aug '22

When creating an attribute group, if it is named a subdirectory is created and the sysfs files are placed into that subdirectory. If no files are created, normally the directory would still be present, but it would be empty. Clean this up by removing the directory if no files were successfully created in the group at all. Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Sanyog Kale <sanyog.r.kale(a)intel.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- v2: new patch Note, totally untested! The following soundwire patches will need this, if a soundwire developer could test this out, it would be most apreciated. fs/sysfs/group.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/fs/sysfs/group.c b/fs/sysfs/group.c index eeb0e3099421..9fe0b47db47f 100644 --- a/fs/sysfs/group.c +++ b/fs/sysfs/group.c @@ -31,12 +31,14 @@ static void remove_files(struct kernfs_node *parent, kernfs_remove_by_name(parent, (*bin_attr)->attr.name); } +/* returns -ERROR if error, or >= 0 for number of files actually created */ static int create_files(struct kernfs_node *parent, struct kobject *kobj, kuid_t uid, kgid_t gid, const struct attribute_group *grp, int update) { struct attribute *const *attr; struct bin_attribute *const *bin_attr; + int files_created = 0; int error = 0, i; if (grp->attrs) { @@ -65,6 +67,8 @@ static int create_files(struct kernfs_node *parent, struct kobject *kobj, gid, NULL); if (unlikely(error)) break; + + files_created++; } if (error) { remove_files(parent, grp); @@ -95,12 +99,15 @@ static int create_files(struct kernfs_node *parent, struct kobject *kobj, NULL); if (error) break; + files_created++; } if (error) remove_files(parent, grp); } exit: - return error; + if (error) + return error; + return files_created; } @@ -146,10 +153,16 @@ static int internal_create_group(struct kobject *kobj, int update, kn = kobj->sd; kernfs_get(kn); error = create_files(kn, kobj, uid, gid, grp, update); - if (error) { + if (error <= 0) { + /* + * If an error happened _OR_ if no files were created in the + * attribute group, and we have a name for this group, delete + * the name so there's not an empty directory. + */ if (grp->name) kernfs_remove(kn); - } + } else + error = 0; kernfs_put(kn); if (grp->name && update) -- 2.37.2

2 8

[PATCH AUTOSEL 5.15 02/20] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Sasha Levin 25 Aug '22

25 Aug '22

From: Biju Das <biju.das.jz(a)bp.renesas.com> [ Upstream commit c75ed9f54ce8d349fee557f2b471a4d637ed2a6b ] We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Link: https://lore.kernel.org/r/20220728092612.38858-1-biju.das.jz@bp.renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 6d794eaaf4c3..2e33a1fa0a6f 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1022,32 +1022,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_reset; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_reset: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 34/38] ALSA: hda/cs8409: Support new Dolphin Variants
by Sasha Levin 25 Aug '22

25 Aug '22

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit 1ff954f9ab054675b9eb02dd14add8f7aa376d71 ] Add 4 new Dolphin Systems, same configuration as older systems. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220816151901.1398007-1-sbinding@opensource.cirr… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_cs8409-tables.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index 4f4cc8215917..5b140301ca66 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -546,6 +546,10 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0BD6, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0BD7, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0BD8, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C43, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C50, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C51, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C52, "Dolphin", CS8409_DOLPHIN), {} /* terminator */ }; -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 18/38] ALSA: hda/realtek: Add quirks for ASUS Zenbooks using CS35L41
by Sasha Levin 25 Aug '22

25 Aug '22

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit 461122b999bda2ebef2086a35d8990f9ccac5ab8 ] These Asus Zenbook laptop use Realtek HDA codec combined with 2xCS35L41 Amplifiers using SPI. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220815141953.25197-1-sbinding@opensource.cirrus… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 619e6025ba97..7aaa021f550d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9246,6 +9246,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x12af, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), @@ -9266,6 +9267,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 03/38] ASoC: rt5640: Fix the JD voltage dropping issue
by Sasha Levin 25 Aug '22

25 Aug '22

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit afb176d45870048eea540991b082208270824037 ] The patch fixes the JD voltage dropping issue in the HDA JD using. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Mohan Kumar D <mkumard(a)nvidia.com> Link: https://lore.kernel.org/r/20220808052836.25791-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5640.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 18b3da9211e3..5ada0d318d0f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1986,7 +1986,7 @@ static int rt5640_set_bias_level(struct snd_soc_component *component, snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000); if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) snd_soc_component_write(component, RT5640_PWR_ANLG1, - 0x0018); + 0x2818); else snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000); @@ -2592,7 +2592,8 @@ static void rt5640_enable_hda_jack_detect( snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0); snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, - RT5640_PWR_VREF2, RT5640_PWR_VREF2); + RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG, + RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG); usleep_range(10000, 15000); snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, RT5640_PWR_FV2, RT5640_PWR_FV2); -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 02/38] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Sasha Levin 25 Aug '22

25 Aug '22

From: Biju Das <biju.das.jz(a)bp.renesas.com> [ Upstream commit c75ed9f54ce8d349fee557f2b471a4d637ed2a6b ] We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Link: https://lore.kernel.org/r/20220728092612.38858-1-biju.das.jz@bp.renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index e392de7a262e..3d74acffec11 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1016,32 +1016,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_reset; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_reset: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.35.1

1 0

[PATCH] ARM: pxa: pxa2xx-ac97-lib: fix tests for platform_get_irq() failure
by Yu Zhe 25 Aug '22

25 Aug '22

The platform_get_irq() returns negative error codes. It can't actually return zero. Signed-off-by: Yu Zhe <yuzhe(a)nfschina.com> --- sound/arm/pxa2xx-ac97-lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index e55c0421718b..1f4c4749a6fb 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -402,7 +402,7 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) goto err_clk2; irq = platform_get_irq(dev, 0); - if (!irq) + if (irq < 0) goto err_irq; ret = request_irq(irq, pxa2xx_ac97_irq, 0, "AC97", NULL); -- 2.11.0

1 0

[PATCH 0/2] dt-bindings: sound: Add SOF field for mt8186
by Chunxu Li 24 Aug '22

24 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> These patches add SOF field for mt8186 machine driver chunxu.li (2): ASoC: mediatek: dt-bindings: modify machine bindings for SOF ASoC: mediatek: dt-bindings: modify machine bindings for SOF .../bindings/sound/mt8186-mt6366-da7219-max98357.yaml | 10 ++++++++++ .../bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml | 10 ++++++++++ 2 files changed, 20 insertions(+) -- 2.25.1

3 5

Have I got a missing ucm files problem ?
by GitHub issues - opened 24 Aug '22

24 Aug '22

alsa-project/alsa-ucm-conf issue #204 was opened from xevilstar: Hello, this is my alsa-info http://alsa-project.org/db/?f=cc1f9ff2eb1e223ebf6024aa1605db1d737ff5fa I am having problems with callaudiod not finding the headset and mic on my audio cards and one of the members on the callaudiod git suggested me I am missing some ucm files .... This is a raspberrypi4 can you please help me ? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/204 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 1/5] soundwire: sysfs: move sdw_slave_dev_attr_group into the existing list of groups
by Greg Kroah-Hartman 24 Aug '22

24 Aug '22

The sysfs logic already creates a list of groups for the device, so add the sdw_slave_dev_attr_group group to that list instead of having to do a two-step process of adding a group list and then an individual group. This is a step on the way to moving all of the sysfs attribute handling into the default driver core attribute group logic so that the soundwire core does not have to do any of it manually. Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Sanyog Kale <sanyog.r.kale(a)intel.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- drivers/soundwire/sysfs_slave.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/sysfs_slave.c b/drivers/soundwire/sysfs_slave.c index 3210359cd944..83e3f6cc3250 100644 --- a/drivers/soundwire/sysfs_slave.c +++ b/drivers/soundwire/sysfs_slave.c @@ -105,7 +105,10 @@ static struct attribute *slave_attrs[] = { &dev_attr_modalias.attr, NULL, }; -ATTRIBUTE_GROUPS(slave); + +static const struct attribute_group slave_attr_group = { + .attrs = slave_attrs, +}; static struct attribute *slave_dev_attrs[] = { &dev_attr_mipi_revision.attr, @@ -190,6 +193,12 @@ static const struct attribute_group dp0_group = { .name = "dp0", }; +static const struct attribute_group *slave_groups[] = { + &slave_attr_group, + &sdw_slave_dev_attr_group, + NULL, +}; + int sdw_slave_sysfs_init(struct sdw_slave *slave) { int ret; @@ -198,10 +207,6 @@ int sdw_slave_sysfs_init(struct sdw_slave *slave) if (ret < 0) return ret; - ret = devm_device_add_group(&slave->dev, &sdw_slave_dev_attr_group); - if (ret < 0) - return ret; - if (slave->prop.dp0_prop) { ret = devm_device_add_group(&slave->dev, &dp0_group); if (ret < 0) -- 2.37.1

3 23

[PATCH -next 1/2] ASoC: SOF: imx8ulp: change es8326_regmap_config to static
by Yang Yingliang 24 Aug '22

24 Aug '22

sof_imx8ulp_ops is only used in imx8ulp.c now, change it to static. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sof/imx/imx8ulp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 02b496165acc..afab7feab5fb 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -413,7 +413,7 @@ static int imx8ulp_dsp_set_power_state(struct snd_sof_dev *sdev, } /* i.MX8 ops */ -struct snd_sof_dsp_ops sof_imx8ulp_ops = { +static struct snd_sof_dsp_ops sof_imx8ulp_ops = { /* probe and remove */ .probe = imx8ulp_probe, .remove = imx8ulp_remove, -- 2.25.1

2 2

[PATCH 1/2] ALSA: seq: oss: Fix data-race for max_midi_devs access
by Takashi Iwai 24 Aug '22

24 Aug '22

ALSA OSS sequencer refers to a global variable max_midi_devs at creating a new port, storing it to its own field. Meanwhile this variable may be changed by other sequencer events at snd_seq_oss_midi_check_exit_port() in parallel, which may cause a data race. OTOH, this data race itself is almost harmless, as the access to the MIDI device is done via get_mdev() and it's protected with a refcount, hence its presence is guaranteed. Though, it's sill better to address the data-race from the code sanity POV, and this patch adds the proper spinlock for the protection. Reported-by: Abhishek Shah <abhishek.shah(a)columbia.edu> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/CAEHB2493pZRXs863w58QWnUTtv3HHfg85aYhLn5HJHCwxqtH… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss_midi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 1e3bf086f867..07efb38f58ac 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -270,7 +270,9 @@ snd_seq_oss_midi_clear_all(void) void snd_seq_oss_midi_setup(struct seq_oss_devinfo *dp) { + spin_lock_irq(&register_lock); dp->max_mididev = max_midi_devs; + spin_unlock_irq(&register_lock); } /* -- 2.35.3

2 5

[PATCH 0/4] ASoC: SOF: Intel: override mclk_id for ES8336 support
by Pierre-Louis Bossart 24 Aug '22

24 Aug '22

This patchset solves a known issue with ES8336 platforms wrt MCLK selection. Most of the devices use the MCLK0 signal, but some devices do use the MCLK1 signal. The MCLK is defined in the topology, it would be a nightmare to generate more topology files just for one MCLK difference. With a minor extension to the intel-nhlt library, the MCLK information can be found by parsing the NHLT table, and we can override the mclk_id at boot time. The only known issues for this platform remain the detection of GPIO and microphone connections, currently only possible with manual quirks. Thanks to Eugene J. Markow for testing this patchset. Pierre-Louis Bossart (4): ASoC: SOF: add quirk to override topology mclk_id ALSA: hda: intel-nhlt: add intel_nhlt_ssp_mclk_mask() ASoC: SOF: Intel: hda: override mclk_id after parsing NHLT SSP blob ASoC: SOF: Intel: hda: refine SSP count support include/sound/intel-nhlt.h | 7 ++++ sound/hda/intel-nhlt.c | 61 +++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.c | 34 +++++++++++++++++++ sound/soc/sof/intel/hda.h | 2 ++ sound/soc/sof/intel/mtl.c | 2 +- sound/soc/sof/intel/tgl.c | 8 ++--- sound/soc/sof/ipc3-topology.c | 7 ++++ sound/soc/sof/sof-priv.h | 4 +++ 8 files changed, 120 insertions(+), 5 deletions(-) -- 2.34.1

3 12

[PATCH v2] ASoC: codes: src4xxx: Avoid clang -Wsometimes-uninitialized in src4xxx_hw_params()
by Nathan Chancellor 24 Aug '22

24 Aug '22

Clang warns: sound/soc/codecs/src4xxx.c:280:3: error: variable 'd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:298:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); ^ sound/soc/codecs/src4xxx.c:223:20: note: initialize the variable 'd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'jd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:293:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); ^~ sound/soc/codecs/src4xxx.c:223:17: note: initialize the variable 'jd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'pj' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:288:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); ^~ sound/soc/codecs/src4xxx.c:223:13: note: initialize the variable 'pj' to silence this warning int val, pj, jd, d; ^ = 0 3 errors generated. The datasheet does not have any default values for these regmap values so pick some arbitrary values and print to the user that this is the case to silence the warnings. Link: https://github.com/ClangBuiltLinux/linux/issues/1691 Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: "Sudip Mukherjee (Codethink)" <sudipm.mukherjee(a)gmail.com> Suggested-by: Matt Flax <flatmax(a)flatmax.com> Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- v1 -> v2: https://lore.kernel.org/20220822183101.1115095-1-nathan@kernel.org/ * Don't return early, just initialize the values to some arbitrary numbers and try to hobble along, as other parts of the chip may be functional. * Add message and comment to describe this situation. sound/soc/codecs/src4xxx.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/src4xxx.c b/sound/soc/codecs/src4xxx.c index a8f143057b41..db4e280dd055 100644 --- a/sound/soc/codecs/src4xxx.c +++ b/sound/soc/codecs/src4xxx.c @@ -280,9 +280,14 @@ static int src4xxx_hw_params(struct snd_pcm_substream *substream, default: /* don't error out here, * other parts of the chip are still functional + * Dummy initialize variables to avoid + * -Wsometimes-uninitialized from clang. */ dev_info(component->dev, - "Couldn't set the RCV PLL as this master clock rate is unknown\n"); + "Couldn't set the RCV PLL as this master clock rate is unknown. Chosen regmap values may not match real world values.\n"); + pj = 0x0; + jd = 0xff; + d = 0xff; break; } ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); base-commit: 94f072748337424c9cf92cd018532a34db3a5516 -- 2.37.2

3 3

[PATCH] ASoC: fsl_sai: Update slots number according to bclk_ratio
by Shengjiu Wang 24 Aug '22

24 Aug '22

The bclk_ratio is set by .set_bclk_ratio API. bclk_ratio = slots * slot_width So if slots is not set by .set_tdm_slot, then it can be calculated by bclk_ratio. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7523bb944b21..acf76ea21138 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -535,12 +535,14 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 pins, bclk; int ret, i; - if (sai->slots) - slots = sai->slots; - if (sai->slot_width) slot_width = sai->slot_width; + if (sai->slots) + slots = sai->slots; + else if (sai->bclk_ratio) + slots = sai->bclk_ratio / slot_width; + pins = DIV_ROUND_UP(channels, slots); /* -- 2.17.1

2 1

[PATCH] ASoC: SOF: imx: imx8ulp: declare ops structure as static
by Pierre-Louis Bossart 24 Aug '22

24 Aug '22

Sparse warning: sound/soc/sof/imx/imx8ulp.c:416:24: error: symbol 'sof_imx8ulp_ops' was not declared. Should it be static? Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/imx/imx8ulp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 02b496165acc3..afab7feab5fb3 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -413,7 +413,7 @@ static int imx8ulp_dsp_set_power_state(struct snd_sof_dev *sdev, } /* i.MX8 ops */ -struct snd_sof_dsp_ops sof_imx8ulp_ops = { +static struct snd_sof_dsp_ops sof_imx8ulp_ops = { /* probe and remove */ .probe = imx8ulp_probe, .remove = imx8ulp_remove, -- 2.34.1

2 1

[PATCH] ASoC: dt-bindings: Add missing (unevaluated|additional)Properties on child nodes
by Rob Herring 24 Aug '22

24 Aug '22

In order to ensure only documented properties are present, node schemas must have unevaluatedProperties or additionalProperties set to false (typically). Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../bindings/sound/amlogic,gx-sound-card.yaml | 2 ++ .../bindings/sound/audio-graph-port.yaml | 2 ++ .../bindings/sound/google,sc7180-trogdor.yaml | 4 ++++ .../devicetree/bindings/sound/imx-audio-card.yaml | 2 ++ .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 4 ++++ .../devicetree/bindings/sound/qcom,sm8250.yaml | 6 ++++++ .../devicetree/bindings/sound/renesas,rsnd.yaml | 14 ++++++++++++-- .../bindings/sound/samsung,aries-wm8994.yaml | 2 ++ .../bindings/sound/samsung,midas-audio.yaml | 2 ++ .../devicetree/bindings/sound/samsung,snow.yaml | 2 ++ .../devicetree/bindings/sound/st,stm32-sai.yaml | 1 + 11 files changed, 39 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index b4b35edcb493..5b8d59245f82 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -40,6 +40,7 @@ properties: patternProperties: "^dai-link-[0-9]+$": type: object + additionalProperties: false description: |- dai-link child nodes: Container for dai-link level properties and the CODEC sub-nodes. @@ -63,6 +64,7 @@ patternProperties: patternProperties: "^codec-[0-9]+$": type: object + additionalProperties: false description: |- Codecs: dai-link representing backend links should have at least one subnode. diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 5c368674d11a..c54ca5eb0baa 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -27,6 +27,8 @@ properties: patternProperties: "^endpoint(@[0-9a-f]+)?": $ref: /schemas/graph.yaml#/$defs/endpoint-base + unevaluatedProperties: false + properties: mclk-fs: description: | diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index 233caa0ade87..67ccddd44489 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -61,6 +61,8 @@ patternProperties: cpu: description: Holds subnode which indicates cpu dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 @@ -68,6 +70,8 @@ patternProperties: codec: description: Holds subnode which indicates codec dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml index bb3a435722c7..b6f5d486600e 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -58,6 +58,7 @@ patternProperties: cpu: description: Holds subnode which indicates cpu dai. type: object + additionalProperties: false properties: sound-dai: maxItems: 1 @@ -65,6 +66,7 @@ patternProperties: codec: description: Holds subnode which indicates codec dai. type: object + additionalProperties: false properties: sound-dai: minItems: 1 diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml index 4fa179909c62..478be7e3fa29 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml @@ -30,6 +30,8 @@ properties: headset-codec: type: object + additionalProperties: false + properties: sound-dai: $ref: /schemas/types.yaml#/definitions/phandle @@ -38,6 +40,8 @@ properties: speaker-codecs: type: object + additionalProperties: false + properties: sound-dai: minItems: 1 diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index e6e27d09783e..a3a4289f713e 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -71,6 +71,8 @@ patternProperties: cpu: description: Holds subnode which indicates cpu dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 @@ -78,6 +80,8 @@ patternProperties: platform: description: Holds subnode which indicates platform dai. type: object + additionalProperties: false + properties: sound-dai: maxItems: 1 @@ -85,6 +89,8 @@ patternProperties: codec: description: Holds subnode which indicates codec dai. type: object + additionalProperties: false + properties: sound-dai: minItems: 1 diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index e17c0245f77a..268895c90bd5 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -129,6 +129,8 @@ properties: patternProperties: "^dvc-[0-1]$": type: object + additionalProperties: false + properties: dmas: maxItems: 1 @@ -145,7 +147,7 @@ properties: patternProperties: "^mix-[0-1]$": type: object - # no properties + additionalProperties: false additionalProperties: false rcar_sound,ctu: @@ -154,7 +156,7 @@ properties: patternProperties: "^ctu-[0-7]$": type: object - # no properties + additionalProperties: false additionalProperties: false rcar_sound,src: @@ -163,6 +165,8 @@ properties: patternProperties: "^src-[0-9]$": type: object + additionalProperties: false + properties: interrupts: maxItems: 1 @@ -186,6 +190,8 @@ properties: patternProperties: "^ssiu-[0-9]+$": type: object + additionalProperties: false + properties: dmas: maxItems: 2 @@ -206,6 +212,8 @@ properties: patternProperties: "^ssi-[0-9]$": type: object + additionalProperties: false + properties: interrupts: maxItems: 1 @@ -243,6 +251,8 @@ properties: patternProperties: "^dai([0-9]+)?$": type: object + additionalProperties: false + properties: playback: $ref: /schemas/types.yaml#/definitions/phandle-array diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml index a01c4ad929b8..447e013f6e17 100644 --- a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -23,6 +23,7 @@ properties: cpu: type: object + additionalProperties: false properties: sound-dai: minItems: 2 @@ -34,6 +35,7 @@ properties: - sound-dai codec: + additionalProperties: false type: object properties: sound-dai: diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml index ec50bcb4af5f..31095913e330 100644 --- a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -19,6 +19,7 @@ properties: cpu: type: object + additionalProperties: false properties: sound-dai: maxItems: 1 @@ -28,6 +29,7 @@ properties: codec: type: object + additionalProperties: false properties: sound-dai: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sound/samsung,snow.yaml b/Documentation/devicetree/bindings/sound/samsung,snow.yaml index 51a83d3c7274..3d49aa4c9be2 100644 --- a/Documentation/devicetree/bindings/sound/samsung,snow.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,snow.yaml @@ -19,6 +19,7 @@ properties: codec: type: object + additionalProperties: false properties: sound-dai: description: List of phandles to the CODEC and HDMI IP nodes. @@ -30,6 +31,7 @@ properties: cpu: type: object + additionalProperties: false properties: sound-dai: description: Phandle to the Samsung I2S controller. diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml index fe2e15504ebc..1a3abc949505 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -60,6 +60,7 @@ required: patternProperties: "^audio-controller@[0-9a-f]+$": type: object + additionalProperties: false description: Two subnodes corresponding to SAI sub-block instances A et B can be defined. Subnode can be omitted for unsused sub-block. -- 2.34.1

2 1

[PATCH] ASoC: SOF: Intel: hda: Skip IMR boot after a firmware crash or boot failure
by Peter Ujfalusi 24 Aug '22

24 Aug '22

To make sure that we start from a clean state next time when the DSP is powered up after a firmware crash or boot failure we must skip the IMR booting attempt. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/intel/hda-dsp.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index eddfd77ad90f..671c3e02d7df 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -620,8 +620,13 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) /* * The memory used for IMR boot loses its content in deeper than S3 state * We must not try IMR boot on next power up (as it will fail). + * + * In case of firmware crash or boot failure set the skip_imr_boot to true + * as well in order to try to re-load the firmware to do a 'cold' boot. */ - if (sdev->system_suspend_target > SOF_SUSPEND_S3) + if (sdev->system_suspend_target > SOF_SUSPEND_S3 || + sdev->fw_state == SOF_FW_CRASHED || + sdev->fw_state == SOF_FW_BOOT_FAILED) hda->skip_imr_boot = true; hda_sdw_int_enable(sdev, false); -- 2.37.2

2 1

[PATCH] ASoC: SOF: ipc4-loader: Verify ext manifest magic number
by Peter Ujfalusi 24 Aug '22

24 Aug '22

Firmware image must start with an extended manifest. Add a check to make sure that the image does contain it. The magic number (the first u32 of a firmware image if manifest is present) for an IPC4 image must be 0x31454124 (ASCI "$AE1"). Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/sof/ipc4-loader.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c index 9fadae8fd011..8bd2132b4f41 100644 --- a/sound/soc/sof/ipc4-loader.c +++ b/sound/soc/sof/ipc4-loader.c @@ -40,6 +40,17 @@ static size_t sof_ipc4_fw_parse_ext_man(struct snd_sof_dev *sdev) ext_man_hdr = (struct sof_ext_manifest4_hdr *)fw->data; + /* + * At the start of the firmware image we must have an extended manifest. + * Verify that the magic number is correct. + */ + if (ext_man_hdr->id != SOF_EXT_MAN4_MAGIC_NUMBER) { + dev_err(sdev->dev, + "Unexpected extended manifest magic number: %#x\n", + ext_man_hdr->id); + return -EINVAL; + } + fw_hdr_offset = ipc4_data->manifest_fw_hdr_offset; if (!fw_hdr_offset) return -EINVAL; -- 2.37.2

2 1

[PATCH] ALSA: control: Fix an out-of-bounds bug in get_ctl_id_hash()
by Zheyu Ma 24 Aug '22

24 Aug '22

Since the user can control the arguments provided to the kernel by the ioctl() system call, an out-of-bounds bug occurs when the 'id->name' provided by the user does not end with '\0'. The following log can reveal it: [ 10.002313] BUG: KASAN: stack-out-of-bounds in snd_ctl_find_id+0x36c/0x3a0 [ 10.002895] Read of size 1 at addr ffff888109f5fe28 by task snd/439 [ 10.004934] Call Trace: [ 10.007140] snd_ctl_find_id+0x36c/0x3a0 [ 10.007489] snd_ctl_ioctl+0x6cf/0x10e0 Fix this by checking the bound of 'id->name' in the loop. Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Signed-off-by: Zheyu Ma <zheyuma97(a)gmail.com> --- sound/core/control.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index f3e893715369..e8fc4c511e5f 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -385,14 +385,14 @@ static bool elem_id_matches(const struct snd_kcontrol *kctl, #define MULTIPLIER 37 static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) { + int i; unsigned long h; - const unsigned char *p; h = id->iface; h = MULTIPLIER * h + id->device; h = MULTIPLIER * h + id->subdevice; - for (p = id->name; *p; p++) - h = MULTIPLIER * h + *p; + for (i = 0; id->name[i] && i < SNDRV_CTL_ELEM_ID_NAME_MAXLEN; i++) + h = MULTIPLIER * h + id->name[i]; h = MULTIPLIER * h + id->index; h &= LONG_MAX; return h; -- 2.25.1

2 1

arcam_av.c: Include missing string.h
by GitHub pull_request - opened 24 Aug '22

24 Aug '22

alsa-project/alsa-plugins pull request #47 was opened from kraj: bzero() function needs this header to be included Signed-off-by: Khem Raj <raj.khem(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-plugins/pull/47 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/47.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v2] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 23 Aug '22

23 Aug '22

Starting with version 2.10a of Synopsys DesignWare HDMI controller the insert_pcuv bit was introduced. On RK3399pro SoM (Radxa Rock Pi N10) for example if we neglect to set this bit and proceed to enable hdmi_sound and i2s2 on the device tree there will be extreme clipping of sound output, to the point that music sounds like white noise. Problem could also manifest as just mild cracking depending of HDMI audio implementation of sink. Setting insert_pcuv bit (bit 2 of aud_conf2 Audio Sample register) fixes this. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- v1->v2: SoC->SoM on description, better commenting, minor style changes, conditional application of fix for L-PCM only --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,16 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in + * version 2.10a. + * + * This single bit (bit 2 of HDMI_AUD_CONF2) + * when set to 1 will enable the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) + * bits on the incoming audio stream. + * + * Support is limited to Linear PCM audio. If + * neglected, the lack of valid PCUV bits + * on L-PCM streams will cause anything from + * mild cracking to full blown extreme + * clipping depending of HDMI audio + * implementation of the sink. + * + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +120,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; } --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -931,6 +931,11 @@ enum { HDMI_AUD_CONF1_WIDTH_16 = 0x10, HDMI_AUD_CONF1_WIDTH_24 = 0x18, +/* AUD_CONF2 field values */ + HDMI_AUD_CONF2_HBR = 0x01, + HDMI_AUD_CONF2_NLPCM = 0x02, + HDMI_AUD_CONF2_INSERT_PCUV = 0x04, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0,

1 0

[PATCH 0/3] ASoC: SOF: Intel: hda: Correct Firmware State Register use
by Peter Ujfalusi 23 Aug '22

23 Aug '22

Hi, The FSR (Firmware State Register) holds the ROM state information, it does not contain error information. The FSR itself is a bit more complicated as well as the state depends on the module currently in use. The error code from ROM or the status code from the firmware is located at the next register. Fix the handling of the FSR in order to provide usable and human readable (in most cases) report on the status and error. Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Intel: hda: Correct the ROM/FW state reporting code ASoC: SOF: Intel: hda-loader: Use the FSR state definitions during bootup ASoC: SOF: Intel: hda: Drop no longer used ROM state definitions sound/soc/sof/intel/hda-loader.c | 10 +-- sound/soc/sof/intel/hda.c | 147 ++++++++++++++++++++++++++----- sound/soc/sof/intel/hda.h | 69 +++++++++++++-- 3 files changed, 194 insertions(+), 32 deletions(-) -- 2.37.0

2 5

[PATCH] ASoC: mediatek: mt8186: rename sound card name
by Chunxu Li 23 Aug '22

23 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> The field 'topology_shortname' in 'snd_soc_card' is defined as char[32], Current card name will be truncated when SOF is enabled, so rename the sound card name. Signed-off-by: chunxu.li <chunxu.li(a)mediatek.com> --- sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c index 84ee5d95a9f0..17a15bec41da 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c @@ -962,7 +962,7 @@ mt8186_mt6366_da7219_max98357_controls[] = { }; static struct snd_soc_card mt8186_mt6366_da7219_max98357_soc_card = { - .name = "mt8186_mt6366_da7219_max98357", + .name = "mt8186_da7219_max98357", .owner = THIS_MODULE, .dai_link = mt8186_mt6366_da7219_max98357_dai_links, .num_links = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links), diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c index 6c41706a5621..393d179d61de 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c @@ -938,7 +938,7 @@ mt8186_mt6366_rt1019_rt5682s_controls[] = { }; static struct snd_soc_card mt8186_mt6366_rt1019_rt5682s_soc_card = { - .name = "mt8186_mt6366_rt1019_rt5682s", + .name = "mt8186_rt1019_rt5682s", .owner = THIS_MODULE, .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links, .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links), -- 2.25.1

2 1

[PATCH linux-next] ASoC: atmel_ssc_dai: Remove the unneeded result variable
by cgel.zte＠gmail.com 23 Aug '22

23 Aug '22

From: ye xingchen <ye.xingchen(a)zte.com.cn> Return the value from asoc_ssc_init() directly instead of storing it in another redundant variable. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: ye xingchen <ye.xingchen(a)zte.com.cn> --- sound/soc/atmel/atmel_ssc_dai.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index e868b7e028d6..3763454436c1 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -891,7 +891,6 @@ static int asoc_ssc_init(struct device *dev) int atmel_ssc_set_audio(int ssc_id) { struct ssc_device *ssc; - int ret; /* If we can grab the SSC briefly to parent the DAI device off it */ ssc = ssc_request(ssc_id); @@ -903,9 +902,7 @@ int atmel_ssc_set_audio(int ssc_id) ssc_info[ssc_id].ssc = ssc; } - ret = asoc_ssc_init(&ssc->pdev->dev); - - return ret; + return asoc_ssc_init(&ssc->pdev->dev); } EXPORT_SYMBOL_GPL(atmel_ssc_set_audio); -- 2.25.1

2 1

[RESEND PATCH] ASoC: fsl_sai: Add support multi fifo sdma script
by Shengjiu Wang 23 Aug '22

23 Aug '22

With disabling combine mode, the multiple successive FIFO registers or non successive FIFO registers of SAI module can work with the sdma multi fifo script. This patch is to configure the necessary information to the SDMA engine driver for support multi fifo script. 'words_per_fifo' is the channels for each dataline 'n_fifos_src' and 'n_fifos_dst' are the fifo number 'stride_fifos_src' and 'stride_fifos_dst' are the stride between enable FIFOs 'maxburst' is the multiply of datalines Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in resend: - rebase to latest for-6.1 sound/soc/fsl/fsl_sai.c | 37 ++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_sai.h | 3 +++ 2 files changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a7fa6f0bf83d..0f92906d7a29 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -527,6 +527,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 slot_width = word_width; int adir = tx ? RX : TX; u32 pins, bclk; + u32 watermark; int ret, i; if (sai->slots) @@ -619,7 +620,15 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } - if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1) + /* + * Combine mode has limation: + * - Can't used for singel dataline/FIFO case except the FIFO0 + * - Can't used for multi dataline/FIFO case except the enabled FIFOs + * are successive and start from FIFO0 + * + * So for common usage, all multi fifo case disable the combine mode. + */ + if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1 || sai->is_multi_fifo_dma) regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_FCOMB_MASK, 0); else @@ -630,6 +639,26 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, dma_params->addr = sai->res->start + FSL_SAI_xDR0(tx) + dl_cfg[dl_cfg_idx].start_off[tx] * 0x4; + if (sai->is_multi_fifo_dma) { + sai->audio_config[tx].words_per_fifo = min(slots, channels); + if (tx) { + sai->audio_config[tx].n_fifos_dst = pins; + sai->audio_config[tx].stride_fifos_dst = dl_cfg[dl_cfg_idx].next_off[tx]; + } else { + sai->audio_config[tx].n_fifos_src = pins; + sai->audio_config[tx].stride_fifos_src = dl_cfg[dl_cfg_idx].next_off[tx]; + } + dma_params->maxburst = sai->audio_config[tx].words_per_fifo * pins; + dma_params->peripheral_config = &sai->audio_config[tx]; + dma_params->peripheral_size = sizeof(sai->audio_config[tx]); + + watermark = tx ? (sai->soc_data->fifo_depth - dma_params->maxburst) : + (dma_params->maxburst - 1); + regmap_update_bits(sai->regmap, FSL_SAI_xCR1(tx, ofs), + FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), + watermark); + } + /* Find a proper tcre setting */ for (i = 0; i < sai->soc_data->pins; i++) { trce_mask = (1 << (i + 1)) - 1; @@ -1257,6 +1286,7 @@ static int fsl_sai_probe(struct platform_device *pdev) char tmp[8]; int irq, ret, i; int index; + u32 dmas[4]; sai = devm_kzalloc(dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -1313,6 +1343,11 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_asoc_get_pll_clocks(&pdev->dev, &sai->pll8k_clk, &sai->pll11k_clk); + /* Use Multi FIFO mode depending on the support from SDMA script */ + ret = of_property_read_u32_array(np, "dmas", dmas, 4); + if (!sai->soc_data->use_edma && !ret && dmas[2] == IMX_DMATYPE_MULTI_SAI) + sai->is_multi_fifo_dma = true; + /* read dataline mask for rx and tx*/ ret = fsl_sai_read_dlcfg(sai); if (ret < 0) { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 17956b5731dc..697f6690068c 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -6,6 +6,7 @@ #ifndef __FSL_SAI_H #define __FSL_SAI_H +#include <linux/dma/imx-dma.h> #include <sound/dmaengine_pcm.h> #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ @@ -281,6 +282,7 @@ struct fsl_sai { bool is_lsb_first; bool is_dsp_mode; bool is_pdm_mode; + bool is_multi_fifo_dma; bool synchronous[2]; struct fsl_sai_dl_cfg *dl_cfg; unsigned int dl_cfg_cnt; @@ -300,6 +302,7 @@ struct fsl_sai { struct pm_qos_request pm_qos_req; struct pinctrl *pinctrl; struct pinctrl_state *pins_state; + struct sdma_peripheral_config audio_config[2]; }; #define TX 1 -- 2.34.1

2 1

[PATCH] ASoC: SOF: compress: Remove dai_posn variable
by Daniel Baluta 23 Aug '22

23 Aug '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> dai_posn is set but never used. Initial intention was to use dai_posn to shorthen one code line but it looks fine without it too. Fixes: c1a731c71359 ("ASoC: SOF: compress: Add support for computing timestamps") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/sof/compress.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 174b3d8e67dd..c8ae778a50d1 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -327,7 +327,6 @@ static int sof_compr_pointer(struct snd_soc_component *component, struct snd_compr_stream *cstream, struct snd_compr_tstamp *tstamp) { - u64 dai_posn; struct snd_sof_pcm *spcm; struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct sof_compr_stream *sstream = cstream->runtime->private_data; @@ -336,8 +335,6 @@ static int sof_compr_pointer(struct snd_soc_component *component, if (!spcm) return -EINVAL; - dai_posn = spcm->stream[cstream->direction].posn.dai_posn; - tstamp->sampling_rate = sstream->sampling_rate; tstamp->copied_total = sstream->copied_total; tstamp->pcm_io_frames = div_u64(spcm->stream[cstream->direction].posn.dai_posn, -- 2.27.0

2 1

[PATCH] ASoC: fsl_sai: Add support multi fifo sdma script
by Shengjiu Wang 23 Aug '22

23 Aug '22

With disabling combine mode, the multiple successive FIFO registers or non successive FIFO registers of SAI module can work with the sdma multi fifo script. This patch is to configure the necessary information to the SDMA engine driver for support multi fifo script. 'words_per_fifo' is the channels for each dataline 'n_fifos_src' and 'n_fifos_dst' are the fifo number 'stride_fifos_src' and 'stride_fifos_dst' are the stride between enable FIFOs 'maxburst' is the multiply of datalines Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 37 ++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_sai.h | 3 +++ 2 files changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index acf76ea21138..a2b375e2b56c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -533,6 +533,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 slot_width = word_width; int adir = tx ? RX : TX; u32 pins, bclk; + u32 watermark; int ret, i; if (sai->slot_width) @@ -627,7 +628,15 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } - if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1) + /* + * Combine mode has limation: + * - Can't used for singel dataline/FIFO case except the FIFO0 + * - Can't used for multi dataline/FIFO case except the enabled FIFOs + * are successive and start from FIFO0 + * + * So for common usage, all multi fifo case disable the combine mode. + */ + if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1 || sai->is_multi_fifo_dma) regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_FCOMB_MASK, 0); else @@ -638,6 +647,26 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, dma_params->addr = sai->res->start + FSL_SAI_xDR0(tx) + dl_cfg[dl_cfg_idx].start_off[tx] * 0x4; + if (sai->is_multi_fifo_dma) { + sai->audio_config[tx].words_per_fifo = min(slots, channels); + if (tx) { + sai->audio_config[tx].n_fifos_dst = pins; + sai->audio_config[tx].stride_fifos_dst = dl_cfg[dl_cfg_idx].next_off[tx]; + } else { + sai->audio_config[tx].n_fifos_src = pins; + sai->audio_config[tx].stride_fifos_src = dl_cfg[dl_cfg_idx].next_off[tx]; + } + dma_params->maxburst = sai->audio_config[tx].words_per_fifo * pins; + dma_params->peripheral_config = &sai->audio_config[tx]; + dma_params->peripheral_size = sizeof(sai->audio_config[tx]); + + watermark = tx ? (sai->soc_data->fifo_depth - dma_params->maxburst) : + (dma_params->maxburst - 1); + regmap_update_bits(sai->regmap, FSL_SAI_xCR1(tx, ofs), + FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), + watermark); + } + /* Find a proper tcre setting */ for (i = 0; i < sai->soc_data->pins; i++) { trce_mask = (1 << (i + 1)) - 1; @@ -1265,6 +1294,7 @@ static int fsl_sai_probe(struct platform_device *pdev) char tmp[8]; int irq, ret, i; int index; + u32 dmas[4]; sai = devm_kzalloc(dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -1321,6 +1351,11 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_asoc_get_pll_clocks(&pdev->dev, &sai->pll8k_clk, &sai->pll11k_clk); + /* Use Multi FIFO mode depending on the support from SDMA script */ + ret = of_property_read_u32_array(np, "dmas", dmas, 4); + if (!sai->soc_data->use_edma && !ret && dmas[2] == IMX_DMATYPE_MULTI_SAI) + sai->is_multi_fifo_dma = true; + /* read dataline mask for rx and tx*/ ret = fsl_sai_read_dlcfg(sai); if (ret < 0) { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 17956b5731dc..697f6690068c 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -6,6 +6,7 @@ #ifndef __FSL_SAI_H #define __FSL_SAI_H +#include <linux/dma/imx-dma.h> #include <sound/dmaengine_pcm.h> #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ @@ -281,6 +282,7 @@ struct fsl_sai { bool is_lsb_first; bool is_dsp_mode; bool is_pdm_mode; + bool is_multi_fifo_dma; bool synchronous[2]; struct fsl_sai_dl_cfg *dl_cfg; unsigned int dl_cfg_cnt; @@ -300,6 +302,7 @@ struct fsl_sai { struct pm_qos_request pm_qos_req; struct pinctrl *pinctrl; struct pinctrl_state *pins_state; + struct sdma_peripheral_config audio_config[2]; }; #define TX 1 -- 2.17.1

2 2

[PATCH] ASoC: mchp-spdiftx: Fix clang -Wbitfield-constant-conversion
by Nathan Chancellor 23 Aug '22

23 Aug '22

A recent change in clang strengthened its -Wbitfield-constant-conversion to warn when 1 is assigned to a 1-bit signed integer bitfield, as it can only be 0 or -1, not 1: sound/soc/atmel/mchp-spdiftx.c:505:20: error: implicit truncation from 'int' to bit-field changes value from 1 to -1 [-Werror,-Wbitfield-constant-conversion] dev->gclk_enabled = 1; ^ ~ 1 error generated. The actual value of the field is never checked, just that it is not zero, so there is not a real bug here. However, it is simple enough to silence the warning by making the bitfield unsigned, which matches the mchp-spdifrx driver. Fixes: 06ca24e98e6b ("ASoC: mchp-spdiftx: add driver for S/PDIF TX Controller") Link: https://github.com/ClangBuiltLinux/linux/issues/1686 Link: https://github.com/llvm/llvm-project/commit/82afc9b169a67e8b8a1862fb9c41a2c… Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/atmel/mchp-spdiftx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index 4850a177803d..ab2d7a791f39 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -196,7 +196,7 @@ struct mchp_spdiftx_dev { struct clk *pclk; struct clk *gclk; unsigned int fmt; - int gclk_enabled:1; + unsigned int gclk_enabled:1; }; static inline int mchp_spdiftx_is_running(struct mchp_spdiftx_dev *dev) base-commit: 15205c2829ca2cbb5ece5ceaafe1171a8470e62b -- 2.37.1

4 7

[PATCH 0/2] ASoC: SOF: Kconfig: Fix the dependency for client modules
by Peter Ujfalusi 23 Aug '22

23 Aug '22

Hi, There is still a chance to end up with a client driver selected as built in while the core SOF is as module. Fix this by making the client drivers depend on SND_SOC_SOF. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: Kconfig: Make IPC_FLOOD_TEST depend on SND_SOC_SOF ASoC: SOF: Kconfig: Make IPC_MESSAGE_INJECTOR depend on SND_SOC_SOF sound/soc/sof/Kconfig | 2 ++ 1 file changed, 2 insertions(+) -- 2.37.2

2 3

[PATCH] ASoC: SOF: ipc4-topology: fix alh_group_ida max value
by Pierre-Louis Bossart 23 Aug '22

23 Aug '22

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> group_id is from 0 ~ ALH_MULTI_GTW_COUNT - 1, not 0 ~ ALH_MULTI_GTW_COUNT. Fixes: 0e357b5290532 ("ASoC: SOF: ipc4-topology: add SoundWire/ALH aggregation support" Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index c6abfaf5d5326..64929dc9af397 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1251,7 +1251,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, if (blob->alh_cfg.count > 1) { int group_id; - group_id = ida_alloc_max(&alh_group_ida, ALH_MULTI_GTW_COUNT, + group_id = ida_alloc_max(&alh_group_ida, ALH_MULTI_GTW_COUNT - 1, GFP_KERNEL); if (group_id < 0) -- 2.34.1

2 2

[PATCH 0/5] ASoC: nau8xxx: Implement hw constraint for rates
by Takashi Iwai 23 Aug '22

23 Aug '22

Hi, this is a series of patches to address the issues on nau8xxx codecs I've stumbled upon while dealing with a bug report for Steam Deck. Most of them are to implement the missing hw constraint for rate restrictions while one patch is to fix the semaphore unbalance in nau8824 driver. Takashi === Takashi Iwai (5): ASoC: nau8821: Implement hw constraint for rates ASoC: nau8824: Fix semaphore unbalance at error paths ASoC: nau8824: Implement hw constraint for rates ASoC: nau8825: Implement hw constraint for rates ASoC: nau8540: Implement hw constraint for rates sound/soc/codecs/nau8540.c | 40 +++++++++++++----- sound/soc/codecs/nau8821.c | 66 ++++++++++++++++-------------- sound/soc/codecs/nau8824.c | 80 ++++++++++++++++++++---------------- sound/soc/codecs/nau8825.c | 83 +++++++++++++++++++++----------------- 4 files changed, 156 insertions(+), 113 deletions(-) -- 2.35.3

2 6

[PATCH 0/2] soundwire: remove use of __func__ in dev_dbg
by Bard Liao 23 Aug '22

23 Aug '22

The module and function information can be added with 'modprobe foo dyndbg=+pmf' Pierre-Louis Bossart (2): soundwire: bus: remove use of __func__ in dev_dbg soundwire: intel: remove use of __func__ in dev_dbg drivers/soundwire/bus.c | 16 ++++++++-------- drivers/soundwire/intel.c | 14 +++++++------- 2 files changed, 15 insertions(+), 15 deletions(-) -- 2.25.1

2 3

[PATCH] soundwire: dmi-quirks: add remapping for HP Omen 16-k0005TX
by Bard Liao 23 Aug '22

23 Aug '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The DSDT for this device has a number of problems: a) it lists rt711 on link0 and link1, but link1 is disabled b) the rt711 entry on link0 uses the wrong v2 instead of v3 (SDCA) c) the rt1316 amplifier on link3 is not listed. Add a remapping table to work-around these BIOS shenanigans. BugLink: https://github.com/thesofproject/sof/issues/5955 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/dmi-quirks.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 747983743a14..f81cdd83ec26 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -55,7 +55,26 @@ static const struct adr_remap dell_sku_0A3E[] = { {} }; +/* + * The HP Omen 16-k0005TX does not expose the correct version of RT711 on link0 + * and does not expose a RT1316 on link3 + */ +static const struct adr_remap hp_omen_16[] = { + /* rt711-sdca on link0 */ + { + 0x000020025d071100ull, + 0x000030025d071101ull + }, + /* rt1316-sdca on link3 */ + { + 0x000120025d071100ull, + 0x000330025d131601ull + }, + {} +}; + static const struct dmi_system_id adr_remap_quirk_table[] = { + /* TGL devices */ { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "HP"), @@ -78,6 +97,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)dell_sku_0A3E, }, + /* ADL devices */ + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_PRODUCT_NAME, "OMEN by HP Gaming Laptop 16-k0xxx"), + }, + .driver_data = (void *)hp_omen_16, + }, {} }; -- 2.25.1

2 1

[PATCH v3 linux-next] soundwire: intel: Remove unnecessary TODO
by Khalid Masum 23 Aug '22

23 Aug '22

The capabilities enabled for multi-link are required as part of the programming sequences, even when a stream uses a single link we still use the syncArm/syncGo sequences. Therefore the TODO is no longer necessary. Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Khalid Masum <khalid.masum.92(a)gmail.com> --- Previous discussions: - v2 Link: https://lore.kernel.org/lkml/20220816175954.5259-1-khalid.masum.92@gmail.co… - v1 Link: https://lore.kernel.org/lkml/20220814080416.7531-1-khalid.masum.92@gmail.co… Changes since v2: - Update commit title. Changes since v1: - Remove the todo, leave rest of the code as is. - Update commit message. drivers/soundwire/bus.c | 1 - 1 file changed, 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 8d4000664fa3..18158825a6d8 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -75,7 +75,6 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, /* * Initialize multi_link flag - * TODO: populate this flag by reading property from FW node */ bus->multi_link = false; if (bus->ops->read_prop) { -- 2.37.1

2 1

[PATCH v2] soundwire: qcom: Update error prints to debug prints
by Srinivasa Rao Mandadapu 23 Aug '22

23 Aug '22

Update error prints to debug prints to avoid redundant logging in kernel boot time, as these prints are informative prints in irq handler. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Reviewed-by: Andrew Halaney <ahalaney(a)redhat.com> --- Changes since v1: -- Remove redundant __func__ argument. -- Replace dev_dbg with dev_dbg_ratelimited. drivers/soundwire/qcom.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 9df970e..976ae75 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -573,11 +573,10 @@ static irqreturn_t qcom_swrm_irq_handler(int irq, void *dev_id) break; case SWRM_INTERRUPT_STATUS_NEW_SLAVE_ATTACHED: case SWRM_INTERRUPT_STATUS_CHANGE_ENUM_SLAVE_STATUS: - dev_err_ratelimited(swrm->dev, "%s: SWR new slave attached\n", - __func__); + dev_dbg_ratelimited(swrm->dev, "SWR new slave attached\n"); swrm->reg_read(swrm, SWRM_MCP_SLV_STATUS, &slave_status); if (swrm->slave_status == slave_status) { - dev_err(swrm->dev, "Slave status not changed %x\n", + dev_dbg(swrm->dev, "Slave status not changed %x\n", slave_status); } else { qcom_swrm_get_device_status(swrm); -- 2.7.4

3 2

[PATCH] ASoC: fsl_mqs: Fix supported clock DAI format
by Shengjiu Wang 23 Aug '22

23 Aug '22

The MQS works as codec DAI, not cpu DAI. It is clock consumer, not clock privider. Fixes: 3b14c15a333b ("ASoC: fsl: Update to use set_fmt_new callback") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_mqs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index c1e2f671191b..4922e6795b73 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -122,7 +122,7 @@ static int fsl_mqs_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) } switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { - case SND_SOC_DAIFMT_BP_FP: + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.34.1

2 1

Re: [PATCH] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 23 Aug '22

23 Aug '22

On Mon, Aug 22, 2022 at 02:27:26PM +0100, Mark Brown wrote: > On Sat, Aug 20, 2022 at 09:36:53PM -0300, Geraldo Nascimento wrote: > > > + /* > > + * dw-hdmi introduced insert_pcuv bit in version 2.10a. > > + * When set (1'b1), this bit enables the insertion of the PCUV > > + * (Parity, Channel Status, User bit and Validity) bits on the > > + * incoming audio stream (support limited to Linear PCM audio) > > + */ > > + > > + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21) > > + conf2 |= HDMI_AUD_CONF2_INSERT_PCUV; > > So what if we're not handlign linear PCM? Hi Mark, I think we may be able to restrict the fix for L-PCM only by checking byte 0 of iec.status, specifically bit 1. Our define is called IEC958_AES0_NONAUDIO but https://tech.ebu.ch/docs/tech/tech3250.pdf calls it the "Linear PCM identification" bit. There's also a supplement to AES/EBU 3250 in https://tech.ebu.ch/docs/n/n009_1.pdf Let me know if the following is OK. I'll be happy to submit V2 if it is. Thanks, Geraldo Nascimento --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,18 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in version 2.10a. + * When set (1'b1), this bit enables the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) bits on the + * incoming audio stream (support limited to Linear PCM audio) + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +122,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; }

2 1

[PATCH] soundwire: qcom: remove duplicate reset control get
by Srinivas Kandagatla 23 Aug '22

23 Aug '22

Looks like adding clock gate flag patch forgot to remove the old code that gets reset control. This causes below crash on platforms that do not need reset. [ 15.653501] reset_control_reset+0x124/0x170 [ 15.653508] qcom_swrm_init+0x50/0x1a0 [ 15.653514] qcom_swrm_probe+0x320/0x668 [ 15.653519] platform_probe+0x68/0xe0 [ 15.653529] really_probe+0xbc/0x2a8 [ 15.653535] __driver_probe_device+0x7c/0xe8 [ 15.653541] driver_probe_device+0x40/0x110 [ 15.653547] __device_attach_driver+0x98/0xd0 [ 15.653553] bus_for_each_drv+0x68/0xd0 [ 15.653559] __device_attach+0xf4/0x188 [ 15.653565] device_initial_probe+0x14/0x20 Fix this by removing old code. Reported-by: Amit Pundir <amit.pundir(a)linaro.org> Fixes: 1fd0d85affe4 ("soundwire: qcom: Add flag for software clock gating check") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 9df970eeca45..a43961ad4614 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1356,10 +1356,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.compute_params = &qcom_swrm_compute_params; ctrl->bus.clk_stop_timeout = 300; - ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); - if (IS_ERR(ctrl->audio_cgcr)) - dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); - ret = qcom_swrm_get_port_config(ctrl); if (ret) goto err_clk; -- 2.21.0

3 2

Re: [PATCH v8 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by 班涛 23 Aug '22

23 Aug '22

Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> 于2022年8月12日周五 23:20写道： > On 8/11/2022 4:49 PM, Ban Tao wrote: > > The Allwinner H6 and later SoCs have an DMIC block > > which is capable of capture. > > > > Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> > > --- > > ... > > > + > > +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) > > Seeing supported rates in hw_params callback, you probably want to take > a look at SNDRV_PCM_RATE_KNOT? > No, I don't need SNDRV_PCM_RATE_KNOT. Dmic does not support unknown rates. > > > +#define SUN50I_DMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | > SNDRV_PCM_FMTBIT_S24_LE) > > + > > +static struct snd_soc_dai_driver sun50i_dmic_dai = { > > + .capture = { > > + .channels_min = 1, > > + .channels_max = 8, > > + .rates = SUN50I_DMIC_RATES, > > + .formats = SUN50I_DMIC_FORMATS, > > + .sig_bits = 21, > > + }, > > + .probe = sun50i_dmic_soc_dai_probe, > > + .ops = &sun50i_dmic_dai_ops, > > + .name = "dmic", > > +}; > > + > > ... > >

1 0

[PATCH] ALSA: hda: intel-nhlt: Correct the handling of fmt_config flexible array
by Peter Ujfalusi 23 Aug '22

23 Aug '22

The struct nhlt_format's fmt_config is a flexible array, it must not be used as normal array. When moving to the next nhlt_fmt_cfg we need to take into account the data behind the ->config.caps (indicated by ->config.size). Fixes: a864e8f159b13 ("ALSA: hda: intel-nhlt: verify config type") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> --- sound/hda/intel-nhlt.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index 9db5ccd9aa2d..13bb0ccfb36c 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -55,16 +55,22 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) /* find max number of channels based on format_configuration */ if (fmt_configs->fmt_count) { + struct nhlt_fmt_cfg *fmt_cfg = fmt_configs->fmt_config; + dev_dbg(dev, "found %d format definitions\n", fmt_configs->fmt_count); for (i = 0; i < fmt_configs->fmt_count; i++) { struct wav_fmt_ext *fmt_ext; - fmt_ext = &fmt_configs->fmt_config[i].fmt_ext; + fmt_ext = &fmt_cfg->fmt_ext; if (fmt_ext->fmt.channels > max_ch) max_ch = fmt_ext->fmt.channels; + + /* Move to the next nhlt_fmt_cfg */ + fmt_cfg = (struct nhlt_fmt_cfg *)(fmt_cfg->config.caps + + fmt_cfg->config.size); } dev_dbg(dev, "max channels found %d\n", max_ch); } else { -- 2.37.2

2 1

data-race in snd_seq_oss_midi_check_exit_port / snd_seq_oss_midi_setup
by Abhishek Shah 23 Aug '22

23 Aug '22

Hi all, We found a race involving the *max_midi_devs* variable. We see an interleaving where the following check here <https://elixir.bootlin.com/linux/v5.18-rc5/source/sound/core/seq/oss/seq_os…> passes before the *snd_seq_oss_midi_check_exit_port*() finishes, but this check should not pass if *max_midi_devs* will become zero, but we are not sure of its implications in terms of security impact. Please let us know what you think. Thanks! *-------------------Report---------------------* *write* to 0xffffffff88382f80 of 4 bytes by task 6541 on cpu 0: snd_seq_oss_midi_check_exit_port+0x1a6/0x270 sound/core/seq/oss/seq_oss_midi.c:237 receive_announce+0x193/0x1b0 sound/core/seq/oss/seq_oss_init.c:143 snd_seq_deliver_single_event+0x30d/0x4e0 sound/core/seq/seq_clientmgr.c:640 deliver_to_subscribers sound/core/seq/seq_clientmgr.c:695 [inline] snd_seq_deliver_event+0x38c/0x490 sound/core/seq/seq_clientmgr.c:830 snd_seq_kernel_client_dispatch+0x189/0x1a0 sound/core/seq/seq_clientmgr.c:2339 snd_seq_system_broadcast+0x98/0xd0 sound/core/seq/seq_system.c:86 snd_seq_ioctl_delete_port+0x9a/0xc0 sound/core/seq/seq_clientmgr.c:1356 snd_seq_ioctl+0x198/0x2d0 sound/core/seq/seq_clientmgr.c:2173 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:870 [inline] __se_sys_ioctl+0xe1/0x150 fs/ioctl.c:856 __x64_sys_ioctl+0x43/0x50 fs/ioctl.c:856 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0x90 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae *read* to 0xffffffff88382f80 of 4 bytes by task 6542 on cpu 1: snd_seq_oss_midi_setup+0x1b/0x40 sound/core/seq/oss/seq_oss_midi.c:273 snd_seq_oss_open+0x364/0x900 sound/core/seq/oss/seq_oss_init.c:198 odev_open+0x55/0x70 sound/core/seq/oss/seq_oss.c:128 soundcore_open+0x315/0x3a0 sound/sound_core.c:593 chrdev_open+0x373/0x3f0 fs/char_dev.c:414 do_dentry_open+0x543/0x8f0 fs/open.c:824 vfs_open+0x47/0x50 fs/open.c:958 do_open fs/namei.c:3476 [inline] path_openat+0x1906/0x1dc0 fs/namei.c:3609 do_filp_open+0xef/0x200 fs/namei.c:3636 do_sys_openat2+0xa5/0x2a0 fs/open.c:1213 do_sys_open fs/open.c:1229 [inline] __do_sys_openat fs/open.c:1245 [inline] __se_sys_openat fs/open.c:1240 [inline] __x64_sys_openat+0xf0/0x120 fs/open.c:1240 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0x90 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae Reported by Kernel Concurrency Sanitizer on: CPU: 1 PID: 6542 Comm: syz-executor2-n Not tainted 5.18.0-rc5+ #107 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.15.0-1 04/01/2014 *Reproducing Inputs* Input CPU 0: r0 = openat$sndseq(0xffffffffffffff9c, &(0x7f0000000040)='/dev/snd/seq\x00', 0x0) ioctl$SNDRV_SEQ_IOCTL_CREATE_PORT(r0, 0xc0a85320, &(0x7f0000000240)={{0x80}, 'port1\x00', 0x10}) ioctl$SNDRV_SEQ_IOCTL_SET_CLIENT_POOL(r0, 0x40a85321, &(0x7f0000000100)={0x80}) Input CPU 1: r0 = openat$sequencer2(0xffffff9c, &(0x7f0000000000)='/dev/sequencer2\x00', 0x0, 0x0) ioctl$SNDCTL_SYNTH_INFO(r0, 0xc08c5102, &(0x7f0000000200)={"02961a3ce6d4828f8b5559726313251b55fa11d8d65406f1f33c9af8e3f8", 0xffffffff})

3 4

[PATCH 0/5] ALSA: Drop hackish GFP giveaway for CONTINUOUS pages
by Takashi Iwai 23 Aug '22

23 Aug '22

Hi, this is a series of cleanup patches for dropping the current hackish way of passing the GFP_* flags for CONTINOUS and VMALLOC memory allocations. There are only three users for this legacy feature, and all of them seem superfluous. And, if any driver requires the memory restriction in future, it can now pass the proper device pointer for specifying the DMA mask. Takashi === Takashi Iwai (5): ALSA: vx: Drop superfluous GFP setup ALSA: pdaudiocf: Drop superfluous GFP setup ASoC: Intel: sst: Switch to standard device pages ALSA: memalloc: Drop special handling of GFP for CONTINUOUS allocation ALSA: doc: Drop snd_dma_continuous_data() usages .../kernel-api/writing-an-alsa-driver.rst | 21 ++-- include/sound/memalloc.h | 3 - sound/core/memalloc.c | 113 ++++++++---------- sound/drivers/vx/vx_pcm.c | 3 +- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 3 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 7 +- 6 files changed, 61 insertions(+), 89 deletions(-) -- 2.35.3

2 6

[PATCH v2 0/4] ASoC platform driver for Apple MCA
by Martin Povišer 23 Aug '22

23 Aug '22

Hi all, this is v2 of the ASoC platform driver for MCA peripheral on Apple M1, to be followed by a machine driver later for sound support on the new Apple devices. This time I attached a DTS patch for t8103 (M1), which I guess should go through Marcan's (in CC) tree. Martin Changes since v1: - reflect the set_fmt provider/consumer change in 6.0-rc1 - fix the probe function requesting optional reset - unroll 'dma-names' in schema - fix binding schema/code discrepancy in register ranges - minor things Changes since 'macaudio RFC v2' [0]: - addition of locking (extra commit) - transition to set_bclk_ratio (instead of getting the bclk ratio from set_sysclk) - using shared reset control and documenting the reset in binding - formatting, comments, and a minor fix to hw driving [0] https://lore.kernel.org/asahi/20220606191910.16580-1-povik+lin@cutebit.org/ Martin Povišer (4): dt-bindings: sound: Add Apple MCA I2S transceiver arm64: dts: apple: t8103: Add MCA and its support ASoC: apple: mca: Start new platform driver ASoC: apple: mca: Add locks on foreign cluster access .../devicetree/bindings/sound/apple,mca.yaml | 131 ++ MAINTAINERS | 8 + arch/arm64/boot/dts/apple/t8103.dtsi | 70 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/apple/Kconfig | 9 + sound/soc/apple/Makefile | 3 + sound/soc/apple/mca.c | 1170 +++++++++++++++++ 8 files changed, 1393 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/apple,mca.yaml create mode 100644 sound/soc/apple/Kconfig create mode 100644 sound/soc/apple/Makefile create mode 100644 sound/soc/apple/mca.c -- 2.33.0

4 19

DSD hardware quirks and usb vendor id hardcoding
by GitHub issues - opened 23 Aug '22

23 Aug '22

alsa-project/alsa-lib issue #266 was opened from soholt: I am using tinyusb for UAC2 audio class and line 492 https://github.com/hathach/tinyusb/blob/master/src/class/audio/audio.h allows: "AUDIO_DATA_FORMAT_TYPE_I_RAW_DATA = 0x80000000" cat /proc/asound/ reports it as "Format: SPECIAL", but I wanted to be "SPECIAL DSD_U32_xx", so for fun I tried: 0x80000001 which reported as = S32_LE SPECIAL, 0x80000002 = U8 SPECIAL, 0x80000003 = U8 S32_LE SPECIAL 0x800000ff = U8 S32_LE FLOAT_LE MU_LAW A_LAW SPECIAL - seem to recursively add more Then I found quirks line 1582 https://github.com/torvalds/linux/blob/master/sound/usb/quirks.c "The UAC standards don't have a designated bit field to denote DSD-capable interfaces, hence all hardware that is known to support this format has to be listed here." So if I change my usb vendor id to XMOS 0x20b1, only then "Format: SPECIAL" becomes "Format: SPECIAL DSD_U32_BE" I have no idea if range 0x80000001++ is used for anything or if it could be used for DSD formats like: 0x80000001 = SPECIAL DSD_U8 0x80000002 = SPECIAL DSD_U16_LE 0x80000003 = SPECIAL DSD_U16_BE 0x80000004 = SPECIAL DSD_U32_LE 0x80000005 = SPECIAL DSD_U32_BE so no future hardcoding is required. Thanks, Gin Issue URL : https://github.com/alsa-project/alsa-lib/issues/266 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda: cleanup definitions for multi-link registers
by Pierre-Louis Bossart 23 Aug '22

23 Aug '22

For some reason two masks are used without the AZX prefix, and the pattern MLCLT should be ML_LCTL for consistency. Pure rename, no functionality change. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- include/sound/hda_register.h | 20 +++++++++++--------- sound/hda/ext/hdac_ext_controller.c | 16 ++++++++-------- sound/pci/hda/hda_intel.c | 14 +++++++------- 3 files changed, 26 insertions(+), 24 deletions(-) diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index ad8b71b1dbb67..d37cf43546eba 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -260,7 +260,18 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LCAP 0x00 #define AZX_REG_ML_LCTL 0x04 + +#define AZX_ML_LCTL_CPA BIT(23) +#define AZX_ML_LCTL_CPA_SHIFT 23 +#define AZX_ML_LCTL_SPA BIT(16) +#define AZX_ML_LCTL_SPA_SHIFT 16 +#define AZX_ML_LCTL_SCF GENMASK(3, 0) + #define AZX_REG_ML_LOSIDV 0x08 + +/* bit0 is reserved, with BIT(1) mapping to stream1 */ +#define AZX_ML_LOSIDV_STREAM_MASK 0xFFFE + #define AZX_REG_ML_LSDIID 0x0C #define AZX_REG_ML_LPSOO 0x10 #define AZX_REG_ML_LPSIO 0x12 @@ -268,15 +279,6 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LOUTPAY 0x20 #define AZX_REG_ML_LINPAY 0x30 -/* bit0 is reserved, with BIT(1) mapping to stream1 */ -#define ML_LOSIDV_STREAM_MASK 0xFFFE - -#define ML_LCTL_SCF_MASK 0xF -#define AZX_MLCTL_SPA (0x1 << 16) -#define AZX_MLCTL_CPA (0x1 << 23) -#define AZX_MLCTL_SPA_SHIFT 16 -#define AZX_MLCTL_CPA_SHIFT 23 - /* registers for DMA Resume Capability Structure */ #define AZX_DRSM_CAP_ID 0x5 #define AZX_REG_DRSM_CTL 0x4 diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index a42f66f561f57..80876b9a87f46 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -170,7 +170,7 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) { int timeout; u32 val; - int mask = (1 << AZX_MLCTL_CPA_SHIFT); + int mask = (1 << AZX_ML_LCTL_CPA_SHIFT); udelay(3); timeout = 150; @@ -178,10 +178,10 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) do { val = readl(link->ml_addr + AZX_REG_ML_LCTL); if (enable) { - if (((val & mask) >> AZX_MLCTL_CPA_SHIFT)) + if (((val & mask) >> AZX_ML_LCTL_CPA_SHIFT)) return 0; } else { - if (!((val & mask) >> AZX_MLCTL_CPA_SHIFT)) + if (!((val & mask) >> AZX_ML_LCTL_CPA_SHIFT)) return 0; } udelay(3); @@ -197,7 +197,7 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link) { snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, - AZX_MLCTL_SPA, AZX_MLCTL_SPA); + AZX_ML_LCTL_SPA, AZX_ML_LCTL_SPA); return check_hdac_link_power_active(link, true); } @@ -209,7 +209,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_up); */ int snd_hdac_ext_bus_link_power_down(struct hdac_ext_link *link) { - snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, AZX_MLCTL_SPA, 0); + snd_hdac_updatel(link->ml_addr, AZX_REG_ML_LCTL, AZX_ML_LCTL_SPA, 0); return check_hdac_link_power_active(link, false); } @@ -226,7 +226,7 @@ int snd_hdac_ext_bus_link_power_up_all(struct hdac_bus *bus) list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, - AZX_MLCTL_SPA, AZX_MLCTL_SPA); + AZX_ML_LCTL_SPA, AZX_ML_LCTL_SPA); ret = check_hdac_link_power_active(hlink, true); if (ret < 0) return ret; @@ -247,7 +247,7 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_bus *bus) list_for_each_entry(hlink, &bus->hlink_list, list) { snd_hdac_updatel(hlink->ml_addr, AZX_REG_ML_LCTL, - AZX_MLCTL_SPA, 0); + AZX_ML_LCTL_SPA, 0); ret = check_hdac_link_power_active(hlink, false); if (ret < 0) return ret; @@ -281,7 +281,7 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, * clear the register to invalidate all the output streams */ snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, - ML_LOSIDV_STREAM_MASK, 0); + AZX_ML_LOSIDV_STREAM_MASK, 0); /* * wait for 521usec for codec to report status * HDA spec section 4.3 - Codec Discovery diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7720978dc1326..bf9df9bc8f1bf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -489,14 +489,14 @@ static int intel_ml_lctl_set_power(struct azx *chip, int state) * If other links are enabled for stream, they need similar fix */ val = readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); - val &= ~AZX_MLCTL_SPA; - val |= state << AZX_MLCTL_SPA_SHIFT; + val &= ~AZX_ML_LCTL_SPA; + val |= state << AZX_ML_LCTL_SPA_SHIFT; writel(val, bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); /* wait for CPA */ timeout = 50; while (timeout) { if (((readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL)) & - AZX_MLCTL_CPA) == (state << AZX_MLCTL_CPA_SHIFT)) + AZX_ML_LCTL_CPA) == (state << AZX_ML_LCTL_CPA_SHIFT)) return 0; timeout--; udelay(10); @@ -514,15 +514,15 @@ static void intel_init_lctl(struct azx *chip) /* 0. check lctl register value is correct or not */ val = readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); /* if SCF is already set, let's use it */ - if ((val & ML_LCTL_SCF_MASK) != 0) + if ((val & AZX_ML_LCTL_SCF) != 0) return; /* * Before operating on SPA, CPA must match SPA. * Any deviation may result in undefined behavior. */ - if (((val & AZX_MLCTL_SPA) >> AZX_MLCTL_SPA_SHIFT) != - ((val & AZX_MLCTL_CPA) >> AZX_MLCTL_CPA_SHIFT)) + if (((val & AZX_ML_LCTL_SPA) >> AZX_ML_LCTL_SPA_SHIFT) != + ((val & AZX_ML_LCTL_CPA) >> AZX_ML_LCTL_CPA_SHIFT)) return; /* 1. turn link down: set SPA to 0 and wait CPA to 0 */ @@ -532,7 +532,7 @@ static void intel_init_lctl(struct azx *chip) goto set_spa; /* 2. update SCF to select a properly audio clock*/ - val &= ~ML_LCTL_SCF_MASK; + val &= ~AZX_ML_LCTL_SCF; val |= intel_get_lctl_scf(chip); writel(val, bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL); -- 2.34.1

2 1

Adding Focusrite Scarlett 2i4 gen2
by GitHub pull_request - opened 23 Aug '22

23 Aug '22

alsa-project/alsa-ucm-conf pull request #203 was opened from PatrickLang: This cleans up the Focusrite Scarlett 2i4 Gen2 device a bit. This device is split into two stereo pairs for output, and has 2 mic/line/instrument inputs. There are no toggles or mixer controls exposed to Alsa. It has hardware switches for: - Headphones - can monitor line outputs 1+2 or 3+4 - Headphone input/playback mix - Monitor out (Line 1+2) - Each input has a level pot, and switches for pad and Z (line/inst) - There is 1 physical +48v switch Please let me know if any more info would be helpful. I have captured some more notes showing before/after state at https://gist.github.com/PatrickLang/79a3090eea6f2d640179457b5d428134#scarle… I'm planning to write a UCM2 profile for the more complicated Focusrite Scarlett 18i8 Gen2 next which has more inputs, but also multiple mix busses controllable through Alsa Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/203 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/203.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: codes: src4xxx: Avoid clang -Wsometimes-uninitialized in src4xxx_hw_params()
by Nathan Chancellor 23 Aug '22

23 Aug '22

Clang warns: sound/soc/codecs/src4xxx.c:280:3: error: variable 'd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:298:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); ^ sound/soc/codecs/src4xxx.c:223:20: note: initialize the variable 'd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'jd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:293:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); ^~ sound/soc/codecs/src4xxx.c:223:17: note: initialize the variable 'jd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'pj' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:288:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); ^~ sound/soc/codecs/src4xxx.c:223:13: note: initialize the variable 'pj' to silence this warning int val, pj, jd, d; ^ = 0 3 errors generated. According to the comment in the default case, other parts of the chip are still functional without these values so just return 0 in the default case to avoid using these variables uninitialized. Link: https://github.com/ClangBuiltLinux/linux/issues/1691 Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: "Sudip Mukherjee (Codethink)" <sudipm.mukherjee(a)gmail.com> Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/codecs/src4xxx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/src4xxx.c b/sound/soc/codecs/src4xxx.c index a8f143057b41..cf45caa4bf7f 100644 --- a/sound/soc/codecs/src4xxx.c +++ b/sound/soc/codecs/src4xxx.c @@ -283,7 +283,7 @@ static int src4xxx_hw_params(struct snd_pcm_substream *substream, */ dev_info(component->dev, "Couldn't set the RCV PLL as this master clock rate is unknown\n"); - break; + return 0; } ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); if (ret < 0) base-commit: 94f072748337424c9cf92cd018532a34db3a5516 -- 2.37.2

2 3

Re: [PATCH/RESEND 2 v1] ASoC: trival spelling fix.
by Maurycy Z 23 Aug '22

23 Aug '22

"system" is mis-spelled as "ststem" in the help for SND_AUDIO_GRAPH_CARD. Acked-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Maurycy Zalewski <10maurycy10(a)gmail.com> --- diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index b6df4e26b..72f3446e1 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -16,7 +16,7 @@ config SND_AUDIO_GRAPH_CARD help This option enables generic simple sound card support with OF-graph DT bindings. - It also support DPCM of multi CPU single Codec ststem. + It also support DPCM of multi CPU single Codec system. config SND_AUDIO_GRAPH_CARD2 tristate "ASoC Audio Graph sound card2 support"

1 0

data-race in snd_seq_client_use_ptr / snd_seq_client_use_ptr
by Abhishek Shah 22 Aug '22

22 Aug '22

Hi all, We found a data race involving the *card_requested[card]* variable here <https://elixir.bootlin.com/linux/v5.18-rc5/source/sound/core/seq/seq_client…>, which can cause *snd_request_card* to be called multiple times. We are not sure if this has security implications however, but we would still like to report it. Please let us know what you think. Thanks! *----------Report-----------* *write* to 0xffffffff88382191 of 1 bytes by task 6541 on cpu 0: snd_seq_client_use_ptr+0x254/0x610 sound/core/seq/seq_clientmgr.c:146 snd_seq_info_clients_read+0xd8/0x480 sound/core/seq/seq_clientmgr.c:2475 snd_info_seq_show+0x81/0xa0 sound/core/info.c:361 seq_read_iter+0x2d2/0x8e0 fs/seq_file.c:230 seq_read+0x1c9/0x210 fs/seq_file.c:162 pde_read fs/proc/inode.c:311 [inline] proc_reg_read+0x123/0x1b0 fs/proc/inode.c:323 vfs_read+0x1b5/0x6e0 fs/read_write.c:480 ksys_read+0xde/0x190 fs/read_write.c:620 __do_sys_read fs/read_write.c:630 [inline] __se_sys_read fs/read_write.c:628 [inline] __x64_sys_read+0x43/0x50 fs/read_write.c:628 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0x90 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae *read* to 0xffffffff88382191 of 1 bytes by task 6542 on cpu 1: snd_seq_client_use_ptr+0x189/0x610 sound/core/seq/seq_clientmgr.c:145 snd_seq_info_clients_read+0xd8/0x480 sound/core/seq/seq_clientmgr.c:2475 snd_info_seq_show+0x81/0xa0 sound/core/info.c:361 seq_read_iter+0x2d2/0x8e0 fs/seq_file.c:230 seq_read+0x1c9/0x210 fs/seq_file.c:162 pde_read fs/proc/inode.c:311 [inline] proc_reg_read+0x123/0x1b0 fs/proc/inode.c:323 vfs_read+0x1b5/0x6e0 fs/read_write.c:480 ksys_read+0xde/0x190 fs/read_write.c:620 __do_sys_read fs/read_write.c:630 [inline] __se_sys_read fs/read_write.c:628 [inline] __x64_sys_read+0x43/0x50 fs/read_write.c:628 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0x90 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae Reported by Kernel Concurrency Sanitizer on: CPU: 1 PID: 6542 Comm: syz-executor2-n Not tainted 5.18.0-rc5+ #107 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.15.0-1 04/01/2014 *Reproducing Inputs* Input CPU 0: r0 = openat$procfs(0xffffffffffffff9c, &(0x7f00000004c0)='/proc/asound/seq/clients\x00', 0x0, 0x0) read$ptp(r0, &(0x7f0000001280)=""/4096, 0x1000) Input CPU 1: r0 = openat$procfs(0xffffff9c, &(0x7f0000000000)='/proc/asound/seq/clients\x00', 0x0, 0x0) syz_fuse_handle_req(r0, &(0x7f0000000040)="b49dac4d60a4ef29274ad6ec771da74d4c8c0fd83d0aeb9dfe8d26d06f9ce49c698d1b2dbca3915be594ab841c00c1928a4c4363e8b5d17b93829abe5cc366527d477458cd9ea1659ccb0c07b05158926dc748aefe7fbac30da85ef1aff793e0b99ff36257ec864a364666b2151e04e8d9a709e4c300368d5ba364e588c6f32a1829b7dab3a920b39918005984975164f1dedcfd04503f3659ab13386ab13bb7f2f160848a41792cb49eb2ffeca5915ee146e2c348b8fd37ff56d515b39d85fbb034b8e5e08c96b6579a19efe90844e30a790ec223767b8f5a409433b87dec579dc420a62ab203228136577a9945a59afda7efbe7dde89be13a4fdf54ead3624b1a5c70efe99d2e36cb5c1001427e116128ce2601e552e2f50642e1dba23f9526788a382acb79f92fd733f10bdcf9a4035122730498ca3481a4d2e7ae2c6bddd8cb1f8899f0f5d36975b1f1aa989a182029e3225d98a503e97073a728978795a6f27457df5fe5d19182d314ba3b15ad8a430e808bcda1bccfc1e6ea459aab7c062fb5d92b54098e73706e4d2b90b3985ce35736e715b04b7e590fa7349b639069928f5153a8d31e443a21546029d24943a69e44f65a4e253b538b8803366e45cdffa3c45c4c012981eb018dc04af8c082acaa2eff05cb91025d6424a2d9600ac2539deef60540c030f90c8adc939d3b68303babc5da851c88b15e238eb0515c1210398235eb76dba454f4e1f460d3cd13d85cc5a531968955064dae07b2c63d17d343fb80494fcce2f84775be037fd6d0e05834da6408fce3dc3c8d45bf33f500d83c4f5ce015544464daa6b4965987a6454eb6e29bf33d654ab0c6682cbbb5520afeb93c9c9b377886ef4b18f31c6c6e67fff8388e19c5ba73efda0e7874dc365c05da5ac4d360161a8a28aa41a3d410da55e464886bc2c69d2c441961f5eb7cd428e410d12c0a083f197abc4a249a7cee1d6f495f4513f8ca194c43e967ddd342ecb3c3cecef199a1beace9e374ebe04e9dfff46fa4d6e3b8296a1a3f8daa431e1a45598100f53292630c7e27efbf18a0236e33fa08b66af22c8325e04f1c6a7e48844fdd6d506663d15cfdd7c63f60803c6e0db5cef6d111a6401bccc09ec05f56bace7eacfbf2f6954c4e9c00f6e4b8cf6622f7f23570fc4b730d8cd8f9b322eabadffa9631105943e621a018d3cf7fc4242d457c068e94342f819808a491f2c62ea07fba2a08b15ab208d64e1cde2020f0ae9f8f903dcb78bafd589832e4e8a02ac3a27749acee64fba7081c782d68df0a7314f95514e9022845062f8caf3052ffe55a5bc538789ec6eca37a26055ff6768cb5d5158a8be6321233efdb9ce135fccf3f20e696dbeec3776986e210d1c36881657b9eb3708c0437955f8b1ba617b9440dee185741f554facdc7da8daa55015b19dc8f5e32499797e1b85941a801771717184ce09c569cacf82992b2f8c33ede8e02a8a3e1fbd4835c2d943f144498aa1edf50d1d2af669b148d433b7a7e55b08a0fbf49572b4815ae96d35a8297b57b73c8333e7d7718169ac83129eecea08565172ee26e8cc29682fc2af3f8f1fa1fa411205b8376389065551c7cfeb87b6036f91f20566f6274df1365f0122bb139ee6c6298854126ce7d2f69233d1d7a8edcaf6b15d97baaaaeafd1d4069b83bea98694dc9ef8fb6beae9dc5d1ede7458bd897b11fe93ef6f39858dc210419331327fd5dac69c6ff352ea04e439ce97e7ae904db5a86a5ed4bec35cd4e16bc1e3d54c591a93da93645eb697065719a7c41bfc0a4b573961eb6f0a944aea68e65e28713d4ce77bcf5e1cc8c25375f075ac66c2b1116f9d89a7614c60cd5afbc86fdad7fac5a629ed5b808480dd2cd0448f85ef5c308fb4cb9a618bed06155cfd7e6c18f4241bc63a960056dbf837de0dbbf742a23256af0acf855a722bf62630e8df2c57d92b3cb7aaa6cc26cbb20670fafee9af231f1d391f560f4257bd3a976ea0d22fc05c592adfe85c5f81ebf885eb93c23dd6ed3f84eb39766d28c1de7fd75250dcfc469cfad4ac3cecc8d148de571669ef7d1fe1f115ec7533fb8a81c87b54602a0726a9b47dd31b536fc51f4b7d292b83b191524c457dd618536e3b7ad20e222a1c9bf725ce5586463262d112f3336b9da8abb55fe4c756808a0b8617fd69d84f91d444a72c22116be94a52d1b2bb79cae60a4b9cf39d19819167d4bd689e752b1ecf5cd50040cf6e6eea9183dfc793c332ce666c66b179363b3fa865846d6774d050e5037776df34c7ab6672ae1b350dd849b540b8b285eaf6794f507e2c7c10acf7bf7ad9521e86a1b72a1418c90ab696f8e984ed6c7f0f1810fc9011e8f3f6cd4d1687ea682a42faffa296a00f9cc7c53210db85c9559207aa664f3c2716f524ad5fc0ed97bd880c14ace7d629c43f02e413c562046eeb2b58f0ead899286ea8b5826af76b3e3d02e71137cbebe552306d1dd1016e097a54928033ea63f0540b05a3bf331773694b2546e4679c05d488b356a2d37d275b4ccee590bb7ac7483ca0a72149cea3b39f687fded587e5e94d25c3b3e1b1b0d47b2bde830331905390837a3fefdda1feba5ffebb6cf9ec2ad3882e3094c39da450daac74592c20ae560c61cbcb1d487f8d54e76c808178b2cc0ab1d072820b291f3092c49b205981f45318135c8075d860e7eefbb640dd8e80498167db26b3ece09c8099eff0c3cf5cbfe6440472ed5d3ce8548b15004423868118b0e87655315856fc2102b804c1e28b6da7dcdba54ecae6b721477fca494bd3b72f46480ae87f334494c602819d6a3e9cbd6c2f2b3a9d32c8eae8c1484b8dc5eea2c006ad95fa3d7e80d89e1aa6ae8a98fcb95ecd8a654d176cc7ada0c6424a898ff8a9859b2a29a51d36b879011a24ae2a4e0f69f02c95ac3da7246a45681d91bc0ebf019e3e2031fb3336b2ec10f7d94186fc0520999b775bac2eb542e5e3cfbfbcc5f5f56df6d9d0ca8e2308c6cbaefc861a0acf0622ec9fa7c4cac8590c6b4692219ed38b136ce5d659d00a8710b1ddf384edf5737101448015ea3f8c3fef0cb07602064035a4579a2b6bfc5edc1e81a8ab9522d6711f606183b3f59ec32c4df67da8289ce54ed9f97c7515746b859027c8ab1ddcbec9ab3c37c6dc0a97cb255961c69bc71936364ec49076400b4d918265cfaba0a9475d5dd1b8179653c3892aeb3f434a9ac2d9d3dbb81598a44cd90beb48a8949dd31f907465dfd8bf651dc60ef0d535cc9d5493dda2c474a10b7726b63291b0d6107b47faf77ce62eea112fbdfb95481c7f84d6cf9312faea8888284cfc04854f58fce61988e25a2dc9dbaf8f7659b21ace61be9d83c04789a202d72f2e411ab3db416dcf63abbc1c6d827aad75579eecb3a81a6e0d0669012db51435d53d3c0bf2426dbb431e2f4bc4938fa458692051f5047610687bc2077edee95274bd653ef1105086563f858c0db245454fe5fc83e6788c82310a02008ef76a3ffd357d8452cf5a712f421f253cf81232f1ca4d6f8364cb96b9bb2f23ab27a6ce3a2682b615ed762c5210bfc17229b83551af482143374e21a115d77c98c8fb17487b2c9f25d8da17ff3e96b155354f611da1ff91bfd6d89436486a466642f96c5cbbbcedeee4d35d26ee982bd152bf309eb1c4f56bd77dc89861bb82eb56405af798c7100768461f1b21d8e98d498ed5e2f74104d0309b00bf376cf32a82397d1adca7ca86950f5e63c60bd0189bac5dcb596ac0d111f9e6d17ed06a47bb92e49ea2ec4a295acd84b37524fbbd3c7516d4135d6d0625781fa85316098c4a8f94c4f10497ada2332b524e5e91c63ba4df47e1b33e0ebb9f748f1d0a338993cd4f155608e8238d3d574b8b166d46211909128848e96a28282b7cb652d103307c0028653b4fb40379b6d545fb1e2f5b9ea9d57a5b270db019e228b1edde431bd44e5728216781a5ba01163fd803c98e46a440e5617a781f55502c73aa1c417cfdfa7ef95fc83cfc0d9e7140df356cc0a3bf0824d4f3233ca963d367f9e6bd53b78edc3f5b0c53c9a836e02eb57e0c16417f9e42338aeada43fe34b4c525c29835d77e7695142361a500e097f7d110f1912a534fa9441f1973b123d76817483f51deef7b2f7b3b180cf062b7acba7b77788d7ae6be6cce00296097ae60ad84e588711a5a63b66aed3e01010381dfe53bda06ee770cd3f96b3fb7ec9ad700949e6ad32062958e807b8b03204c3a77b5034143a6b342bb913dfc4a5db48c44235b56ee464e933b2bd7e46b49ab01166492d4d46a48b730532b13e231984c7afd18ae13663ab84c385e367021ad3dfa918860243454ac08e00e27ec0b193f67fa33cd43d9bd69b34ad29184924c98e61454afefc5cc69e32d0792c65aa0638b5d128fdee58bdaeae09969fc7493296c0f7689093e24609f44a1ea4a4cb7aa4e46082e89fbcae15862c5f1423ccb651a5b12f952797d0854ebfcf6dc494add5d140a5deee50f3c9729f2779aa0bb0339690770f090b3719a53b9e0a4521d3a9811bcd1702642f7c5d4142044826f97edf713bb6a695b5f34e705eefd386df6ab621fb8f4e39641926908c9a37df76bef9b40fb352c8b2266a079625c81f0b6823348e7dfe593fd6c155fc2b0c082b04a8098712b02a4bb9a60a4f68cbf60b58d61f32798c66edf64d387576630902a13b5a1f8f5f51a79c708d7d2c68d2be1c68e5fd8757a3c3ccf951e65727722868688133ce0cc22a7d3db72fd0d3ae51d9c169055d754011b1de6a6be680c55bf92e6fcbc645941b2e89afa7b91ad61726c87e3c4e131608ce244fea2018925ba74dd8cac6e6f5c455978b7151eb6e6e7a797590d784cd9464450f89a671c29804a1eb7c96612ba025fd4d2564a0199a1c18c20964c2fecc0c463fcf0c1d3089b46ae523b6fb259cce1a0171254f847945a862ceb3ae1fe7082958d06ce952f3f079df3490b448cda60acddbaed7ae50eb32eed1feab15b075e7fac69c4142dab32f9abd562c6a3ffcc6d3fc0595e202f3430080a8c85665a25889fa5f78f403f9ac93cfa160fd242d9cf123e43010891bfb21f7518588aa9f94393daf212c7a396cacfa27f29bdf8ce0c9269c3270466aae1df02832e605dd3aabfde47b93cfc812feb65bf1f0b326f48fa89f00059d0bcab950f229f128eea46d16a9d3feb5c04f5e707f835de686df16ceaae1ac98cb5a7abd0852cd80f8f2c093ca86b4fcdfcfb4648d244b5bc734ce2bff50880b4de5cad96aff1eca5e3f93dcecbde2a06b0371529d471908c05654e463c787e56e24edcdc363a93b22fa7925a4f205e62ef6d7f7f64b7e9525b7e8c694f19738d5311fbe12f1b8123035323775ba87e1e77bb9eddb09e22f65a175f51bce14ec6bceeca1bfba4bd69d4e18d8442c8f60d2747ec7373e8e49b412b769dde6486eca1b98231ddac5164ace299b5bc3b6cf82c00ae55a1a47af14a72a5f05ce3c94ad6dfb2af8712262a0d2dce422946d84fe643530a0528eafabe77a14c9da86eca5d7556832b2dff7005f79f61ba802b6c855ea38a043d18d996c21897f7c820b1ea63cd5d823472adfd8b56df62d0a14f15c9e48b9ca7d4aceb012a16778d88ebe48c501e9dd460d3a21a0a4bcc4bc7129bda1912eff1a3d43fdd75cf7c2c50978d94fbf65b34fe73d003084fcf6133f2336ffa4a4b927fed2ae7bcb010360805fa497af8cb3dfd2ea25419f3857899bdab0bdb61f3ea1de0f7d6aa6e79f5493fab9751eb0d9c5fe65956072125c47d94740f2a3ace54719f09ebf3e27d895d51935332265571596731ce3144330ea48d8995038d756986b111899c8630a24920261a1f9907fdabd970f7dc69058a8efc4381b414752b12f3974ded28a149760c7fef4ddf906b8210825d1c98dcdee80994996f6d93fe70d93f1736e362c311ea1fa5d3bce28fe7111c2a9d0baabf6bb77cb9054c863f2d40b990f41a0766364b31f4cb68bafc05a757abe072d0b389091acff20f3ea906d01b9ca4fb4b251b84be30e93cb1017f8e1b1147120f6788548c759cf42c399c6b643deeb66a98eee89ade40807724a20e1a3423641b5f03040141914d3d81f9717a85140ffc052c781a59a18a52e08f00da35a5754ca4fbfa77c0c1265d9e28644ba89d4499a31b60e350eb61746207a0469b464303dbec4dbe13d297de3fc3b4e6bd37805ea69bf79e847633a8f68d8a053401f82fafc0c1dbc5f7d03729f306f0d048994b53504aeb0e9dafb93394e6d425d22ff198579d546c63c0c77407c43bf9fbfbfd62ca8ac316ca4d19e4631cbbd9cdf21ecfbb3eeacdfa75605d803184c11bf9ee730942962e688c5fad693a6e88d21d61dfb51d8e84ffd090a5f7c31ef8f77e1a3e7ee77f71531e3d3e190bc8bbfd5e7b7792a0e361baee79c9db165033a83fda2f1972b75ad31e299ee69af7bb1c50870be803b17f6adc284bd18d10444b9af2a44be791515f2f229d549b8b85d0e05ac418c852336b917cf6b5cfde4f0521a19d3b888f72b71c5a5b11f96853f76da5cd954ecfde9b37f4f44f3187034c57790be1e696390b9ed4db6d660900b1bdf9f76447037ae56a1ec6d78560afa27aed79ef6b592db1056d97104d5cf8f16e5dc9e8c45915486c042ad29ed2c8084fb94ee9ebf18792b83220dc38936c3d2e09a1ef541fc7a440b6a09674a55860fd9467efe1580a6601720b36f128612b20837c6f4cc44a43deb771a001885bb17467d7ec350ddc648f3c13b579213b91d35c02481280efdee61cae175923fe9e5f218d61b146b71d9d7d60dbf740ee3f53fe46e7952cbca6595e56dfa02aecd623b6be5719f87de725d92df32838f1b61b4e7bae65b683619ed8286e70382f801c067548d6f83c701c92fb84d0e0256114882af2c8fe2b67e456f7c77637560a610693b6302d0f5e135c96036966aa6868b77d10e2272887491fce4ce8ec1e0fadc326707fa1a404384baec7b1ba0524ee1ee873662c1a530ea396fc5cf5fa3c8b117829cb144577bdce0cca193a180b0b2ba69ca8fec31e70f7f6d5c2f5cbed9bd5a5998ed896fb4d36432b9ba8a6508fac5529cba750da9e2b651127c171fa7774d4c75b5b62fafc114c5775275a8ba20cbe7441a1709c2b0444d7f1721c368f35278c0b294b3cf8cb7fcad332f0f1bd2fdccb4c7e402c7efbfe9260b95ce72dc608897dc394490607c8914d29fd0a37036cdbe69df74e78cad0f11221306fd15acd622dd709a19fdcf134362c358efc9398162c38b32ae9e66d12d50489643c43a19da922d14a90823d92bf3253a6916dee9768af98ec865f2d2413aeb1455a0d0264a1348757aae7746ad86ebdfe6971dbeec2b30c36802e007da501bf27c68c5a7103d9e437ae6ab58f64b0ed67e258a7c7f9a9b07d6bb11e3e511d877085170d8c33e20a62f34c655cdfc2a788e69339ed9c08be0019130ec88b4a4b213d089bc182f8595547353bb36e1fb2df57405c982070a4ceff26e4ea2a3d2faa45f1f6a965c83429ce4ce07c6b5db04a4031247babd3cd8cf39861f4702c92ee85251ebe2456e4429f8a32efbac1e57c6baa36554dc807720077a8ca19a9db71710eefa1e9c202878b8b7f4a124db303ccd3a2dfe31cb5f0be4df44b5835cb95b089f5495ce35651a84ac26552c343dc63832a36c966c92ba8e661ddbdab08775d50c40790e597bdc5b452565d0465cd93d6bb681b2596252fd255e4a208eed904dfbab2e6fe2329c97cb3d15ecb63d3aff17e226f2e40c88ea6322bc6bbba4f6d750ed7f50908c3da02b488a4fc86f0270227efb45b6a5382c1994d49fdd2544c0d03b88e410df64cc8dabf83916f3a40257c02ee9d4b67f8dd3297a9695c3ef0d06a28ad66e7619c0bf0fd2542483bd6ec8e9fa829e7a521c522a372364b4c20e5cab35442ce1ac237c3fd9d177c229bdb4ee9667430826e059c59b12bbb5cf03f7291b0afc303844dcf1df174357d7e2f833043595f2d8bcf6271bd2e844238f460ae082f59aebb3d18c87d482688c58c5541c79d10d44ca6dc221faeab71a36c0ca8fa5eb0cf337077ae07bab63f7aca2caaa588340f8796a564788d7436276039e83f25c6350121078c10d896eb6d9cf6e7ea99290c10378e88d5e7309f17518e28454078d04bcb38f15d1a624cbc5c0d76878df58f220726c8365641e1fd2a66c9233bb3143af62c72bb06fc3cc627fbc66c9b778e254c4680c143b936c69a977fdbd3d5fad2ad6d974796d03e96989863d25de45313bb49421bacb3cd52e379c182b28bdbf18622026619be32e059d267bb1342e82dd0004f48e838ba2a379cd1e2bfc978b8aab80b79630a139c4dd89302547dc3296838f95a211f7a6f74d9e9f977832e7a6c4ccdce6e79c4735d700c81d6bb4c461d3288f41037abade34b7998ace0af5e7f9697e3042b171cdac15a7102d1b146db3b9057e22c274e6a6c156da70fbf9a07e0b7b22b90400aca401c05f5168483f66d5fc1db6e467463e652ca41ead157d29ce14c19c493234287578e0e6c21b6aca9f59d6591c09c81649d5c1e4bbb43745e83cd9c23d5b7b89282efba61e3e790dbce7aaae0011c81dae1c48fc4400bd81da44b2417682744bd06ae1b9c63a004d9345202e63eb4a4708e6ddb20be55c843ee3fa19ddacf7ad32d1ded060d9d8bd46bff386e0a3fe41260a8c90a8eb0089083e1b72278e0a7398b6e2e27c256ea426174d6a1fbe4067685f9eab61ace1a655d3ca35cf566fd148112e11ca4f39d0fbe041aba99a93e11c44e4ff07e830a1acfa07132d287fbcdb2eec82e4fb4fad847c29f601c578e6c7109340555fe16f82c058d60aa57d19baf6eccd59b8ffd789de6376dc9356b78494c5ea3d1c29bbeeba8e76433d3caa4adf0db15377834f5bb90f88b680030eb82b63fdd9f922d5bb0b054a50c8ac82c0925902dd351e8d9d61141d34b36f0a1d2a7b6ef6205ee1ceba611964e9e1268c57bed9cd5779961f38c7bc42fb9b66c024727ebe72ed9f6374eb44fa29ac11bf71c626a43c67b917a4481829a053644f769d0024104c9fd8f72fc7534323aab7efb20572fdbf68851c79aeee8137726145e4328fa70c38189f8441438947d6a49e3dd3dc9d5185d810aef71afd8dfbf96339ebd0e2a159f75d8c34c9bf37ca5cdd791371e4e835962b7d92b5d0924ccb071b841e7f96d0552abcdb3fc692f216fc687f72fbd360f534e2407d01f30ba9427f2fa5007dccb470c60fce127a577f91f07aaeaeff524b3fcdf9311171f443c5ad08b74b60ecbd8bfbb0fa842c18a5c8487f0247acdbeb393dc7cf376f6de99b13550c8db32a08c02a182dd15760da6c8bd3a7259724edec715570bb18465330666bff3a7147bae0d500fe771410efb11b348a194d7ce949ce074ec24fe5f7716671eadb824ff5e1668563a5398b833b1cab4e9d2df97783a52fd91b60d1c700ec0616208f99ad376de03162dd07dd42ffaa7c9ff4af7d48cde82c9dd040bd724039c929219079d23a7ca412aab0ddc833aac5a9afe819fae46cd8c8006dc4e96e11fe826df073b1ad9520b0442af3bde997b97d23bcf7a284dc89703e0b41b597ca1c64a23944cf6ab196ecd33b1cd764396eded169bcf62c7faa61f386ac64910852ef409c4c4242eb11a0ce5d13205891f3f6e8f568d781fd86636af3393918adcee3a73fac340f486d3c511a64c6587d0e0c553d087c3706bb8498dc059056b8ffd0d185df6c52b482bd55532550e1c9d37279ccf0ba698b90f185dc4c43aee53dacff0887fe842ecffb450088407443b709691fb10a0e9b38c2af46a60e9a02db9d8efcb3e89d0a19ba31ed3975eaa95d8fc6aed274f3a0ebccc99d3c5fb1be2d5a2f5814688db811d8af8fc273a8e47620e23934c1ea642cf776fa276ad857cb023988ef39b237b00c72b85d86fed11fbdeaf31ced509c265822b560e7f0793eb5b53c5c7be3d855d146e2997f9974677dfa7a49f3c685bf24e8dcae56b3352c1038d8c7392af62ed22c5054157812ed4b14e6c7b9e4f8cba3fd99543ea467019ad74e2025b2b1c4195581ef91a88b999254ac748e748c4a50480a6cd3f5fd7d8bc919bdce7bf51925dc187df60af567ef232cb59d5583d604a19928c2699905446f2758fc7b640fbd0a2b9494b2b13cea88da31c5b643663f592e9b9ba580c16a2371dfdb223c7c803389b331e2412862c26cd9254e1db521f0851d9fe1d9d3cec7718fa785e227ffeda95e47944bd76ad718a1e27fdc7a51bea6f128f593cff860e82a9b474e0956830a35340616f4e8fbb10c94698b4310ee49b9398d0501af47982a7add81fef360000860926d09236532751a54107636120454e9b952ad891e645c74606a9869e5b6a4e20178decbf842d3aba603be3f4ff149b23776954b4b9898755f05b91fcee1574dc66de42547cee0b4f23c320224a737776bc4f04bcb996bef301b692d8b64d1d11d54dc5ef5760fda3e16571cac787d1f0e6de4a88b7f6ef8886734a451c4c971545d9ffe39d80eeaf9f1c77ab04364ad3281a4d41a250e410e138fc759cdf4e408a38d26a72bee16e1ddb1055c391a09fae22209e891d93be9c0cb1a8515e5ff69b55c5d9bc83bf92316d3addc29ed0d3cc1f8c6635f0726014fc7cf5236e7757cac9d391821f239267010b89d3918df46e1b483e5afa999ae7345850e17cf93d2da84d84e1a0d2e2bb95eb890ac6fca1ebfb7387b799b02c275f915b5e4780dde3f738fc2432636d89d620f541d218d39dc7b1854e77c85d3661030e79d7838935724804e5885e901856debc53fd947b6f5239961943ad48d581d88e4ce26c0fbd342fac6159d2264ef4051fa29cb850649650537bcbf3ad06cf8736cf09df521eba9098f532953e1541b223db5f37de855f8673c59fdb1acce1b2cfc9fe50d38dc13f9809ea382cd4863c7c1c691eae1863f563054fa6a0bfb039d1dda3f250f52d155269e3f2ecfbfc57cb3add84f986ef9cb84f4c4b75d32d3edd960ceafbe69ac71a0a524bd79777f20fdfdd29d063e06c0bae7fd247592f05432406afb207f4476e0bed4e7b76395361a1bb7a16db6a95c59ddc1930053b93c51eee3f405f62a9e412d5c1aef4cc7a7ccb70d6d91ae27d477fcee7d84e1e3e519995127a43eb576d9518bd6112686fe7a90c7053d94f67c2878d3537af078ddfbdda06ba4c01888dbe6be9ef4f38e2f8d49a7d023996c516f1289fd9cfd4936089f0bcd0de5156d580b68554e8e9c3874a6b2d56bc736f49217c88859aa63ea1384f3b9dbdc8b30262d0efffddec03fc4241740d23917ef92fc7de3ae3665ff3c9397bf4fcde524cc403f28806317a0059ae93d82e3fb5cc49596c785f46427317d2f3c9d285d239383dfbea914ad84213e1e0121533dc696bee622d0f93ed8eef2e1eb408364adfca90eef0236a85c8d5c4a55dd42793d18b9eaceb8727e1a5bf01b1156ca04452485c6e4a9d40185664eec10c5fecc9dfd48536d5aa70c29eeb43c7a1a2ed73ff66a2604ed959a811e2268980958b2ab1682fb2a5966ab85163a34a9efc27d349625970f1", 0x2000, &(0x7f00000048c0)={0x0, &(0x7f00000020c0)={0x18}, 0x0, 0x0, 0x0, 0x0, 0x0, &(0x7f0000002280)={0x18, 0x0, 0x0, {0x9}}, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0})

3 2

[PATCH v2] ASoC: codecs: add uspport for the TI SRC4392 codec
by Matt Flax 22 Aug '22

22 Aug '22

The src4xxx keyword is used for future capability to integrate other codecs similar to the src4392 to the same code base. Signed-off-by: Matt Flax <flatmax(a)flatmax.com> --- Notes: V2 changes : * Fixed indents in Kconfig * Comment header of source code is now one block * Documented the devicetree binding * sample rate converter MCLK src is non-user settable * hushed spurious dev_info output * src4xxx_remove was empty is removed. * Using switch in hw_params to select port reg. for clarity * Removed interrupt reg. settings - will re-intro. properly later * probe function is now EXPORT_SYMBOL_GPL .../devicetree/bindings/sound/src4xxx.txt | 20 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/src4xxx-i2c.c | 55 ++ sound/soc/codecs/src4xxx.c | 508 ++++++++++++++++++ sound/soc/codecs/src4xxx.h | 115 ++++ 6 files changed, 715 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/src4xxx.txt create mode 100644 sound/soc/codecs/src4xxx-i2c.c create mode 100644 sound/soc/codecs/src4xxx.c create mode 100644 sound/soc/codecs/src4xxx.h diff --git a/Documentation/devicetree/bindings/sound/src4xxx.txt b/Documentation/devicetree/bindings/sound/src4xxx.txt new file mode 100644 index 000000000000..f8d845d42f42 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/src4xxx.txt @@ -0,0 +1,20 @@ +Texas Instruments src4392 DT bindings + +SRC4392 is a digital audio codec that can be connected via +I2C or SPI. Currently, only I2C bus is supported. + +Required properties: + + - compatible: "ti,src4392" + +Required properties on I2C: + + - reg: the I2C address + +Examples: + + src4392@70 { + compatible = "ti,src4392"; + reg = <0x70>; + #sound-dai-cells = <0>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d16b4efb88a7..9c9d74074888 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -205,6 +205,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_SIMPLE_AMPLIFIER imply SND_SOC_SIMPLE_MUX imply SND_SOC_SPDIF + imply SND_SOC_SRC4XXX_I2C imply SND_SOC_SSM2305 imply SND_SOC_SSM2518 imply SND_SOC_SSM2602_SPI @@ -1471,6 +1472,18 @@ config SND_SOC_SIMPLE_MUX config SND_SOC_SPDIF tristate "S/PDIF CODEC" +config SND_SOC_SRC4XXX_I2C + tristate + depends on I2C + select SND_SOC_SRC4XXX + +config SND_SOC_SRC4XXX + tristate "Texas Instruments SRC4XXX DIR/DIT and SRC codecs" + help + Enable support for the TI SRC4XXX family of codecs. These include the + scr4392 which has digital receivers, transmitters, and + a sample rate converter, including numerous ports. + config SND_SOC_SSM2305 tristate "Analog Devices SSM2305 Class-D Amplifier" help diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 92fd441d426a..a599b727c65b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -231,6 +231,8 @@ snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o snd-soc-si476x-objs := si476x.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-src4xxx-objs := src4xxx.o +snd-soc-src4xxx-i2c-objs := src4xxx-i2c.o snd-soc-ssm2305-objs := ssm2305.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o @@ -579,6 +581,8 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SRC4XXX) += snd-soc-src4xxx.o +obj-$(CONFIG_SND_SOC_SRC4XXX_I2C) += snd-soc-src4xxx-i2c.o obj-$(CONFIG_SND_SOC_SSM2305) += snd-soc-ssm2305.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/src4xxx-i2c.c b/sound/soc/codecs/src4xxx-i2c.c new file mode 100644 index 000000000000..a42d1b463dbf --- /dev/null +++ b/sound/soc/codecs/src4xxx-i2c.c @@ -0,0 +1,55 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * Driver for SRC4XXX codecs + * + * Copyright 2021-2022 Deqx Pty Ltd + * Author: Matt Flax <flatmax(a)flatmax.com> + */ + +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "src4xxx.h" + +static int src4xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + return src4xxx_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &src4xxx_regmap_config), NULL); +} + +static int src4xxx_i2c_remove(struct i2c_client *i2c) +{ + src4xxx_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id src4xxx_i2c_ids[] = { + { "src4392", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, src4xxx_i2c_ids); + +static const struct of_device_id src4xxx_of_match[] = { + { .compatible = "ti,src4392", }, + { } +}; +MODULE_DEVICE_TABLE(of, src4xxx_of_match); + + +static struct i2c_driver src4xxx_i2c_driver = { + .driver = { + .name = "src4xxx", + .of_match_table = of_match_ptr(src4xxx_of_match), + }, + .probe = src4xxx_i2c_probe, + .remove = src4xxx_i2c_remove, + .id_table = src4xxx_i2c_ids, +}; +module_i2c_driver(src4xxx_i2c_driver); + +MODULE_DESCRIPTION("ASoC SRC4392 CODEC I2C driver"); +MODULE_AUTHOR("Matt Flax <flatmax(a)flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.c b/sound/soc/codecs/src4xxx.c new file mode 100644 index 000000000000..46f693a59186 --- /dev/null +++ b/sound/soc/codecs/src4xxx.c @@ -0,0 +1,508 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * TI SRC4xxx Audio Codec driver + * + * Copyright 2021-2022 Deqx Pty Ltd + * Author: Matt Flax <flatmax(a)flatmax.com> + */ + +#include <linux/module.h> + +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "src4xxx.h" + +struct src4xxx { + struct regmap *regmap; + bool master[2]; + int mclk_hz; + struct device *dev; +}; + +enum {SRC4XXX_PORTA, SRC4XXX_PORTB}; + +/* SRC attenuation */ +static const DECLARE_TLV_DB_SCALE(src_tlv, -12750, 50, 0); + +static const struct snd_kcontrol_new src4xxx_controls[] = { + SOC_DOUBLE_R_TLV("SRC vol", + SRC4XXX_SCR_CTL_30, SRC4XXX_SCR_CTL_31, 0, 255, 1, src_tlv), +}; + +/* I2S port control */ +static const char * const port_out_src_text[] = { + "loopback", "other_port", "DIR", "SRC" +}; +static SOC_ENUM_SINGLE_DECL(porta_out_src_enum, SRC4XXX_PORTA_CTL_03, 4, + port_out_src_text); +static SOC_ENUM_SINGLE_DECL(portb_out_src_enum, SRC4XXX_PORTB_CTL_05, 4, + port_out_src_text); +static const struct snd_kcontrol_new porta_out_control = + SOC_DAPM_ENUM("Port A source select", porta_out_src_enum); +static const struct snd_kcontrol_new portb_out_control = + SOC_DAPM_ENUM("Port B source select", portb_out_src_enum); + +/* Digital audio transmitter control */ +static const char * const dit_mux_text[] = {"Port A", "Port B", "DIR", "SRC"}; +static SOC_ENUM_SINGLE_DECL(dit_mux_enum, SRC4XXX_TX_CTL_07, 3, dit_mux_text); +static const struct snd_kcontrol_new dit_mux_control = + SOC_DAPM_ENUM("DIT source", dit_mux_enum); + +/* SRC control */ +static const char * const src_in_text[] = {"Port A", "Port B", "DIR"}; +static SOC_ENUM_SINGLE_DECL(src_in_enum, SRC4XXX_SCR_CTL_2D, 0, src_in_text); +static const struct snd_kcontrol_new src_in_control = + SOC_DAPM_ENUM("SRC source select", src_in_enum); + +/* DIR control */ +static const char * const dir_in_text[] = {"Ch 1", "Ch 2", "Ch 3", "Ch 4"}; +static SOC_ENUM_SINGLE_DECL(dir_in_enum, SRC4XXX_RCV_CTL_0D, 0, dir_in_text); +static const struct snd_kcontrol_new dir_in_control = + SOC_DAPM_ENUM("Digital Input", dir_in_enum); + +static const struct snd_soc_dapm_widget src4xxx_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("loopback_A"), + SND_SOC_DAPM_INPUT("other_port_A"), + SND_SOC_DAPM_INPUT("DIR_A"), + SND_SOC_DAPM_INPUT("SRC_A"), + SND_SOC_DAPM_MUX("Port A source", + SND_SOC_NOPM, 0, 0, &porta_out_control), + + SND_SOC_DAPM_INPUT("loopback_B"), + SND_SOC_DAPM_INPUT("other_port_B"), + SND_SOC_DAPM_INPUT("DIR_B"), + SND_SOC_DAPM_INPUT("SRC_B"), + SND_SOC_DAPM_MUX("Port B source", + SND_SOC_NOPM, 0, 0, &portb_out_control), + + SND_SOC_DAPM_INPUT("Port_A"), + SND_SOC_DAPM_INPUT("Port_B"), + SND_SOC_DAPM_INPUT("DIR_"), + + /* Digital audio receivers and transmitters */ + SND_SOC_DAPM_OUTPUT("DIR_OUT"), + SND_SOC_DAPM_OUTPUT("SRC_OUT"), + SND_SOC_DAPM_MUX("DIT Out Src", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIT_SHIFT, 1, &dit_mux_control), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF_A_RX", "Playback A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_A_TX", "Capture A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_IN("AIF_B_RX", "Playback B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_B_TX", "Capture B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + + SND_SOC_DAPM_MUX("SRC source", SND_SOC_NOPM, 0, 0, &src_in_control), + + SND_SOC_DAPM_INPUT("MCLK"), + SND_SOC_DAPM_INPUT("RXMCLKI"), + SND_SOC_DAPM_INPUT("RXMCLKO"), + + SND_SOC_DAPM_INPUT("RX1"), + SND_SOC_DAPM_INPUT("RX2"), + SND_SOC_DAPM_INPUT("RX3"), + SND_SOC_DAPM_INPUT("RX4"), + SND_SOC_DAPM_MUX("Digital Input", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIR_SHIFT, 1, &dir_in_control), +}; + +static const struct snd_soc_dapm_route src4xxx_audio_routes[] = { + /* I2S Input to Output Routing */ + {"Port A source", "loopback", "loopback_A"}, + {"Port A source", "other_port", "other_port_A"}, + {"Port A source", "DIR", "DIR_A"}, + {"Port A source", "SRC", "SRC_A"}, + {"Port B source", "loopback", "loopback_B"}, + {"Port B source", "other_port", "other_port_B"}, + {"Port B source", "DIR", "DIR_B"}, + {"Port B source", "SRC", "SRC_B"}, + /* DIT muxing */ + {"DIT Out Src", "Port A", "Capture A"}, + {"DIT Out Src", "Port B", "Capture B"}, + {"DIT Out Src", "DIR", "DIR_OUT"}, + {"DIT Out Src", "SRC", "SRC_OUT"}, + + /* SRC input selection */ + {"SRC source", "Port A", "Port_A"}, + {"SRC source", "Port B", "Port_B"}, + {"SRC source", "DIR", "DIR_"}, + /* SRC mclk selection */ + {"SRC mclk source", "Master (MCLK)", "MCLK"}, + {"SRC mclk source", "Master (RXCLKI)", "RXMCLKI"}, + {"SRC mclk source", "Recovered receiver clk", "RXMCLKO"}, + /* DIR input selection */ + {"Digital Input", "Ch 1", "RX1"}, + {"Digital Input", "Ch 2", "RX2"}, + {"Digital Input", "Ch 3", "RX3"}, + {"Digital Input", "Ch 4", "RX4"}, +}; + + +static const struct snd_soc_component_driver src4xxx_driver = { + .controls = src4xxx_controls, + .num_controls = ARRAY_SIZE(src4xxx_controls), + + .dapm_widgets = src4xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(src4xxx_dapm_widgets), + .dapm_routes = src4xxx_audio_routes, + .num_dapm_routes = ARRAY_SIZE(src4xxx_audio_routes), +}; + +static int src4xxx_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int ctrl; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl = SRC4XXX_BUS_MASTER; + src4xxx->master[dai->id] = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl = 0; + src4xxx->master[dai->id] = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl |= SRC4XXX_BUS_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl |= SRC4XXX_BUS_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl |= SRC4XXX_BUS_RIGHT_J_24; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + regmap_update_bits(src4xxx->regmap, SRC4XXX_BUS_FMT(dai->id), + SRC4XXX_BUS_FMT_MS_MASK, ctrl); + + return 0; +} + +static int src4xxx_set_mclk_hz(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + + dev_info(component->dev, "changing mclk rate from %d to %d Hz\n", + src4xxx->mclk_hz, freq); + src4xxx->mclk_hz = freq; + + return 0; +} + +static int src4xxx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int mclk_div; + int val, pj, jd, d; + int reg; + int ret; + + switch (dai->id) { + case SRC4XXX_PORTB: + reg = SRC4XXX_PORTB_CTL_06; + break; + default: + reg = SRC4XXX_PORTA_CTL_04; + } + + if (src4xxx->master[dai->id]) { + mclk_div = src4xxx->mclk_hz/params_rate(params); + if (src4xxx->mclk_hz != mclk_div*params_rate(params)) { + dev_err(component->dev, + "mclk %d / rate %d has a remainder.\n", + src4xxx->mclk_hz, params_rate(params)); + return -EINVAL; + } + + val = ((int)mclk_div - 128) / 128; + if ((val < 0) | (val > 3)) { + dev_err(component->dev, + "div register setting %d is out of range\n", + val); + dev_err(component->dev, + "unsupported sample rate %d Hz for the master clock of %d Hz\n", + params_rate(params), src4xxx->mclk_hz); + return -EINVAL; + } + + /* set the TX DIV */ + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + /* set the PLL for the digital receiver */ + switch (src4xxx->mclk_hz) { + case 24576000: + pj = 0x22; + jd = 0x00; + d = 0x00; + break; + case 22579200: + pj = 0x22; + jd = 0x1b; + d = 0xa3; + break; + default: + /* don't error out here, + * other parts of the chip are still functional + */ + dev_info(component->dev, + "Couldn't set the RCV PLL as this master clock rate is unknown\n"); + } + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_0F); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_10); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_11); + + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret < 0) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + return regmap_update_bits(src4xxx->regmap, reg, + SRC4XXX_MCLK_DIV_MASK, val); + } else + dev_info(dai->dev, "not setting up MCLK as not master\n"); + + return 0; +}; + +static const struct snd_soc_dai_ops src4xxx_dai_ops = { + .hw_params = src4xxx_hw_params, + .set_sysclk = src4xxx_set_mclk_hz, + .set_fmt = src4xxx_set_dai_fmt, +}; + +#define SRC4XXX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) +#define SRC4XXX_RATES (SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000|\ + SNDRV_PCM_RATE_88200|\ + SNDRV_PCM_RATE_96000|\ + SNDRV_PCM_RATE_176400|\ + SNDRV_PCM_RATE_192000) + +static struct snd_soc_dai_driver src4xxx_dai_driver[] = { + { + .id = SRC4XXX_PORTA, + .name = "src4xxx-portA", + .playback = { + .stream_name = "Playback A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, + { + .id = SRC4XXX_PORTB, + .name = "src4xxx-portB", + .playback = { + .stream_name = "Playback B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, +}; + +static const struct reg_default src4xxx_reg_defaults[] = { + { SRC4XXX_PWR_RST_01, 0x00 }, /* all powered down intially */ + { SRC4XXX_PORTA_CTL_03, 0x00 }, + { SRC4XXX_PORTA_CTL_04, 0x00 }, + { SRC4XXX_PORTB_CTL_05, 0x00 }, + { SRC4XXX_PORTB_CTL_06, 0x00 }, + { SRC4XXX_TX_CTL_07, 0x00 }, + { SRC4XXX_TX_CTL_08, 0x00 }, + { SRC4XXX_TX_CTL_09, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MSK_0B, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MODE_0C, 0x00 }, + { SRC4XXX_RCV_CTL_0D, 0x00 }, + { SRC4XXX_RCV_CTL_0E, 0x00 }, + { SRC4XXX_RCV_PLL_0F, 0x00 }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_10, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_11, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RVC_IRQ_MSK_16, 0x00 }, + { SRC4XXX_RVC_IRQ_MSK_17, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_18, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_19, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_1A, 0x00 }, + { SRC4XXX_GPIO_1_1B, 0x00 }, + { SRC4XXX_GPIO_2_1C, 0x00 }, + { SRC4XXX_GPIO_3_1D, 0x00 }, + { SRC4XXX_GPIO_4_1E, 0x00 }, + { SRC4XXX_SCR_CTL_2D, 0x00 }, + { SRC4XXX_SCR_CTL_2E, 0x00 }, + { SRC4XXX_SCR_CTL_2F, 0x00 }, + { SRC4XXX_SCR_CTL_30, 0x00 }, + { SRC4XXX_SCR_CTL_31, 0x00 }, +}; + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)) +{ + struct src4xxx *src4xxx; + int ret; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + src4xxx = devm_kzalloc(dev, sizeof(*src4xxx), GFP_KERNEL); + if (!src4xxx) + return -ENOMEM; + + src4xxx->regmap = regmap; + src4xxx->dev = dev; + src4xxx->mclk_hz = 0; /* mclk has not been configured yet */ + dev_set_drvdata(dev, src4xxx); + + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_RESET); + if (ret < 0) + dev_err(dev, "Failed to issue reset: %d\n", ret); + usleep_range(1, 500); /* sleep for more then 500 ns */ + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_POWER_DOWN); + if (ret < 0) + dev_err(dev, "Failed to decommission reset: %d\n", ret); + usleep_range(500, 1000); /* sleep for 500 us or more */ + + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_PWR_RST_01, + SRC4XXX_POWER_ENABLE, SRC4XXX_POWER_ENABLE); + if (ret < 0) + dev_err(dev, "Failed to port A and B : %d\n", ret); + + /* set receiver to use master clock (rcv mclk is most likely jittery) */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0D, + SRC4XXX_RXCLK_MCLK, SRC4XXX_RXCLK_MCLK); + if (ret < 0) + dev_err(dev, + "Failed to enable mclk as the PLL1 DIR reference : %d\n", ret); + + /* default to leaving the PLL2 running on loss of lock, divide by 8 */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0E, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL); + if (ret < 0) + dev_err(dev, "Failed to enable mclk rec and div : %d\n", ret); + + ret = devm_snd_soc_register_component(dev, &src4xxx_driver, + src4xxx_dai_driver, ARRAY_SIZE(src4xxx_dai_driver)); + if (ret == 0) + dev_info(dev, "src4392 probe ok %d\n", ret); + return ret; +} +EXPORT_SYMBOL_GPL(src4xxx_probe); + +static bool src4xxx_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SRC4XXX_RES_00: + case SRC4XXX_GLOBAL_ITR_STS_02: + case SRC4XXX_SRC_DIT_STS_0A: + case SRC4XXX_NON_AUDIO_D_12: + case SRC4XXX_RVC_STS_13: + case SRC4XXX_RVC_STS_14: + case SRC4XXX_RVC_STS_15: + case SRC4XXX_SUB_CODE_1F: + case SRC4XXX_SUB_CODE_20: + case SRC4XXX_SUB_CODE_21: + case SRC4XXX_SUB_CODE_22: + case SRC4XXX_SUB_CODE_23: + case SRC4XXX_SUB_CODE_24: + case SRC4XXX_SUB_CODE_25: + case SRC4XXX_SUB_CODE_26: + case SRC4XXX_SUB_CODE_27: + case SRC4XXX_SUB_CODE_28: + case SRC4XXX_PC_PREAMBLE_HI_29: + case SRC4XXX_PC_PREAMBLE_LO_2A: + case SRC4XXX_PD_PREAMBLE_HI_2B: + case SRC4XXX_PC_PREAMBLE_LO_2C: + case SRC4XXX_IO_RATIO_32: + case SRC4XXX_IO_RATIO_33: + return true; + } + + if (reg > SRC4XXX_IO_RATIO_33 && reg < SRC4XXX_PAGE_SEL_7F) + return true; + + return false; +} + +const struct regmap_config src4xxx_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + .max_register = SRC4XXX_IO_RATIO_33, + + .reg_defaults = src4xxx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(src4xxx_reg_defaults), + .volatile_reg = src4xxx_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(src4xxx_regmap_config); + +MODULE_DESCRIPTION("ASoC SRC4XXX CODEC driver"); +MODULE_AUTHOR("Matt Flax <flatmax(a)flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.h b/sound/soc/codecs/src4xxx.h new file mode 100644 index 000000000000..0407db8a8445 --- /dev/null +++ b/sound/soc/codecs/src4xxx.h @@ -0,0 +1,115 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * src4xxx.h -- SRC4XXX ALSA SoC audio driver + * + * Copyright 2021-2022 Deqx Pty Ltd + * Author: Matt R Flax <flatmax(a)flatmax.com> + */ + +#ifndef __SRC4XXX_H__ +#define __SRC4XXX_H__ + +#define SRC4XXX_RES_00 0x00 +#define SRC4XXX_PWR_RST_01 0x01 +#define SRC4XXX_RESET 0x80 +#define SRC4XXX_POWER_DOWN 0x00 +#define SRC4XXX_POWER_ENABLE 0x20 +#define SRC4XXX_ENABLE_SRC 0x1 +#define SRC4XXX_ENABLE_SRC_SHIFT 0 +#define SRC4XXX_ENABLE_DIR 0x2 +#define SRC4XXX_ENABLE_DIR_SHIFT 1 +#define SRC4XXX_ENABLE_DIT 0x4 +#define SRC4XXX_ENABLE_DIT_SHIFT 2 +#define SRC4XXX_ENABLE_PORT_B 0x8 +#define SRC4XXX_ENABLE_PORT_B_SHIFT 3 +#define SRC4XXX_ENABLE_PORT_A 0x10 +#define SRC4XXX_ENABLE_PORT_A_SHIFT 4 + +#define SRC4XXX_PORTA_CTL_03 0x03 +#define SRC4XXX_BUS_MASTER 0x8 +#define SRC4XXX_BUS_LEFT_J 0x0 +#define SRC4XXX_BUS_I2S 0x1 +#define SRC4XXX_BUS_RIGHT_J_16 0x4 +#define SRC4XXX_BUS_RIGHT_J_18 0x5 +#define SRC4XXX_BUS_RIGHT_J_20 0x6 +#define SRC4XXX_BUS_RIGHT_J_24 0x7 +#define SRC4XXX_BUS_FMT_MS_MASK 0xf + +#define SRC4XXX_PORTA_CTL_04 0x04 +#define SRC4XXX_MCLK_DIV_MASK 0x3 + +#define SRC4XXX_BUS_FMT(id) (SRC4XXX_PORTA_CTL_03+2*id) +#define SRC4XXX_BUS_CLK(id) (SRC4XXX_PORTA_CTL_04+2*id) + +#define SRC4XXX_PORTB_CTL_05 0x05 +#define SRC4XXX_PORTB_CTL_06 0x06 + +#define SRC4XXX_TX_CTL_07 0x07 +#define SRC4XXX_TX_MCLK_DIV_MASK 0x60 +#define SRC4XXX_TX_MCLK_DIV_SHIFT 5 + +#define SRC4XXX_TX_CTL_08 0x08 +#define SRC4XXX_TX_CTL_09 0x09 +#define SRC4XXX_SRC_DIT_IRQ_MSK_0B 0x0B +#define SRC4XXX_SRC_BTI_EN 0x01 +#define SRC4XXX_SRC_TSLIP_EN 0x02 +#define SRC4XXX_SRC_DIT_IRQ_MODE_0C 0x0C +#define SRC4XXX_RCV_CTL_0D 0x0D +#define SRC4XXX_RXCLK_RXCKI 0x0 +#define SRC4XXX_RXCLK_MCLK 0x8 +#define SRC4XXX_RCV_CTL_0E 0x0E +#define SRC4XXX_REC_MCLK_EN 0x1 +#define SRC4XXX_PLL2_DIV_0 (0x0<<1) +#define SRC4XXX_PLL2_DIV_2 (0x1<<1) +#define SRC4XXX_PLL2_DIV_4 (0x2<<1) +#define SRC4XXX_PLL2_DIV_8 (0x3<<1) +#define SRC4XXX_PLL2_LOL 0x8 +#define SRC4XXX_RCV_PLL_0F 0x0F +#define SRC4XXX_RCV_PLL_10 0x10 +#define SRC4XXX_RCV_PLL_11 0x11 +#define SRC4XXX_RVC_IRQ_MSK_16 0x16 +#define SRC4XXX_RVC_IRQ_MSK_17 0x17 +#define SRC4XXX_RVC_IRQ_MODE_18 0x18 +#define SRC4XXX_RVC_IRQ_MODE_19 0x19 +#define SRC4XXX_RVC_IRQ_MODE_1A 0x1A +#define SRC4XXX_GPIO_1_1B 0x1B +#define SRC4XXX_GPIO_2_1C 0x1C +#define SRC4XXX_GPIO_3_1D 0x1D +#define SRC4XXX_GPIO_4_1E 0x1E +#define SRC4XXX_SCR_CTL_2D 0x2D +#define SRC4XXX_SCR_CTL_2E 0x2E +#define SRC4XXX_SCR_CTL_2F 0x2F +#define SRC4XXX_SCR_CTL_30 0x30 +#define SRC4XXX_SCR_CTL_31 0x31 +#define SRC4XXX_PAGE_SEL_7F 0x7F + +// read only registers +#define SRC4XXX_GLOBAL_ITR_STS_02 0x02 +#define SRC4XXX_SRC_DIT_STS_0A 0x0A +#define SRC4XXX_NON_AUDIO_D_12 0x12 +#define SRC4XXX_RVC_STS_13 0x13 +#define SRC4XXX_RVC_STS_14 0x14 +#define SRC4XXX_RVC_STS_15 0x15 +#define SRC4XXX_SUB_CODE_1F 0x1F +#define SRC4XXX_SUB_CODE_20 0x20 +#define SRC4XXX_SUB_CODE_21 0x21 +#define SRC4XXX_SUB_CODE_22 0x22 +#define SRC4XXX_SUB_CODE_23 0x23 +#define SRC4XXX_SUB_CODE_24 0x24 +#define SRC4XXX_SUB_CODE_25 0x25 +#define SRC4XXX_SUB_CODE_26 0x26 +#define SRC4XXX_SUB_CODE_27 0x27 +#define SRC4XXX_SUB_CODE_28 0x28 +#define SRC4XXX_PC_PREAMBLE_HI_29 0x29 +#define SRC4XXX_PC_PREAMBLE_LO_2A 0x2A +#define SRC4XXX_PD_PREAMBLE_HI_2B 0x2B +#define SRC4XXX_PC_PREAMBLE_LO_2C 0x2C +#define SRC4XXX_IO_RATIO_32 0x32 +#define SRC4XXX_IO_RATIO_33 0x33 + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)); +void src4xxx_remove(struct device *dev); +extern const struct regmap_config src4xxx_regmap_config; + +#endif /* __SRC4XXX_H__ */ -- 2.34.1

4 12

[PATCH] ASoC: ops: Don't modify the driver's plaform_max when reading state
by Mark Brown 22 Aug '22

22 Aug '22

Currently snd_soc_info_volsw() will set a platform_max based on the limit the control has if one is not already set. This isn't really great, we shouldn't be modifying the passed in driver data especially in a path like this which may not ever be executed or where we may execute other callbacks before this one. Instead make this function leave the data unchanged, and clarify things a bit by referring to max rather than platform_max within the function. platform_max is now applied as a limit after working out the natural maximum value for the control. This means that platform_max is no longer treated as a direct register value for controls were min is non-zero. The put() callbacks already validate on this basis, and there do not appear to be any in tree users that would be affected. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2d39370ddeca..93e72a016b4d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -176,20 +176,21 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; + int max; - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; + max = uinfo->value.integer.max = mc->max - mc->min; + if (mc->platform_max && mc->platform_max < max) + max = mc->platform_max; - if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - mc->min; + uinfo->value.integer.max = max; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw); -- 2.30.2

2 6

build failure of next-20220822 due to 4e6bedd3c396 ("ASoC: codecs: add support for the TI SRC4392 codec")
by Sudip Mukherjee (Codethink) 22 Aug '22

22 Aug '22

Hi All, Not sure if it has been reported, builds of arm64 with clang failed to build next-20220822 with the error: sound/soc/codecs/src4xxx.c:280:3: error: variable 'd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:298:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); ^ sound/soc/codecs/src4xxx.c:223:20: note: initialize the variable 'd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'jd' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:293:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); ^~ sound/soc/codecs/src4xxx.c:223:17: note: initialize the variable 'jd' to silence this warning int val, pj, jd, d; ^ = 0 sound/soc/codecs/src4xxx.c:280:3: error: variable 'pj' is used uninitialized whenever switch default is taken [-Werror,-Wsometimes-uninitialized] default: ^~~~~~~ sound/soc/codecs/src4xxx.c:288:59: note: uninitialized use occurs here ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); ^~ sound/soc/codecs/src4xxx.c:223:13: note: initialize the variable 'pj' to silence this warning int val, pj, jd, d; ^ = 0 3 errors generated. git bisect pointed to 4e6bedd3c396 ("ASoC: codecs: add support for the TI SRC4392 codec"). I will be happy to test any patch or provide any extra log if needed. -- Regards Sudip

2 1

[PATCH] ASoC: soc-pcm.c: summarize related settings at soc_new_pcm()
by Kuninori Morimoto 22 Aug '22

22 Aug '22

soc_new_pcm() sets pcm->no_device_suspend, but it sets other pcm->xxx at the same function with different timing. pcm->no_device_suspend setup timing has no effect. This patch tidyup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ce4a4151e784..4baa24fbf924 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2898,6 +2898,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->nonatomic = rtd->dai_link->nonatomic; pcm->private_data = rtd; + pcm->no_device_suspend = true; if (rtd->dai_link->no_pcm || rtd->dai_link->params) { if (playback) @@ -2952,8 +2953,6 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) ret = snd_soc_pcm_component_new(rtd); if (ret < 0) return ret; - - pcm->no_device_suspend = true; out: dev_dbg(rtd->card->dev, "%s <-> %s mapping ok\n", soc_codec_dai_name(rtd), soc_cpu_dai_name(rtd)); -- 2.25.1

2 1

[PATCH] ASoC: samsung: Use iio_get_channel_type() accessor.
by Jonathan Cameron 22 Aug '22

22 Aug '22

From: Jonathan Cameron <Jonathan.Cameron(a)huawei.com> struct iio_chan_spec is meant to be opaque to IIO consumer drivers which should only use the interfaces in linux/iio/consumer.h. Use the provided accessor function to find get the type of the channel instead of directly reading it form the structure. Signed-off-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com> Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/aries_wm8994.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index e7d52d27132e..0fbbf3b02c09 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -1,7 +1,6 @@ // SPDX-License-Identifier: GPL-2.0+ #include <linux/extcon.h> #include <linux/iio/consumer.h> -#include <linux/iio/iio.h> #include <linux/input-event-codes.h> #include <linux/mfd/wm8994/registers.h> #include <linux/module.h> @@ -543,6 +542,7 @@ static int aries_audio_probe(struct platform_device *pdev) struct aries_wm8994_data *priv; struct snd_soc_dai_link *dai_link; const struct of_device_id *match; + enum iio_chan_type channel_type; int ret, i; if (!np) @@ -594,7 +594,11 @@ static int aries_audio_probe(struct platform_device *pdev) return dev_err_probe(dev, PTR_ERR(priv->adc), "Failed to get ADC channel"); - if (priv->adc->channel->type != IIO_VOLTAGE) + ret = iio_get_channel_type(priv->adc, &channel_type); + if (ret) + return dev_err_probe(dev, ret, + "Failed to get ADC channel type"); + if (channel_type != IIO_VOLTAGE) return -EINVAL; priv->gpio_headset_key = devm_gpiod_get(dev, "headset-key", -- 2.37.2

2 1

Re: [PATCH] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 22 Aug '22

22 Aug '22

On Mon, Aug 22, 2022 at 02:27:26PM +0100, Mark Brown wrote: > On Sat, Aug 20, 2022 at 09:36:53PM -0300, Geraldo Nascimento wrote: > > > + /* > > + * dw-hdmi introduced insert_pcuv bit in version 2.10a. > > + * When set (1'b1), this bit enables the insertion of the PCUV > > + * (Parity, Channel Status, User bit and Validity) bits on the > > + * incoming audio stream (support limited to Linear PCM audio) > > + */ > > + > > + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21) > > + conf2 |= HDMI_AUD_CONF2_INSERT_PCUV; > > So what if we're not handlign linear PCM? Good question, Mark. For HBR and NL-PCM there are different bits that need to be set in aud_conf2 Sample Register, and then insert_pcuv bit needs to be dropped. I don't have a sink handy however to test the other modes. Do you have a suggestion on how to make this the default for L-PCM specifically? Thanks, Geraldo Nascimento

1 0

[PATCH] ASoC: mediatek: mt8186: fix DMIC record noise
by Jiaxin Yu 22 Aug '22

22 Aug '22

When the first DMIC recording is power down, mtkaif_dmic will be reset. This will cause configuration error in the second DMIC recording. So do not reset mtkaif_dmic except in "MTKAIF_DMIC Switch" kcontrol. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c index 266704556f37..094402470dc2 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-adda.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-adda.c @@ -271,9 +271,6 @@ static int mtk_adda_ul_event(struct snd_soc_dapm_widget *w, /* should delayed 1/fs(smallest is 8k) = 125us before afe off */ usleep_range(125, 135); mt8186_afe_gpio_request(afe->dev, false, MT8186_DAI_ADDA, 1); - - /* reset dmic */ - afe_priv->mtkaif_dmic = 0; break; default: break; -- 2.25.1

2 1

[PATCH] Add quirk to enable Avid Mbox 3 support
by connerknoxpublic＠gmail.com 22 Aug '22

22 Aug '22

From: Conner Knox <connerknoxpublic(a)gmail.com> Hi all, I believe these are the right emails to send this patch to, but if I've missed something please let me know. This patch enables the Avid Mbox 3 audio interface to be used through ALSA. It locks the device at a 48khz sample rate. Then it mutes and raises the appropriate channels/inputs to allow for expected functionality. Thanks for your time. --- sound/usb/quirks-table.h | 76 ++++++++++ sound/usb/quirks.c | 311 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 387 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 0ea39565e623..5f43863e5a5f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2978,6 +2978,82 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +/* DIGIDESIGN MBOX 3 */ +{ + USB_DEVICE(0x0dba, 0x5000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "Mbox 3", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x00, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x81, + .attributes = 0x00, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .maxpacksize = 0x009c, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, { /* Tascam US122 MKII - playback-only support */ USB_DEVICE_VENDOR_SPEC(0x0644, 0x8021), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..e7caa7eeb955 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1020,6 +1020,313 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } +static void mbox3_setup_48_24_magic(struct usb_device *dev) +{ + u8 com_buff[4]; + + /* Load 48000Hz rate into buffer */ + com_buff[0] = 0x80; + com_buff[1] = 0xbb; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + + /* Set 48000Hz sample rate */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x0001, &com_buff, 0x0004); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &com_buff, 0x0004); + + /* Set clock source to Internal (as opposed to S/PDIF) */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x01; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x8001, &com_buff, 1); + + /* Mute the hardware loopbacks to start the device in a known state. + * The Mbox 3 is little endian and + * the data written here seems a bit odd. + * 0x8000 (shown in big endian form) is muted + * and volume increases to 0xffff it seems. + * I've yet to observe 0xffff being sent. + * Instead, full volume seems to be 0x0000. + * Per my understanding of 16 bit integers, this is seems strange, but maybe I'm missing something + */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue input 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0110, 0x4001, &com_buff, 2); + /* Analogue input 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0111, 0x4001, &com_buff, 2); + /* Analogue input 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0114, 0x4001, &com_buff, 2); + /* Analogue input 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0115, 0x4001, &com_buff, 2); + /* Analogue input 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0118, 0x4001, &com_buff, 2); + /* Analogue input 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0119, 0x4001, &com_buff, 2); + /* Analogue input 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011c, 0x4001, &com_buff, 2); + /* Analogue input 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011d, 0x4001, &com_buff, 2); + + /* Set software sends to output */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue software return 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4001, &com_buff, 2); + /* Analogue software return 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4001, &com_buff, 2); + /* Analogue software return 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4001, &com_buff, 2); + /* Analogue software return 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4001, &com_buff, 2); + /* Analogue software return 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0108, 0x4001, &com_buff, 2); + /* Analogue software return 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0109, 0x4001, &com_buff, 2); + /* Analogue software return 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010c, 0x4001, &com_buff, 2); + /* Analogue software return 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010d, 0x4001, &com_buff, 2); + + /* Return to muting sends */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue fx return left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0120, 0x4001, &com_buff, 2); + /* Analogue fx return right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0121, 0x4001, &com_buff, 2); + + /* Analogue software input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4201, &com_buff, 2); + /* Analogue software input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4201, &com_buff, 2); + /* Analogue software input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4201, &com_buff, 2); + /* Analogue software input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4201, &com_buff, 2); + /* Analogue input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4201, &com_buff, 2); + /* Analogue input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4201, &com_buff, 2); + /* Analogue input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4201, &com_buff, 2); + /* Analogue input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4201, &com_buff, 2); + + /* Toggle allowing host control */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x02; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + + /* Do not dim fx returns */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0002, 0x2001, &com_buff, 1); + + /* Do not set fx returns to mono */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0001, 0x2001, &com_buff, 1); + + /* Mute the S/PDIF hardware loopback + * same odd volume logic here as above + */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* S/PDIF hardware input 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0112, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0113, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0116, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0117, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011a, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011b, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011e, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011f, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4001, &com_buff, 2); + /* S/PDIF software return 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010a, 0x4001, &com_buff, 2); + /* S/PDIF software return 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010b, 0x4001, &com_buff, 2); + /* S/PDIF software return 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010e, 0x4001, &com_buff, 2); + /* S/PDIF software return 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010f, 0x4001, &com_buff, 2); + /* S/PDIF fx returns left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0122, 0x4001, &com_buff, 2); + /* S/PDIF fx returns right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0123, 0x4001, &com_buff, 2); + + /* Set the dropdown "Effect" to the first option */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0200, 0x4301, &com_buff, 1); + + /* Set the effect duration to 0 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0400, 0x4301, &com_buff, 2); + + /* Set the effect volume and feedback to 0 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* feedback: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0500, 0x4301, &com_buff, 1); + /* volume: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0300, 0x4301, &com_buff, 1); + + /* Set soft button hold duration */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x05; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0005, 0x2001, &com_buff, 1); + + /* Use dim LEDs for button of state */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0004, 0x2001, &com_buff, 1); + + return; +} + +#define MBOX3_DESCRIPTOR_SIZE 464 + +static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) +{ + struct usb_host_config *config = dev->actconfig; + int err; + int descriptor_size; + + descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength); + + if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) { + dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size); + return -ENODEV; + } + + dev_dbg(&dev->dev, "device initialised!\n"); + + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) + dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err); + + err = usb_reset_configuration(dev); + if (err < 0) + dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err); + dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + + mbox3_setup_48_24_magic(dev); + dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + + return 0; /* Successful boot */ +} #define MICROBOOK_BUF_SIZE 128 @@ -1324,6 +1631,10 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x0dba, 0x3000): /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); + case USB_ID(0x0dba, 0x5000): + /* Digidesign Mbox 3 */ + return snd_usb_mbox3_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ -- 2.35.1

4 8

The cs35l41 driver does not work on several laptop models that lack the _DSD block.
by XiaoYan Li 22 Aug '22

22 Aug '22

Hi, Several laptop models employ CSC3551 but lack a corresponding _DSD block in the ACPI table, including but not limited to: - Zenbook S 13 OLED (UM5302TA) - HP Envy x360 ey0xxx - Lenovo Legion 7i Gen 7 Here are some relevant links: - https://bugzilla.kernel.org/show_bug.cgi?id=216194 - https://lore.kernel.org/all/20220703053225.2203-1-xw897002528@gmail.com/T/#u - https://lore.kernel.org/all/20220811152923.10143-1-faenkhauser@gmail.com/T/… The following is a snippet of the ACPI table from my laptop (UM5302TA): --- Device (SPKR) { Name (_HID, "CSC3551") // _HID: Hardware ID Name (_SUB, "10431F12") // _SUB: Subsystem ID Name (_UID, One) // _UID: Unique ID Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { Name (RBUF, ResourceTemplate () { I2cSerialBusV2 (0x0040, ControllerInitiated, 0x000F4240, AddressingMode7Bit, "\\_SB.I2CA", 0x00, ResourceConsumer, , Exclusive, ) I2cSerialBusV2 (0x0041, ControllerInitiated, 0x000F4240, AddressingMode7Bit, "\\_SB.I2CA", 0x00, ResourceConsumer, , Exclusive, ) GpioIo (Exclusive, PullDown, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPIO", 0x00, ResourceConsumer, , ) { // Pin list 0x0004 } GpioIo (Exclusive, PullUp, 0x0000, 0x0000, IoRestrictionInputOnly, "\\_SB.GPIO", 0x00, ResourceConsumer, , ) { // Pin list 0x009B } GpioIo (Shared, PullUp, 0x0064, 0x0000, IoRestrictionInputOnly, "\\_SB.GPIO", 0x00, ResourceConsumer, , ) { // Pin list 0x0009 } GpioInt (Edge, ActiveBoth, Shared, PullUp, 0x0064, "\\_SB.GPIO", 0x00, ResourceConsumer, , ) { // Pin list 0x0009 } }) Return (RBUF) /* \_SB_.I2CA.SPKR._CRS.RBUF */ } Method (_STA, 0, NotSerialized) // _STA: Status { If ((AMPD == Zero)) { Return (Zero) } Else { Return (0x0F) } } Method (_DIS, 0, NotSerialized) // _DIS: Disable Device { } } --- The Cirrus driver for Windows works perfectly. I don't see anything related to the _DSD stuff in the INF files that came with the driver (if I read them correctly). For reference, here are the two INF files from UM5302TA: https://gist.github.com/lixiaoyan/7e1f11b153b38339007f6ec9aa149f81 I think we should support those devices out of the box, as the Windows driver does. This problem does not appear to be limited to a single model; rather, it appears to be a widespread issue. At least, we should add some quirks for the models listed above. Thanks.

1 1

missing sound on kernel-5.15
by GitHub issues - opened 22 Aug '22

22 Aug '22

alsa-project/alsa-lib issue #264 was opened from chihhao-chen: I am testing many types of headsets on Android platform with different kernel versions. On kernel-5.10, calling sequence before playback is snd_usb_hw_params -> set highest sampling rate, no SAMPLING_FREQ_CONTROL USB request in this stage snd_pcm_release snd_usb_hw_params -> set proper sampling rate, still no SAMPLING_FREQ_CONTROL USB request snd_usb_pcm_parepare -> in configure_endpoint SAMPLING_FREQ_CONTROL USB request was sent With the same calling sequence, a slightly behavior change on kernel-5.15 is snd_usb_hw_params -> set highest sampling rate, SAMPLING_FREQ_CONTROL USB request was sent snd_pcm_release snd_usb_hw_params -> set proper sampling rate, SAMPLING_FREQ_CONTROL USB request was sent too (because configure_endpoint was moved to snd_usb_hw_params function) snd_usb_pcm_parepare -> no SAMPLING_FREQ_CONTROL USB request because of USB EP flag check I checked all USB packets and confirmed audio data output was correct. But there may be missing sound problem for the first sound with many headsets (for example Samsung headset). I found this issue is related to two-times sampling rate set request. (I tried forcibly skip first USB request, everything became okay.) So do you guys know why configure_endpoint was moved to snd_usb_hw_params? Or could you provide patch for this problem? Many thanks Chihhao Issue URL : https://github.com/alsa-project/alsa-lib/issues/264 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Do we need to check state for hw_free ?
by Kuninori Morimoto 22 Aug '22

22 Aug '22

Hi ALSA ML I wonder do we need to check state (A) during dpcm_be_dai_hw_free() ? I think users count check (B) is very enough... void dpcm_be_dai_hw_free() { ... (B) if (be->dpcm[stream].users > 1) continue; ^ if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && | (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && | (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && (A) (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && | (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && | (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND)) v continue; ... __soc_pcm_hw_free(be, be_substream); } Thank you for your help !! Best regards --- Kuninori Morimoto

1 0

bug: Fix verbose logging in GitHub Workflow
by GitHub pull_request - opened 22 Aug '22

22 Aug '22

alsa-project/alsa-ucm-conf pull request #202 was opened from PatrickLang: I noticed the GitHub Workflow `ucm_validate` was setting `LOGLEVEL` instead of `V` which is used in the `Makefile` to set the log level. The tests in this repo are still failing, but fixing the logging first should help. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/202 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/202.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ALSA: memalloc: Revive x86-specific WC page allocations again
by Takashi Iwai 21 Aug '22

21 Aug '22

We dropped the x86-specific hack for WC-page allocations with a hope that the standard dma_alloc_wc() works nowadays. Alas, it doesn't, and we need to take back some workaround again, but in a different form, as the previous one was broken for some platforms. This patch re-introduces the x86-specific WC-page allocations, but it uses rather the manual page allocations instead of dma_alloc_coherent(). The use of dma_alloc_coherent() was also a potential problem in the recent addition of the fallback allocation for noncontig pages, and this patch eliminates both at once. Fixes: 9882d63bea14 ("ALSA: memalloc: Drop x86-specific hack for WC allocations") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216363 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Use gfp_t sound/core/memalloc.c | 87 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 71 insertions(+), 16 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index d3885cb02270..b665ac66ccbe 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -20,6 +20,13 @@ static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab); +#ifdef CONFIG_SND_DMA_SGBUF +static void *do_alloc_fallback_pages(struct device *dev, size_t size, + dma_addr_t *addr, bool wc); +static void do_free_fallback_pages(void *p, size_t size, bool wc); +static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); +#endif + /* a cast to gfp flag from the dev pointer; for CONTINUOUS and VMALLOC types */ static inline gfp_t snd_mem_get_gfp_flags(const struct snd_dma_buffer *dmab, gfp_t default_gfp) @@ -277,16 +284,21 @@ EXPORT_SYMBOL(snd_sgbuf_get_chunk_size); /* * Continuous pages allocator */ -static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) +static void *do_alloc_pages(size_t size, dma_addr_t *addr, gfp_t gfp) { - gfp_t gfp = snd_mem_get_gfp_flags(dmab, GFP_KERNEL); void *p = alloc_pages_exact(size, gfp); if (p) - dmab->addr = page_to_phys(virt_to_page(p)); + *addr = page_to_phys(virt_to_page(p)); return p; } +static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + return do_alloc_pages(size, &dmab->addr, + snd_mem_get_gfp_flags(dmab, GFP_KERNEL)); +} + static void snd_dma_continuous_free(struct snd_dma_buffer *dmab) { free_pages_exact(dmab->area, dmab->bytes); @@ -463,6 +475,25 @@ static const struct snd_malloc_ops snd_dma_dev_ops = { /* * Write-combined pages */ +/* x86-specific allocations */ +#ifdef CONFIG_SND_DMA_SGBUF +static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + return do_alloc_fallback_pages(dmab->dev.dev, size, &dmab->addr, true); +} + +static void snd_dma_wc_free(struct snd_dma_buffer *dmab) +{ + do_free_fallback_pages(dmab->area, dmab->bytes, true); +} + +static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return snd_dma_continuous_mmap(dmab, area); +} +#else static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) { return dma_alloc_wc(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); @@ -479,6 +510,7 @@ static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, return dma_mmap_wc(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } +#endif /* CONFIG_SND_DMA_SGBUF */ static const struct snd_malloc_ops snd_dma_wc_ops = { .alloc = snd_dma_wc_alloc, @@ -486,10 +518,6 @@ static const struct snd_malloc_ops snd_dma_wc_ops = { .mmap = snd_dma_wc_mmap, }; -#ifdef CONFIG_SND_DMA_SGBUF -static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); -#endif - /* * Non-contiguous pages allocator */ @@ -669,6 +697,37 @@ static const struct snd_malloc_ops snd_dma_sg_wc_ops = { .get_chunk_size = snd_dma_noncontig_get_chunk_size, }; +/* manual page allocations with wc setup */ +static void *do_alloc_fallback_pages(struct device *dev, size_t size, + dma_addr_t *addr, bool wc) +{ + gfp_t gfp = GFP_KERNEL | __GFP_NORETRY | __GFP_NOWARN; + void *p; + + again: + p = do_alloc_pages(size, addr, gfp); + if (!p || (*addr + size - 1) & ~dev->coherent_dma_mask) { + if (IS_ENABLED(CONFIG_ZONE_DMA32) && !(gfp & GFP_DMA32)) { + gfp |= GFP_DMA32; + goto again; + } + if (IS_ENABLED(CONFIG_ZONE_DMA) && !(gfp & GFP_DMA)) { + gfp = (gfp & ~GFP_DMA32) | GFP_DMA; + goto again; + } + } + if (p && wc) + set_memory_wc((unsigned long)(p), size >> PAGE_SHIFT); + return p; +} + +static void do_free_fallback_pages(void *p, size_t size, bool wc) +{ + if (wc) + set_memory_wb((unsigned long)(p), size >> PAGE_SHIFT); + free_pages_exact(p, size); +} + /* Fallback SG-buffer allocations for x86 */ struct snd_dma_sg_fallback { size_t count; @@ -679,14 +738,11 @@ struct snd_dma_sg_fallback { static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab, struct snd_dma_sg_fallback *sgbuf) { + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; size_t i; - if (sgbuf->count && dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) - set_pages_array_wb(sgbuf->pages, sgbuf->count); for (i = 0; i < sgbuf->count && sgbuf->pages[i]; i++) - dma_free_coherent(dmab->dev.dev, PAGE_SIZE, - page_address(sgbuf->pages[i]), - sgbuf->addrs[i]); + do_free_fallback_pages(page_address(sgbuf->pages[i]), PAGE_SIZE, wc); kvfree(sgbuf->pages); kvfree(sgbuf->addrs); kfree(sgbuf); @@ -698,6 +754,7 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) struct page **pages; size_t i, count; void *p; + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); if (!sgbuf) @@ -712,15 +769,13 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) goto error; for (i = 0; i < count; sgbuf->count++, i++) { - p = dma_alloc_coherent(dmab->dev.dev, PAGE_SIZE, - &sgbuf->addrs[i], DEFAULT_GFP); + p = do_alloc_fallback_pages(dmab->dev.dev, PAGE_SIZE, + &sgbuf->addrs[i], wc); if (!p) goto error; sgbuf->pages[i] = virt_to_page(p); } - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) - set_pages_array_wc(pages, count); p = vmap(pages, count, VM_MAP, PAGE_KERNEL); if (!p) goto error; -- 2.35.3

1 0

[PATCH] ALSA: memalloc: Revive x86-specific WC page allocations again
by Takashi Iwai 21 Aug '22

21 Aug '22

We dropped the x86-specific hack for WC-page allocations with a hope that the standard dma_alloc_wc() works nowadays. Alas, it doesn't, and we need to take back some workaround again, but in a different form, as the previous one was broken for some platforms. This patch re-introduces the x86-specific WC-page allocations, but it uses rather the manual page allocations instead of dma_alloc_coherent(). The use of dma_alloc_coherent() was also a potential problem in the recent addition of the fallback allocation for noncontig pages, and this patch eliminates both at once. Fixes: 9882d63bea14 ("ALSA: memalloc: Drop x86-specific hack for WC allocations") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=216363 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 87 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 71 insertions(+), 16 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index d3885cb02270..90d74855d988 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -20,6 +20,13 @@ static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab); +#ifdef CONFIG_SND_DMA_SGBUF +static void *do_alloc_fallback_pages(struct device *dev, size_t size, + dma_addr_t *addr, bool wc); +static void do_free_fallback_pages(void *p, size_t size, bool wc); +static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); +#endif + /* a cast to gfp flag from the dev pointer; for CONTINUOUS and VMALLOC types */ static inline gfp_t snd_mem_get_gfp_flags(const struct snd_dma_buffer *dmab, gfp_t default_gfp) @@ -277,16 +284,21 @@ EXPORT_SYMBOL(snd_sgbuf_get_chunk_size); /* * Continuous pages allocator */ -static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) +static void *do_alloc_pages(size_t size, dma_addr_t *addr, unsigned long gfp) { - gfp_t gfp = snd_mem_get_gfp_flags(dmab, GFP_KERNEL); void *p = alloc_pages_exact(size, gfp); if (p) - dmab->addr = page_to_phys(virt_to_page(p)); + *addr = page_to_phys(virt_to_page(p)); return p; } +static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + return do_alloc_pages(size, &dmab->addr, + snd_mem_get_gfp_flags(dmab, GFP_KERNEL)); +} + static void snd_dma_continuous_free(struct snd_dma_buffer *dmab) { free_pages_exact(dmab->area, dmab->bytes); @@ -463,6 +475,25 @@ static const struct snd_malloc_ops snd_dma_dev_ops = { /* * Write-combined pages */ +/* x86-specific allocations */ +#ifdef CONFIG_SND_DMA_SGBUF +static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + return do_alloc_fallback_pages(dmab->dev.dev, size, &dmab->addr, true); +} + +static void snd_dma_wc_free(struct snd_dma_buffer *dmab) +{ + do_free_fallback_pages(dmab->area, dmab->bytes, true); +} + +static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return snd_dma_continuous_mmap(dmab, area); +} +#else static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) { return dma_alloc_wc(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); @@ -479,6 +510,7 @@ static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, return dma_mmap_wc(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } +#endif /* CONFIG_SND_DMA_SGBUF */ static const struct snd_malloc_ops snd_dma_wc_ops = { .alloc = snd_dma_wc_alloc, @@ -486,10 +518,6 @@ static const struct snd_malloc_ops snd_dma_wc_ops = { .mmap = snd_dma_wc_mmap, }; -#ifdef CONFIG_SND_DMA_SGBUF -static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); -#endif - /* * Non-contiguous pages allocator */ @@ -669,6 +697,37 @@ static const struct snd_malloc_ops snd_dma_sg_wc_ops = { .get_chunk_size = snd_dma_noncontig_get_chunk_size, }; +/* manual page allocations with wc setup */ +static void *do_alloc_fallback_pages(struct device *dev, size_t size, + dma_addr_t *addr, bool wc) +{ + void *p; + unsigned long gfp = DEFAULT_GFP & ~__GFP_COMP; + + again: + p = do_alloc_pages(size, addr, gfp); + if (!p || (*addr + size - 1) & ~dev->coherent_dma_mask) { + if (IS_ENABLED(CONFIG_ZONE_DMA32) && !(gfp & GFP_DMA32)) { + gfp |= GFP_DMA32; + goto again; + } + if (IS_ENABLED(CONFIG_ZONE_DMA) && !(gfp & GFP_DMA)) { + gfp = (gfp & ~GFP_DMA32) | GFP_DMA; + goto again; + } + } + if (p && wc) + set_memory_wc((unsigned long)(p), size >> PAGE_SHIFT); + return p; +} + +static void do_free_fallback_pages(void *p, size_t size, bool wc) +{ + if (wc) + set_memory_wb((unsigned long)(p), size >> PAGE_SHIFT); + free_pages_exact(p, size); +} + /* Fallback SG-buffer allocations for x86 */ struct snd_dma_sg_fallback { size_t count; @@ -679,14 +738,11 @@ struct snd_dma_sg_fallback { static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab, struct snd_dma_sg_fallback *sgbuf) { + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; size_t i; - if (sgbuf->count && dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) - set_pages_array_wb(sgbuf->pages, sgbuf->count); for (i = 0; i < sgbuf->count && sgbuf->pages[i]; i++) - dma_free_coherent(dmab->dev.dev, PAGE_SIZE, - page_address(sgbuf->pages[i]), - sgbuf->addrs[i]); + do_free_fallback_pages(page_address(sgbuf->pages[i]), PAGE_SIZE, wc); kvfree(sgbuf->pages); kvfree(sgbuf->addrs); kfree(sgbuf); @@ -698,6 +754,7 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) struct page **pages; size_t i, count; void *p; + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); if (!sgbuf) @@ -712,15 +769,13 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) goto error; for (i = 0; i < count; sgbuf->count++, i++) { - p = dma_alloc_coherent(dmab->dev.dev, PAGE_SIZE, - &sgbuf->addrs[i], DEFAULT_GFP); + p = do_alloc_fallback_pages(dmab->dev.dev, PAGE_SIZE, + &sgbuf->addrs[i], wc); if (!p) goto error; sgbuf->pages[i] = virt_to_page(p); } - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) - set_pages_array_wc(pages, count); p = vmap(pages, count, VM_MAP, PAGE_KERNEL); if (!p) goto error; -- 2.35.3

2 2

Re: Rockchip I2S commit possibly ignored
by sugar zhang 21 Aug '22

21 Aug '22

Hi Geraldo, Mark, 在 2022/8/17 1:54, Geraldo Nascimento 写道: > On Tue, Aug 16, 2022 at 04:22:37PM +0100, Mark Brown wrote: >> On Tue, Aug 16, 2022 at 10:37:18AM -0300, Geraldo Nascimento wrote: >> >>> This isn't my patch, it's a patch by Sugar Zhang from Rockchip that was >>> supposed to have been applied one year ago, give or take 10 days. I >>> can't resend a patch that wasn't authored by me. >> There's absolutely no problem with reposting someone else's patch - just >> add your Signed-off-by at the end of the signoff chain. > Mark, I'm in no position to lecture anyone, least of who, you, hard-working > ASoC maintainer that you are. But there are *tons* of problems with > reposting someone else's patch, even if they had been previously given > the green-light but misteriously vanished. > > You are assuming responsibility for someone else's work! Let's see in > this case what would happen: > > In this case, the Rockchip I2S missing reset affair, I have tested > the proposed changes, and they are OK, *if* you add the reset properties > in the device tree. > > For example, I was playing with HDMI sound on the RK3399pro Radxa Rock Pi > N10 and I added the following properties to my private rk3399.dtsi: > > --- a/arch/arm64/boot/dts/rockchip/rk3399.dtsi > +++ b/arch/arm64/boot/dts/rockchip/rk3399.dtsi > @@ -1695,6 +1695,8 @@ i2s2: i2s@ff8a0000 { > dma-names = "tx", "rx"; > clock-names = "i2s_clk", "i2s_hclk"; > clocks = <&cru SCLK_I2S2_8CH>, <&cru HCLK_I2S2_8CH>; > + reset-names = "reset-m", "reset-h"; > + resets = <&cru SRST_I2S2_8CH>, <&cru SRST_H_I2S2_8CH>; > power-domains = <&power RK3399_PD_SDIOAUDIO>; > #sound-dai-cells = <0>; > status = "disabled"; > > --- > > > And once I okay &i2s2 and &hdmi_sound inside rk3399pro-vmarc-som.dtsi I > get kind of working HDMI sound, very clipped, I have to software adjust > the volume to 30% to get clean, not speaker-blowing sound. But I > digress, the point that it kinda works is not the point at all. > > My main point is that without adding "resets" and "reset-names" to *at > least* every Rockchip device tree that enables sound over HDMI, just an > example, you get systems with non-working HDMI. I just tested it, I > omitted "resets" and "reset-names" from my device tree and then had > to SSH into the black screen to revert the changes to my boot partition. Actually, the reset is used for i2s SLAVE mode which may fail to clear if the MASTER side stop the clk before i2s done. in this situation, we do reset the controller and bring it back to normal state. But, for HDMI sound, the i2s works as MASTER mode and should not be fail to clear. could you share your I2S and HDMI register dump to me to dig out the root cause. OTOH，I have noticed that some patchs ignored on upstream, I will check and repost those patchs. > So it's not trivial to RESEND this. It amounts to device tree overhaul > of arch/arm64/boot/dts/rockchip > > This would require community effort of say, equivalent to LibreELEC > resources, which are not many, but they have enough patience to test > every proposed change to Rockchip device trees, and could help upstream > this. > > Thank you, > Geraldo Nascimento -- Best Regards! 张学广/Sugar 瑞芯微电子股份有限公司 Rockchip Electronics Co., Ltd.

2 1

Rockchip/rk3399-gru-sound: Use 48KHz sample rate for all devices
by GitHub pull_request - opened 20 Aug '22

20 Aug '22

alsa-project/alsa-ucm-conf pull request #201 was opened from alpernebbi: The rk3399-gru-kevin board has problems when trying to simultaneously playback and record audio with the default PulseAudio sample rate of 44.1KHz. When the following command is run, the playback starts a few seconds after the recording finishes. arecord -vvv -d 4 /dev/null & sleep 0.2; speaker-test -l 1 -p 100000 -t sine When the sample rates are set to 48KHz either with PA configuration or in the UCM, playback and recording both start immediately. Another example is when a music player is running and we start `arecord`, the music starts stuttering but `arecord` can't record anything either. Apparently, this is a hardware limitation due to the I2S lines being shared between the devices. Set all the device rates to 48KHz like Chrome OS does to make things work smoothly. [1] https://chromium-review.googlesource.com/389695 [2] https://chromium-review.googlesource.com/898682 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/201 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/201.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH RESEND 0/3] ASoC: mediatek: support SOF for mt8186
by Chunxu Li 20 Aug '22

20 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> In these patches, we add SOF support for mt8186 which have HiFi5 DSP inside. Chunxu Li (3): ASoC: mediatek: mt8186: support DSP downlink ASoC: mediatek: mt8186: add SOF support on mt8186-mt6366-rt1019-rt5682s ASoC: mediatek: mt8186: add SOF support on mt8186-mt6366-da7219-max98357 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3 + sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 12 ++ .../mt8186/mt8186-mt6366-da7219-max98357.c | 168 ++++++++++++++++-- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 168 ++++++++++++++++-- 4 files changed, 325 insertions(+), 26 deletions(-) -- 2.25.1

2 4

[RESEND PATCH v2 1/2] dt-bindings: dsp: fsl: Add SOF compatile string for i.MX8ULP
by Shengjiu Wang 20 Aug '22

20 Aug '22

Add SOF compatile string "fsl,imx8ulp-dsp" for supporting DSP device on i.MX8ULP platform. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Rob Herring <robh(a)kernel.org> --- changes in v2 - add acked-by Rob Documentation/devicetree/bindings/dsp/fsl,dsp.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml index e66ef2da7879..9af40da5688e 100644 --- a/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml +++ b/Documentation/devicetree/bindings/dsp/fsl,dsp.yaml @@ -20,6 +20,7 @@ properties: - fsl,imx8qxp-dsp - fsl,imx8qm-dsp - fsl,imx8mp-dsp + - fsl,imx8ulp-dsp - fsl,imx8qxp-hifi4 - fsl,imx8qm-hifi4 - fsl,imx8mp-hifi4 -- 2.34.1

3 4

[PATCH -next 1/3] ASoC: amd: acp: add missing platform_device_unregister() in acp_pci_probe()
by Yang Yingliang 20 Aug '22

20 Aug '22

Add missing platform_device_unregister() in error path in acp_pci_probe(). Fixes: c49f5e74a11e ("ASoC: amd: acp: Add error handling cases") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/amd/acp/acp-pci.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 2c8e960cc9a6..5bb23ebe1216 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -104,6 +104,7 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id addr = pci_resource_start(pci, 0); chip->base = devm_ioremap(&pci->dev, addr, pci_resource_len(pci, 0)); if (!chip->base) { + platform_device_unregister(dmic_dev); ret = -ENOMEM; goto release_regions; } -- 2.25.1

2 3

[PATCH] ASoC: amd: acp: remove unnecessary NULL checks
by Dan Carpenter 20 Aug '22

20 Aug '22

The list iterator can never be NULL. Delete the bogus NULL checks. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/acp/acp-platform.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index beee53aedeaf..85a81add4ef9 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -106,14 +106,14 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) spin_lock(&adata->acp_lock); list_for_each_entry(stream, &adata->stream_list, list) { - if (stream && (ext_intr_stat & stream->irq_bit)) { + if (ext_intr_stat & stream->irq_bit) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; } if (adata->rsrc->no_of_ctrls == 2) { - if (stream && (ext_intr_stat1 & stream->irq_bit)) { + if (ext_intr_stat1 & stream->irq_bit) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, (rsrc->irqp_used - 1))); snd_pcm_period_elapsed(stream->substream); -- 2.35.1

2 1

[PATCH] ASoC: Change handling of unimplemented set_bclk_ratio
by Martin Povišer 20 Aug '22

20 Aug '22

If a 'set_bclk_ratio' call is attempted on a DAI not implementing the method, make it an -ENOSUPP error instead of -EINVAL. Assume the DAI can still be okay with the ratio, just does not care to register a handler. No current in-tree users of snd_soc_dai_set_bclk_ratio seem to inspect the return value, but -ENOSUPP disables an error print from within the common soc_dai_ret return filter. With the new behavior a machine driver can do a blanket 'set_bclk_ratio' on all DAIs on a bus, some of which may care about the ratio, some of which may not. Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/soc-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index d530e8c2b77b..49752af0e205 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -124,7 +124,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) { - int ret = -EINVAL; + int ret = -ENOTSUPP; if (dai->driver->ops && dai->driver->ops->set_bclk_ratio) -- 2.33.0

2 1

[PATCH] soundwire: qcom: remove unneeded check
by Steev Klimaszewski 19 Aug '22

19 Aug '22

commit 1fd0d85affe4d6 ("soundwire: qcom: Add flag for software clock gating check") added a flag for software clock gating check, however in commit 33ba01788889666 ("soundwire: qcom: Add support for controlling audio CGCR from HLOS") the same check was added without the flag, so we would still end up failing the software clock gating check. Originally reported by Amit Pundir on DB845c, I also saw it on the Lenovo Yoga C630, right before the splat, we would see qcom-soundwire wcd934x-soundwire.6.auto: Failed to get audio_cgcr reset required for soundwire-v1.6.0 however, SDM845 has a soundwire-v1.3.0 Since the flag was added in 1fd0d85affe, lets just remove this one. Fixes: 33ba01788889 ("soundwire: qcom: Add support for controlling audio CGCR from HLOS") Signed-off-by: Steev Klimaszewski <steev(a)kali.org> Reported-by: Amit Pundir <amit.pundir(a)linaro.org> --- drivers/soundwire/qcom.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 709a7c1e0704..b621f7fb866f 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1355,10 +1355,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.compute_params = &qcom_swrm_compute_params; ctrl->bus.clk_stop_timeout = 300; - ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); - if (IS_ERR(ctrl->audio_cgcr)) - dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); - ret = qcom_swrm_get_port_config(ctrl); if (ret) goto err_clk; -- 2.35.1

2 2

[PATCH v2] ASoC: SOF: Fix compilation when HDA_AUDIO_CODEC config is disabled
by Cezary Rojewski 19 Aug '22

19 Aug '22

hda_codec_device_init() expects three parameters, not two. Fixes: 3fd63658caed ("ASoC: Intel: Drop hdac_ext usage for codec device creation") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- Changes in v2: - adjusted SHA-1 provided in Fixes tag sound/soc/sof/intel/hda-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 73336648cd25..1e9afc48394c 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -213,7 +213,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, put_device(&codec->core.dev); } #else - codec = hda_codec_device_init(&hbus->core, address); + codec = hda_codec_device_init(&hbus->core, address, HDA_DEV_ASOC); ret = PTR_ERR_OR_ZERO(codec); #endif -- 2.25.1

3 2

[PATCH] ASoC: SOF: Fix compilation when HDA_AUDIO_CODEC config is disabled
by Cezary Rojewski 19 Aug '22

19 Aug '22

hda_codec_device_init() expects three parameters, not two. Fixes: f2183b08c81a ("ASoC: Intel: Drop hdac_ext usage for codec device creation") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/sof/intel/hda-codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 73336648cd25..1e9afc48394c 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -213,7 +213,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, put_device(&codec->core.dev); } #else - codec = hda_codec_device_init(&hbus->core, address); + codec = hda_codec_device_init(&hbus->core, address, HDA_DEV_ASOC); ret = PTR_ERR_OR_ZERO(codec); #endif -- 2.25.1

2 2

chromium - cracking pulseaudio issue (small buffer size settings - 512)
by GitHub issues - edited 19 Aug '22

19 Aug '22

alsa-project/alsa-lib issue #262 was edited from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. alsa-lib 1.2.7.2 alsa-utils 1.2.7 alsa-plugins 1.2.2 Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/3] ASoC platform driver for Apple MCA
by Martin Povišer 19 Aug '22

19 Aug '22

Hi, for review I am posting another version of the ASoC platform driver to be used on Apple M1/M2 platforms (some previous version was attached to the macaudio RFC v2 [0]). Martin Changes since [0]: - addition of locking (extra commit) - transition to set_bclk_ratio (instead of getting the bclk ratio from set_sysclk) - using shared reset control and documenting the reset in binding - formatting, comments, and a minor fix to hw driving [0] https://lore.kernel.org/asahi/20220606191910.16580-1-povik+lin@cutebit.org/ Martin Povišer (3): dt-bindings: sound: Add Apple MCA I2S transceiver ASoC: apple: mca: Start new platform driver ASoC: apple: mca: Add locks on foreign cluster access .../devicetree/bindings/sound/apple,mca.yaml | 109 ++ MAINTAINERS | 8 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/apple/Kconfig | 9 + sound/soc/apple/Makefile | 3 + sound/soc/apple/mca.c | 1173 +++++++++++++++++ 7 files changed, 1304 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/apple,mca.yaml create mode 100644 sound/soc/apple/Kconfig create mode 100644 sound/soc/apple/Makefile create mode 100644 sound/soc/apple/mca.c -- 2.33.0

3 11

ucm2: Add intial acpes8336 support
by GitHub pull_request - opened 19 Aug '22

19 Aug '22

alsa-project/alsa-ucm-conf pull request #200 was opened from yangxiaohua2009: This add support for es8336 on AMD Jadeite platform. ASLAINFO http://alsa-project.org/db/?f=b5aae41350b63c40c7743ba43ff8a7178ee93229 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/200 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/200.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

How trivial would it be to port the DICE driver from FFADO?
by forevernoob＠tutanota.com 19 Aug '22

19 Aug '22

Hello, I'm new to these kinds of mailing lists so please forgive me if I'm doing something dumb. I would love to see Allen & Heath Zed R16 as well as Midas Venice F32 support in ALSA. Currently (at least according to libffado-2.4.4/configuration), they are supported via the DICE driver: { # Allen and Heath Zed R16. Information from Brendan Pike. vendorid = 0x000004C4; modelid = 0x00000000; vendorname = "Allen and Heath"; modelname = "Zed R16"; driver = "DICE"; mixer = "Generic_Dice_EAP"; }, { # Midas Venice F32. Information from Jano Svitok. vendorid = 0x0010C73F; modelid = 0x00000001; vendorname = "Midas"; modelname = "Venice F32"; driver = "DICE"; mixer = "Generic_Dice_EAP"; }, In their source code I think it's this file: http://subversion.ffado.org/browser/branches/2.4.x/libffado/src/dice/dice_e… So my question would be: How difficult would it be to port this driver (more specifically: Port it sufficiently enough to support the aforementioned devices) ? Are we talking trivial enough for someone with minimal C experience, a dedicated team of ALSA devs or something else? Thanks in advance, FN

2 2

cracking pulseaudio issue
by GitHub issues - edited 19 Aug '22

19 Aug '22

alsa-project/alsa-lib issue #262 was edited from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. alsa-lib 1.2.7.2 alsa-utils 1.2.7 alsa-plugins 1.2.2 Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_hda_intel: Woke up but no data
by GitHub issues - reopened 19 Aug '22

19 Aug '22

alsa-project/alsa-lib issue #262 was reopened from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. alsa-lib 1.2.7.2 alsa-utils 1.2.7 alsa-plugins 1.2.2 Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Raising volume past 100%
by David Griffith 18 Aug '22

18 Aug '22

After reinstalling Debian on a laptop, I once again had to face the problem of how to get volume out of the built-in speakers boosted. The last time I reinstalled, I somehow got the MATE volume control applet to let me turn the volume past 100% using the mouse wheel. Over the past two days I've been trying to rediscover what I neglected to write down. I'm already aware that by bringing up the Sound Preferences dialog, I can raise volume to 153% and no further. I also know that I can use pulseaudio-ctl(1) or pactl(1) to set the volume at any arbitrary level past 100%. I remember that I went looking for solutions other people found for this that involved editing /etc/asound.conf or ~/.asoundrc, but the best I can get with that is add an extra slider to alsamixer's display that acts as apreamplifier. I was going off of http://alien.slackbook.org/blog/adding-an-alsa-software-pre-amp-to-fix-low-…. What could I have possibly done to make this work the last time? -- David Griffith dave(a)661.org A: Because it fouls the order in which people normally read text. Q: Why is top-posting such a bad thing? A: Top-posting. Q: What is the most annoying thing in e-mail?

1 0

[PATCH 0/2] ASoC: Add a driver for the Iron Device SMA1303 Amp
by Kiseok Jo 18 Aug '22

18 Aug '22

This adds basic audio support for the Iron Device SMA1303 amplifier. Kiseok Jo (2): ASoC: sma1303: Add driver for Iron Device SMA1303 Amp ASoC: dt-bindings: sma1303: Iron Device SMA1303 Amp .../bindings/sound/irondevice,sma1303.yaml | 56 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1303.c | 3388 +++++++++++++++++ sound/soc/codecs/sma1303.h | 604 +++ 5 files changed, 4057 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml create mode 100644 sound/soc/codecs/sma1303.c create mode 100644 sound/soc/codecs/sma1303.h -- 2.30.2

2 3

[PATCH -next] ASoC: SOF: ipc4-topology: fix wrong use of sizeof in sof_ipc4_widget_setup_comp_src()
by Yang Yingliang 18 Aug '22

18 Aug '22

It should be size of the struct sof_ipc4_src, not data pointer pass to sof_update_ipc_object(). Fixes: b85f4fc40d56 ("ASoC: SOF: add ipc4 SRC module support") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index af072b484a60..c6abfaf5d532 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -771,7 +771,7 @@ static int sof_ipc4_widget_setup_comp_src(struct snd_sof_widget *swidget) goto err; ret = sof_update_ipc_object(scomp, src, SOF_SRC_TOKENS, swidget->tuples, - swidget->num_tuples, sizeof(src), 1); + swidget->num_tuples, sizeof(*src), 1); if (ret) { dev_err(scomp->dev, "Parsing SRC tokens failed\n"); goto err; -- 2.25.1

3 2

Add ASUS ROG Zenith II Extreme configuration
by GitHub pull_request - opened 18 Aug '22

18 Aug '22

alsa-project/alsa-lib pull request #263 was opened from nabajour: This points iec958_device to the correct PCM device. complements this [ALSA kernel patch](https://bugzilla.kernel.org/show_bug.cgi?id=211005). Request URL : https://github.com/alsa-project/alsa-lib/pull/263 Patch URL : https://github.com/alsa-project/alsa-lib/pull/263.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[bug report] ASoC: amd: Add common framework to support I2S on ACP SOC
by Dan Carpenter 18 Aug '22

18 Aug '22

Hello Ajit Kumar Pandey, The patch 623621a9f9e1: "ASoC: amd: Add common framework to support I2S on ACP SOC" from Oct 19, 2021, leads to the following Smatch static checker warning: sound/soc/amd/acp/acp-platform.c:199 acp_dma_open() warn: '&stream->list' not removed from list sound/soc/amd/acp/acp-platform.c 173 static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) 174 { 175 struct snd_pcm_runtime *runtime = substream->runtime; 176 struct device *dev = component->dev; 177 struct acp_dev_data *adata = dev_get_drvdata(dev); 178 struct acp_stream *stream; 179 int ret; 180 181 stream = kzalloc(sizeof(*stream), GFP_KERNEL); 182 if (!stream) 183 return -ENOMEM; 184 185 stream->substream = substream; 186 187 spin_lock_irq(&adata->acp_lock); 188 list_add_tail(&stream->list, &adata->stream_list); Stream added to the list 189 spin_unlock_irq(&adata->acp_lock); 190 191 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) 192 runtime->hw = acp_pcm_hardware_playback; 193 else 194 runtime->hw = acp_pcm_hardware_capture; 195 196 ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); 197 if (ret < 0) { 198 dev_err(component->dev, "set integer constraint failed\n"); --> 199 kfree(stream); Use after free. 200 return ret; 201 } 202 runtime->private_data = stream; 203 204 writel(1, ACP_EXTERNAL_INTR_ENB(adata)); 205 206 return ret; Could we delay adding the stream to the list until right before this "return 0;"? 207 } regards, dan carpenter

1 0

[PATCH 1/2] ASoC: Intel: Skylake: fix error message of NHLT blob selection
by Icenowy Zheng 18 Aug '22

18 Aug '22

The query to select NHLT blob had been changed to use parameters of the pipeline instead of PCM, however the error message returned when it fails still uses PCM properties. Fix this inconsistency. Fixes: 87b265260046 ("ASoC: Intel: Skylake: Select proper format for NHLT blob") Signed-off-by: Icenowy Zheng <uwu(a)icenowy.me> --- sound/soc/intel/skylake/skl-topology.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 9bdf020a2b64..19994ec8bba1 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1864,7 +1864,8 @@ static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai, } else { dev_err(dai->dev, "Blob NULL for id:%d type:%d dirn:%d ch:%d, freq:%d, fmt:%d\n", mconfig->vbus_id, link_type, params->stream, - params->ch, params->s_freq, params->s_fmt); + pipe_fmt->channels, pipe_fmt->freq, + pipe_fmt->bps); return -EINVAL; } -- 2.37.1

3 9

[RESEND PATCH v2 0/6] ALSA: hda: Unify codec construction
by Cezary Rojewski 18 Aug '22

18 Aug '22

A follow up to the recent HDAudio fixes series [1]. Given the recently reported regression [2], before the page fault occurring on codec shutdown can be fixed, codec construction procedure needs to be updated for skylake and sof-intel drivers. Drivers: pci-hda and avs need no changes - already making use of snd_hda_codec_device_init(). As subject touches code used by the sof-driver, additional review has been conducted on thesofproject/linux [3]. Changes in v2: - dropped snd_hda_ext_core <-> snd_hda_codec dependency by calling snd_hda_codec_device_init() directly in skylake and sof drivers probe enumeration routines, as suggested by Takashi - skylake/sof portion of the change has been split into two separate patches - new functions that aim to replace hdac_ext codec init & exit functionality are added first - for skylake and sof drivers both - third patch in the series now combines the "field -> pointer" change for hdac_hda_priv->codec plus the codec-enumeration adjustments for skylake and sof drivers Both above are here to keep git bisect happy, as suggested by Pierre [1]: https://lore.kernel.org/alsa-devel/20220706120230.427296-7-cezary.rojewski@… [2]: https://lore.kernel.org/alsa-devel/3c40df55-3aee-1e08-493b-7b30cd84dc00@lin… [3]: https://github.com/thesofproject/linux/pull/3775 Cezary Rojewski (6): ASoC: Intel: Skylake: Introduce HDA codec init and exit routines ASoC: SOF: Intel: Introduce HDA codec init and exit routines ASoC: Intel: Drop hdac_ext usage for codec device creation ALSA: hda: Always free codec on the device release ALSA: hda: Remove codec init and exit routines ALSA: hda: Fix page fault in snd_hda_codec_shutdown() include/sound/hda_codec.h | 2 - include/sound/hdaudio_ext.h | 3 -- sound/hda/ext/hdac_ext_bus.c | 53 ------------------- sound/pci/hda/hda_codec.c | 49 ++++++++--------- sound/soc/codecs/hdac_hda.c | 26 ++++----- sound/soc/codecs/hdac_hda.h | 2 +- sound/soc/intel/boards/hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/skylake/skl.c | 53 ++++++++++++++----- sound/soc/sof/intel/hda-codec.c | 55 ++++++++++++++------ 10 files changed, 113 insertions(+), 134 deletions(-) -- 2.25.1

3 9

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NS50PU, NS70PU
by Werner Sembach 18 Aug '22

18 Aug '22

From: Christoffer Sandberg <cs(a)tuxedo.de> Fixes headset microphone detection on Clevo NS50PU and NS70PU. Signed-off-by: Christoffer Sandberg <cs(a)tuxedo.de> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- 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 fd630d62b5a0..fceac608214e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9389,6 +9389,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x7716, "Clevo NS50PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x7717, "Clevo NS70PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x7718, "Clevo L140PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.34.1

2 2

HDA ATI HDMI not supported
by GitHub issues - edited 18 Aug '22

18 Aug '22

alsa-project/alsa-ucm-conf issue #194 was edited from yangxiaohua2009: I am using the latest ucm2 file and alsalib alsainfo [http://alsa-project.org/db/?f=92c58f3a6b72fee05b678386858dbc816d25dbf3](htt… The `alsaucm reload` command returns error saying: `ALSA lib parser.c:244:(error_node) UCM is not supported for this HDA model (HDA ATI HDMI at 0xfeb60000 irq 28)` the HDMI works but the I2S and HDMI all use fallback ucm config. despite the fact that I already added the acpes8336.conf in conf.d and acpes8336 folder. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/194 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

RE: [PATCH 1/2] ASoC: sma1303: Add driver for Iron Device SMA1303 Amp
by Ki-Seok Jo 18 Aug '22

18 Aug '22

Dear Mark Brown, Thank you for your detailed feedback. I will apply what you told me as soon as possible and request it again. I'll ask you again if there's anything wrong with the correction. I appreciate your quick response. Thank you very much for your time Best Regards, Kiseok Jo Iron Device Corporation Tel.: +82-2-541-2896 Mobile: +82-10-3320-0376 Email: kiseok.jo(a)irondevice.com The information in this email and any attachment is confidential and may be legally protected. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure or actions taken as a result of the information in this email is prohibited and may be unlawful. If you are not the intended recipient, please notify us immediately and then delete this e-mail and any attachment. Thank you. -----Original Message----- From: Mark Brown <broonie(a)kernel.org> Sent: Wednesday, August 17, 2022 9:09 PM To: Ki-Seok Jo <kiseok.jo(a)irondevice.com> Cc: Gyu-Hwa Park <gyuhwa.park(a)irondevice.com>; alsa-devel(a)alsa-project.org; Suk-Min Kang <sukmin.kang(a)irondevice.com> Subject: Re: [PATCH 1/2] ASoC: sma1303: Add driver for Iron Device SMA1303 Amp On Wed, Aug 17, 2022 at 12:29:37PM +0900, Kiseok Jo wrote: I got part way through reviewing this driver. The top level feedback here is that the driver appears to be exposing everything in the device to userspace as enumeration controls rather than using standard features like DAPM, or appropriate other control types where there's something that's a better fit than an enum. If there's a strong reason for the driver to do something so unusual then theere should be something in here which explains what's going on. > +// SPDX-License-Identifier: GPL-2.0-or-later > +/* sma1303.c -- sma1303 ALSA SoC Audio driver > + * > + * Copyright 2019 Iron Device Corporation 2019? Please make the entire block a C++ one so things look more intentional. > +static int power_up_down_control_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) { > + struct snd_soc_component *component = > + snd_soc_kcontrol_component(kcontrol); > + struct sma1303_priv *sma1303 = snd_soc_component_get_drvdata(component); > + int sel = (int)ucontrol->value.integer.value[0]; > + > + if (sel && !(sma1303->force_amp_power_down)) > + sma1303_startup(component); > + else > + sma1303_shutdown(component); > + > + return 0; > +} This and all the other controls need to return 1 on change, please make sure a card with your driver your driver passes mixer-test cleanly - it will spot this and other problems. However I wouldn't expect the power controls to exist at all, generally power is managed via DAPM rather than userspace. > +static int force_sdo_bypass_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) { > + struct snd_soc_component *component = > + snd_soc_kcontrol_component(kcontrol); > + struct sma1303_priv *sma1303 = snd_soc_component_get_drvdata(component); > + int sel = (int)ucontrol->value.integer.value[0]; > + > + sma1303->sdo_bypass_flag = (bool)sel; > + > + return 0; > +} What is this bypassing? > +static const char * const sma1303_input_format_text[] = { > + "I2S", "LJ", "Reserved", "Reserved", > + "RJ_16", "RJ_18", "RJ_20", "RJ_24"}; > + > +static const struct soc_enum sma1303_input_format_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_input_format_text), > + sma1303_input_format_text); This should be controlled by the machine driver through set_dai_fmt() and hw_params(). > +static const char * const sma1303_pcm_n_slot_text[] = { > + "Slot_1", "Slot_2", "Slot_3", "Slot_4", "Slot_5", "Slot_6", > + "Slot_7", "Slot_8", "Slot_9", "Slot_10", "Slot_11", "Slot_12", > + "Slot_13", "Slot_14", "Slot_15", "Slot_16"}; > + > +static const struct soc_enum sma1303_pcm_n_slot_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_pcm_n_slot_text), > + sma1303_pcm_n_slot_text); set_tdm_slot() > +static const char * const sma1303_port_config_text[] = { > + "IN_Port", "Reserved", "OUT_Port", "Reserved"}; This should be DT properties, possibly using the pinctrl schema depending on what's actually going on. > +static const char * const sma1303_port_out_sel_text[] = { > + "Disable", "Format_C", "Mixer_out", "After_DSP", > + "Postscaler", "Reserved", "Reserved", "Reserved"}; > + > +static const struct soc_enum sma1303_port_out_sel_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_port_out_sel_text), > + sma1303_port_out_sel_text); This looks like it should be a DAPM routing control. > +static const char * const sma1303_spk_off_slope_text[] = { > + "00", "01", "10", "11"}; > + > +static const struct soc_enum sma1303_spk_off_slope_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_spk_off_slope_text), > + sma1303_spk_off_slope_text); Why make this an enum and not just a numerical control with values 0-3? > +static const char * const sma1303_spkmode_text[] = { > + "Off", "Mono", "Reserved", "Reserved", > + "Stereo", "Reserved", "Reserved", "Reserved"}; > + > +static const struct soc_enum sma1303_spkmode_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_spkmode_text), > + sma1303_spkmode_text); Use a value enum to eliminate the reserved values from user selection, however if this is controlling if the device has mono or stereo speakers it should be a DT property since that's unlikely to change at runtime. > +static const char * const sma1303_input_gain_text[] = { > + "Gain_0dB", "Gain_M6dB", "Gain_M12dB", "Gain_MInf"}; > + > +static const struct soc_enum sma1303_input_gain_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_input_gain_text), > + sma1303_input_gain_text); This should just be a normal volume control. I've stopped reviewing at this point.

1 0

snd_hda_intel: Woke up but no data
by GitHub issues - edited 18 Aug '22

18 Aug '22

alsa-project/alsa-lib issue #262 was edited from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. alsa-lib 1.2.7.2 alsa-utils 1.2.7 alsa-plugins 1.2.2 Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_hda_intel: Woke up but no data
by GitHub issues - opened 18 Aug '22

18 Aug '22

alsa-project/alsa-lib issue #262 was opened from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: fsl_sai: Remove unnecessary FIFO reset in ISR
by Shengjiu Wang 17 Aug '22

17 Aug '22

The FIFO reset drops the words in the FIFO, which may cause channel swap when SAI module is running, especially when the DMA speed is low. So it is not good to do FIFO reset in ISR, then remove the operation. Fixes: e2681a1bf5ae ("ASoC: fsl_sai: Add isr to deal with error flag") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d430eece1d6b..a7fa6f0bf83d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -114,11 +114,8 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) if (flags & FSL_SAI_CSR_SEF) dev_dbg(dev, "isr: Tx Frame sync error detected\n"); - if (flags & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) dev_dbg(dev, "isr: Transmit underrun detected\n"); - /* FIFO reset for safety */ - xcsr |= FSL_SAI_CSR_FR; - } if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n"); @@ -148,11 +145,8 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) if (flags & FSL_SAI_CSR_SEF) dev_dbg(dev, "isr: Rx Frame sync error detected\n"); - if (flags & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) dev_dbg(dev, "isr: Receive overflow detected\n"); - /* FIFO reset for safety */ - xcsr |= FSL_SAI_CSR_FR; - } if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled receive FIFO is full\n"); -- 2.34.1

4 4

Re: [PATCH] ASoC: soc-utils-test: Add test for snd_soc_params_to_bclk()
by Mark Brown 17 Aug '22

17 Aug '22

On Wed, 17 Aug 2022 13:55:08 +0100, Richard Fitzgerald wrote: > snd_soc_params_to_bclk() calculates the BCLK from only the information in > snd_pcm_hw_params. It is therefore a subset of the functionality of > snd_soc_tdm_params_to_bclk() so can use a subset of the test case table. > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: soc-utils-test: Add test for snd_soc_params_to_bclk() commit: e32e23a2b588424aec0c4c4435530f8022318b8f 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

1 0

[PATCH 0/5] ASoC: cs42l42: Some small code improvements
by Richard Fitzgerald 17 Aug '22

17 Aug '22

This is a collection of minor improvements to the code or comments. Richard Fitzgerald (5): ASoC: cs42l42: Don't include kernel.h ASoC: cs42l42: Add include dependencies to cs42l42.h ASoC: cs42l42: Move cs42l42_supply_names to .c file ASoC: cs42l42: Fix comment typo in cs42l42_slow_start_put() ASoC: cs42l42: Use snd_soc_tdm_params_to_bclk() sound/soc/codecs/cs42l42.c | 24 ++++++++++++++++-------- sound/soc/codecs/cs42l42.h | 13 +++++-------- 2 files changed, 21 insertions(+), 16 deletions(-) -- 2.30.2

2 3

[PATCH 0/4] ASoC/soundwire: log actual PING status on resume issues
by Bard Liao 17 Aug '22

17 Aug '22

we've been stuck with problems in the dual-amplifier configurations where one of the two devices seems to become UNATTACHED and never regains sync, see https://github.com/thesofproject/linux/issues/3638. This is a rather infrequent issue that may happen once or twice per month, but still it remains a concern. One possibility is that the device does lose sync but somehow our hardware detection fails to see it resync. This series just adds a basic read directly from the PING frames to help confirm if yes/no the device regain sync. The change is mainly on soundwire. @Mark, Could you ack the ASoC patch if it looks good to you? Pierre-Louis Bossart (4): soundwire: add read_ping_status helper definition in manager ops soundwire: intel/cadence: expose PING status in manager ops soundwire: add sdw_show_ping_status() helper ASoC: codecs: show PING status on resume failures drivers/soundwire/bus.c | 32 ++++++++++++++++++++++++++++++ drivers/soundwire/cadence_master.c | 8 ++++++++ drivers/soundwire/cadence_master.h | 2 ++ drivers/soundwire/intel.c | 1 + include/linux/soundwire/sdw.h | 5 +++++ sound/soc/codecs/max98373-sdw.c | 2 ++ sound/soc/codecs/rt1308-sdw.c | 2 ++ sound/soc/codecs/rt1316-sdw.c | 2 ++ sound/soc/codecs/rt5682-sdw.c | 2 ++ sound/soc/codecs/rt700-sdw.c | 2 ++ sound/soc/codecs/rt711-sdca-sdw.c | 2 ++ sound/soc/codecs/rt715-sdca-sdw.c | 2 ++ sound/soc/codecs/rt715-sdw.c | 2 ++ 13 files changed, 64 insertions(+) -- 2.25.1

4 10

[PATCH -next] ASoC: codecs: es8326: change es8326_regmap_config to static
by Yang Yingliang 17 Aug '22

17 Aug '22

es8326_regmap_config is only used in es8326.c now, change it to static. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/es8326.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 975302b2a61d..87c1cc16592b 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -207,7 +207,7 @@ static const struct regmap_access_table es8326_volatile_table = { .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), }; -const struct regmap_config es8326_regmap_config = { +static const struct regmap_config es8326_regmap_config = { .reg_bits = 8, .val_bits = 8, .max_register = 0xff, -- 2.25.1

2 1

[PATCH 0/4] Enable timestamp support for SOF compress driver
by Daniel Baluta 17 Aug '22

17 Aug '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries computes pcm_io_frames from the DAI position reported by SOF firmware. Using pcm_io_frames userspace applications can later compute timestamps for compressed stream. Daniel Baluta (2): ASoC: SOF: compress: Remove byte offset computation ASoC: SOF: compress: Introduce sof_compr_stream Laurentiu Mihalcea (2): ASoC: SOF: compress: Save channel count and sample bytes ASoC: SOF: compress: Add support for computing timestamps sound/soc/sof/compress.c | 54 ++++++++++++++++++++++++---------------- sound/soc/sof/sof-priv.h | 7 ++++++ 2 files changed, 40 insertions(+), 21 deletions(-) -- 2.27.0

2 5

[PATCH] ASoC: qcom: qdsp6: q6prm: add new clocks
by Srinivas Kandagatla 17 Aug '22

17 Aug '22

Add support to new clocks that are added in Q6DSP as part of newer version of LPASS support on SM8450 and SC8280XP. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../sound/qcom,q6dsp-lpass-ports.h | 18 ++++++++++++++++++ sound/soc/qcom/qdsp6/q6prm-clocks.c | 9 +++++++++ sound/soc/qcom/qdsp6/q6prm.h | 19 +++++++++++++++++++ 3 files changed, 46 insertions(+) diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h index 0d3276c8fc11..9f7c5103bc82 100644 --- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h +++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h @@ -193,6 +193,24 @@ #define LPASS_CLK_ID_RX_CORE_MCLK 59 #define LPASS_CLK_ID_RX_CORE_NPL_MCLK 60 #define LPASS_CLK_ID_VA_CORE_2X_MCLK 61 +/* Clock ID for MCLK for WSA2 core */ +#define LPASS_CLK_ID_WSA2_CORE_MCLK 62 +/* Clock ID for NPL MCLK for WSA2 core */ +#define LPASS_CLK_ID_WSA2_CORE_2X_MCLK 63 +/* Clock ID for RX Core TX MCLK */ +#define LPASS_CLK_ID_RX_CORE_TX_MCLK 64 +/* Clock ID for RX CORE TX 2X MCLK */ +#define LPASS_CLK_ID_RX_CORE_TX_2X_MCLK 65 +/* Clock ID for WSA core TX MCLK */ +#define LPASS_CLK_ID_WSA_CORE_TX_MCLK 66 +/* Clock ID for WSA core TX 2X MCLK */ +#define LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK 67 +/* Clock ID for WSA2 core TX MCLK */ +#define LPASS_CLK_ID_WSA2_CORE_TX_MCLK 68 +/* Clock ID for WSA2 core TX 2X MCLK */ +#define LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK 69 +/* Clock ID for RX CORE MCLK2 2X MCLK */ +#define LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK 70 #define LPASS_HW_AVTIMER_VOTE 101 #define LPASS_HW_MACRO_VOTE 102 diff --git a/sound/soc/qcom/qdsp6/q6prm-clocks.c b/sound/soc/qcom/qdsp6/q6prm-clocks.c index a26cda5140c1..73b0cbac73d4 100644 --- a/sound/soc/qcom/qdsp6/q6prm-clocks.c +++ b/sound/soc/qcom/qdsp6/q6prm-clocks.c @@ -50,6 +50,15 @@ static const struct q6dsp_clk_init q6prm_clks[] = { Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_MCLK), Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_NPL_MCLK), Q6PRM_CLK(LPASS_CLK_ID_VA_CORE_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK), Q6DSP_VOTE_CLK(LPASS_HW_MACRO_VOTE, Q6PRM_HW_CORE_ID_LPASS, "LPASS_HW_MACRO"), Q6DSP_VOTE_CLK(LPASS_HW_DCODEC_VOTE, Q6PRM_HW_CORE_ID_DCODEC, diff --git a/sound/soc/qcom/qdsp6/q6prm.h b/sound/soc/qcom/qdsp6/q6prm.h index fea4d1954bc1..a988a32086fe 100644 --- a/sound/soc/qcom/qdsp6/q6prm.h +++ b/sound/soc/qcom/qdsp6/q6prm.h @@ -64,6 +64,25 @@ #define Q6PRM_LPASS_CLK_ID_RX_CORE_MCLK 0x30e #define Q6PRM_LPASS_CLK_ID_RX_CORE_NPL_MCLK 0x30f +/* Clock ID for MCLK for WSA2 core */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_MCLK 0x310 +/* Clock ID for NPL MCLK for WSA2 core */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_2X_MCLK 0x311 +/* Clock ID for RX Core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_TX_MCLK 0x312 +/* Clock ID for RX CORE TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_TX_2X_MCLK 0x313 +/* Clock ID for WSA core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA_CORE_TX_MCLK 0x314 +/* Clock ID for WSA core TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK 0x315 +/* Clock ID for WSA2 core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_TX_MCLK 0x316 +/* Clock ID for WSA2 core TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK 0x317 +/* Clock ID for RX CORE MCLK2 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK 0x318 + #define Q6PRM_LPASS_CLK_SRC_INTERNAL 1 #define Q6PRM_LPASS_CLK_ROOT_DEFAULT 0 #define Q6PRM_HW_CORE_ID_LPASS 1 -- 2.21.0

2 1

Re: [PATCH] ASoC: core: remove setting platform_max in kcontrol macros
by Mark Brown 17 Aug '22

17 Aug '22

On Tue, 16 Aug 2022 18:21:29 +0100, Srinivas Kandagatla wrote: > platform_max should not be set by the driver, its intended for machine drivers > or DT to override the max value for platform specific reasons. > > So remove setting this from Kcontrol macros. > > Setting this to max in these macros would limit the range when min > value is less then zero. > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: core: remove setting platform_max in kcontrol macros commit: 26bdcc4ba12351642cd94339aa6996f96434dd47 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

1 0

Re: [PATCH 0/4] ALSA: hda: Minor cleanups
by Cezary Rojewski 17 Aug '22

17 Aug '22

On 2022-08-17 3:23 PM, Cezary Rojewski wrote: > On 2022-08-17 3:11 PM, Amadeusz Sławiński wrote: >> First two patches are small refactoring of code to use poll macros >> instead of open coding register checks. >> Following two patches remove unused defines from code. >> >> Amadeusz Sławiński (4): >> ALSA: hda: Move stream-register polling macros >> ALSA: hda: Rework snd_hdac_stream_reset() to use macros >> ALSA: hda: Remove unused MAX_PIN_CONFIGS constant >> ALSA: hda: Remove unused defines > > > Reviewed by me before sending, plus from the test results patchset > causes no regression in the avs-driver CI. > > Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> RESEND. Patch was blocked so my tag did not make it to the list.

1 0

[RESEND][PATCH] ALSA: info: Fix llseek return value when using callback
by Amadeusz Sławiński 17 Aug '22

17 Aug '22

When using callback there was a flow of ret = -EINVAL if (callback) { offset = callback(); goto out; } ... offset = some other value in case of no callback; ret = offset; out: return ret; which causes the snd_info_entry_llseek() to return -EINVAL when there is callback handler. Fix this by setting "ret" directly to callback return value before jumping to "out". 73029e0ff18d ("ALSA: info - Implement common llseek for binary mode") Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/core/info.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/info.c b/sound/core/info.c index b8058b341178..0b2f04dcb589 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -111,9 +111,9 @@ static loff_t snd_info_entry_llseek(struct file *file, loff_t offset, int orig) entry = data->entry; mutex_lock(&entry->access); if (entry->c.ops->llseek) { - offset = entry->c.ops->llseek(entry, - data->file_private_data, - file, offset, orig); + ret = entry->c.ops->llseek(entry, + data->file_private_data, + file, offset, orig); goto out; } -- 2.25.1

3 4

[PATCH 0/4] ALSA: hda: Minor cleanups
by Amadeusz Sławiński 17 Aug '22

17 Aug '22

First two patches are small refactoring of code to use poll macros instead of open coding register checks. Following two patches remove unused defines from code. Amadeusz Sławiński (4): ALSA: hda: Move stream-register polling macros ALSA: hda: Rework snd_hdac_stream_reset() to use macros ALSA: hda: Remove unused MAX_PIN_CONFIGS constant ALSA: hda: Remove unused defines include/sound/hdaudio.h | 6 ++++++ include/sound/hdaudio_ext.h | 6 ------ sound/hda/hdac_stream.c | 27 +++++++-------------------- sound/pci/hda/hda_intel.c | 7 ------- sound/pci/hda/hda_sysfs.c | 2 -- 5 files changed, 13 insertions(+), 35 deletions(-) -- 2.25.1

1 4

[PATCH] ASoC: Variable type completion
by Xin Gao 17 Aug '22

17 Aug '22

'unsigned int' is better than 'unsigned'. Signed-off-by: Xin Gao <gaoxin(a)cdjrlc.com> --- sound/soc/soc-ac97.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 5f49e3dec3fc..32c5be61e2ec 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -57,7 +57,7 @@ static inline struct snd_soc_component *gpio_to_component(struct gpio_chip *chip return gpio_priv->component; } -static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) +static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned int offset) { if (offset >= AC97_NUM_GPIOS) return -EINVAL; @@ -66,7 +66,7 @@ static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) } static int snd_soc_ac97_gpio_direction_in(struct gpio_chip *chip, - unsigned offset) + unsigned int offset) { struct snd_soc_component *component = gpio_to_component(chip); @@ -75,7 +75,7 @@ static int snd_soc_ac97_gpio_direction_in(struct gpio_chip *chip, 1 << offset, 1 << offset); } -static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) +static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned int offset) { struct snd_soc_component *component = gpio_to_component(chip); int ret; @@ -88,7 +88,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) return !!(ret & (1 << offset)); } -static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, +static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned int offset, int value) { struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); -- 2.30.2

2 1

[PATCH] ASoC: qcom: sm8250: add missing module owner
by Srinivas Kandagatla 17 Aug '22

17 Aug '22

Add missing module owner to able to build and load this driver as module. Fixes: aa2e2785545a ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/sm8250.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index ce4a5713386a..98a2fde9e004 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -270,6 +270,7 @@ static int sm8250_platform_probe(struct platform_device *pdev) if (!card) return -ENOMEM; + card->owner = THIS_MODULE; /* Allocate the private data */ data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); if (!data) -- 2.21.0

2 1

Channel mapping randomly change on usb soundcard and sucpicious dmesg log
by GitHub issues - opened 17 Aug '22

17 Aug '22

alsa-project/alsa-lib issue #261 was opened from sprudent: Hello erveryone, I am getting random channel swap when using speaker test to output sequencially a pink noise on every channel of the soundcard. I tested two distinct usb soundcard getting the same issue. I could identify a suspicious log showing at the same time the channels swapped. Tested and reproduced on a recent archlinux image (see system information). ``` [Aug16 18:26] xhci_hcd 0000:00:14.0: ERROR Transfer event TRB DMA ptr not part of current TD ep_index 11 comp_code 1 [ +0.000134] xhci_hcd 0000:00:14.0: Looking for event-dma 0000000014ed98e0 trb-start 0000000014ed98f0 trb-end 0000000014ed98f0 seg-start 0000000014ed9000 seg-end 0000000014ed9ff0 ``` ### system information ``` uname -r 5.18.7-arch1-1 ``` ``` aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Audio [Intel HDMI/DP LPE Audio], device 0: HdmiLpeAudio [Intel HDMI/DP LPE Audi] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [Intel HDMI/DP LPE Audio], device 1: HdmiLpeAudio [Intel HDMI/DP LPE Audi] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [Intel HDMI/DP LPE Audio], device 2: HdmiLpeAudio [Intel HDMI/DP LPE Audi] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: ICUSBAUDIO7D [ICUSBAUDIO7D], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: bytchtes8316 [bytcht-es8316], device 0: Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: bytchtes8316 [bytcht-es8316], device 1: Deep-Buffer Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/9358483/alsa-… ### Test with dmix #### Description First test was made with alsa dmix plugin. Multiple processes running in parrallel and outputing to dmix pcm device. #### Alsa configuration ``` pcm.play_8 { @args [ CARD ] @args.CARD { type string } type dmix ipc_key 5678 slave { pcm $CARD channels 8 period_time 0 period_size 1024 buffer_size 4096 rate 44100 } bindings { 0 0 1 1 2 2 3 3 4 4 5 5 6 6 7 7 } } ``` #### Step to reproduce Run this script to simulate multiple processes writing to dmix device at the same time ``` for i in $(seq 20) do speaker-test -r44100 -D"play_8:\"hw:1\"" -c8 -t pink & done ``` Then on another tty, start a speaker-test command outputing a sine wave in order to identify channel mapping by correlating speaker test output with audio output. `speaker-test -r44100 -D"play_8:\"hw:1\"" -c8 -t sine` ``` 0 - Front Left 4 - Front Center 1 - Front Right 3 - Side Right 7 - Side Right 2 - Side Left 6 - Side Left 5 - LFE Time per period = 21.998519 ... ``` #### Expected result Audio output to the same channels during all the test #### Actual result Random audio channel swap correlated with dmesg log mentioned above. ### Second test with direct output #### Alsa configuration No specific configuration required #### Step to reproduce Run speaker-following test command and wait for channel swap `speaker-test -r44100 -Dsurround71:ICUSBAUDIO7D -c8` #### Expected result Audio output to the same channels during all the test #### Actual result Random audio channel swap correlated with dmesg log mentioned above. The chanel swap happen far less often than with dmix. ### Interpretation The DMA transfer ring buffer get messed up after a while (see dmesg log) and interleaved audio frames get shifted in soundcard buffer. If I unplug/replug usb soundcard and try again, channel mapping goes back to the correct order. My alsa, kernel and usb driver knowledge is limited, so I come to you for a little help. Tell me if you need more information to reproduce. Thank's, Sylvain Issue URL : https://github.com/alsa-project/alsa-lib/issues/261 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ALSA: hda/cs8409: Support new Dolphin Variants
by Stefan Binding 17 Aug '22

17 Aug '22

Add 4 new Dolphin Systems, same configuration as older systems. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_cs8409-tables.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index e0d3a8be2e38..b288874e401e 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -546,6 +546,10 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0BD6, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0BD7, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0BD8, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C43, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C50, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C51, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C52, "Dolphin", CS8409_DOLPHIN), {} /* terminator */ }; -- 2.34.1

2 1

[PATCH] ALSA: hda/realtek: Add CLSA0101 to serial-multi-instantiate.
by Cameron Berkenpas 17 Aug '22

17 Aug '22

Added CLSA0101 to serial-multi-instantiate.c. Signed-off-by: Cameron Berkenpas <cam(a)neo-zeon.de> --- drivers/platform/x86/serial-multi-instantiate.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/platform/x86/serial-multi-instantiate.c b/drivers/platform/x86/serial-multi-instantiate.c index 1e8063b7c169..bc9bf591dd10 100644 --- a/drivers/platform/x86/serial-multi-instantiate.c +++ b/drivers/platform/x86/serial-multi-instantiate.c @@ -327,8 +327,9 @@ static const struct acpi_device_id smi_acpi_ids[] = { { "BSG2150", (unsigned long)&bsg2150_data }, { "INT3515", (unsigned long)&int3515_data }, { "CSC3551", (unsigned long)&cs35l41_hda }, - /* Non-conforming _HID for Cirrus Logic already released */ + /* Non-conforming _HIDs for Cirrus Logic already released */ { "CLSA0100", (unsigned long)&cs35l41_hda }, + { "CLSA0101", (unsigned long)&cs35l41_hda }, { } }; MODULE_DEVICE_TABLE(acpi, smi_acpi_ids); -- 2.34.1

2 3

Re: [PATCH v2 linux-next] Remove Unnecessary TODO
by Pierre-Louis Bossart 17 Aug '22

17 Aug '22

On 8/16/22 19:59, Khalid Masum wrote: > The capabilities enabled for multi-link are required as part of the > programming sequences, even when a stream uses a single link we still > use the syncArm/syncGo sequences. Therefore the TODO is no longer > necessary. > > Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > Signed-off-by: Khalid Masum <khalid.masum.92(a)gmail.com> the title of your patch should be something like soundwire: intel: remove unnecessary TODO so that the patch scope and context is clearly identified. > --- > Previous discussions: > - v1 Link: https://lore.kernel.org/lkml/20220814080416.7531-1-khalid.masum.92@gmail.co… > > Changes since v1: > - Remove the todo, leave rest of the code as is > - Update commit message > > drivers/soundwire/bus.c | 1 - > 1 file changed, 1 deletion(-) > > diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c > index 8d4000664fa3..18158825a6d8 100644 > --- a/drivers/soundwire/bus.c > +++ b/drivers/soundwire/bus.c > @@ -75,7 +75,6 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, > > /* > * Initialize multi_link flag > - * TODO: populate this flag by reading property from FW node > */ > bus->multi_link = false; > if (bus->ops->read_prop) {

2 1

bat: Add 'readcapture' option to support analyzing external audio
by GitHub pull_request - opened 16 Aug '22

16 Aug '22

alsa-project/alsa-utils pull request #166 was opened from tbird20d: If audio data is captured on another device (ie we are NOT using loopback mode), then allow alsabat to analyze that data, by passing a filename reference on the command line. Add the '--readcapture' option to the argument parser. When this option is specified, avoid doing a local capture, and instead just read the audio data from the indicated file, and analyze that. Signed-off-by: Tim Bird <tim.bird(a)sony.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/166 Patch URL : https://github.com/alsa-project/alsa-utils/pull/166.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Re: Rockchip I2S commit possibly ignored
by Mark Brown 16 Aug '22

16 Aug '22

On Tue, Aug 16, 2022 at 02:54:22PM -0300, Geraldo Nascimento wrote: > On Tue, Aug 16, 2022 at 04:22:37PM +0100, Mark Brown wrote: > > There's absolutely no problem with reposting someone else's patch - just > > add your Signed-off-by at the end of the signoff chain. > Mark, I'm in no position to lecture anyone, least of who, you, hard-working > ASoC maintainer that you are. But there are *tons* of problems with > reposting someone else's patch, even if they had been previously given > the green-light but misteriously vanished. > You are assuming responsibility for someone else's work! Let's see in Oh, sure - but TBH if you're chasing a patch via e-mail you're pretty much going to be in a similar situation if that results in the patch getting applied. A big part of the goal behind getting things reposted is to push them through the normal test/review cycle properly so if there was some reason for it to not get applied the first time around it's more likely that someone will notice than if it's just pulled out of list archives or whatever. > My main point is that without adding "resets" and "reset-names" to *at > least* every Rockchip device tree that enables sound over HDMI, just an > example, you get systems with non-working HDMI. I just tested it, I > omitted "resets" and "reset-names" from my device tree and then had > to SSH into the black screen to revert the changes to my boot partition. > So it's not trivial to RESEND this. It amounts to device tree overhaul > of arch/arm64/boot/dts/rockchip > This would require community effort of say, equivalent to LibreELEC > resources, which are not many, but they have enough patience to test > every proposed change to Rockchip device trees, and could help upstream > this. Right, that's a problem. Even with those changes if we start requiring new properties that'd be an ABI break anyway which isn't something we want for DT - ideally the patch should be reworked so that existing systems continue to work even without DT updates.

1 0

[PATCH] ASoC: Variable type completion
by Xin Gao 16 Aug '22

16 Aug '22

'unsigned int' is better than 'unsigned'. Signed-off-by: Xin Gao <gaoxin(a)cdjrlc.com> --- sound/soc/ti/davinci-evm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c index 68d69e32681a..392bf17564bf 100644 --- a/sound/soc/ti/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -23,7 +23,7 @@ struct snd_soc_card_drvdata_davinci { struct clk *mclk; - unsigned sysclk; + unsigned int sysclk; }; static int evm_startup(struct snd_pcm_substream *substream) @@ -57,7 +57,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_card *soc_card = rtd->card; int ret = 0; - unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *) + unsigned int sysclk = ((struct snd_soc_card_drvdata_davinci *) snd_soc_card_get_drvdata(soc_card))->sysclk; /* set the codec system clock */ -- 2.30.2

1 0

[PATCH] ASoC: Variable type completion
by Xin Gao 16 Aug '22

16 Aug '22

'unsigned int' is better than 'unsigned'. Signed-off-by: Xin Gao <gaoxin(a)cdjrlc.com> --- sound/soc/codecs/pcm3008.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h index f7f4fbbd89db..45270bcf1c02 100644 --- a/sound/soc/codecs/pcm3008.h +++ b/sound/soc/codecs/pcm3008.h @@ -10,10 +10,10 @@ #define __LINUX_SND_SOC_PCM3008_H struct pcm3008_setup_data { - unsigned dem0_pin; - unsigned dem1_pin; - unsigned pdad_pin; - unsigned pdda_pin; + unsigned int dem0_pin; + unsigned int dem1_pin; + unsigned int pdad_pin; + unsigned int pdda_pin; }; #endif -- 2.30.2

1 0

ASoC: snd_soc_info_volsw and platfrom_max
by Srinivas Kandagatla 16 Aug '22

16 Aug '22

Hi Mark, After patch 30ac49841386 (ASoC: ops: Don't modify the driver's plaform_max when reading state) all the controls that are using signed TLV range like below /* -84dB min - 40dB max */ SOC_SINGLE_S8_TLV("RX0 Digital Volume", WCD934X_CDC_RX0_RX_VOL_CTL, -84, 40, digital_gain), reports max value as 40 instead of 124. before this patch the controls max value was calculated considering the min value, but with this patch this calculation has changed resulting in low volume on most of the codecs that are using SOC_SINGLE_S8_TLV. snd_soc_put_volsw does the right thing by considering mc->min, but info_volsw does it differently. Below change fixes the issue for me, but I am bit confused with the first line of this function that calculates max value as max = mc->max - mc->min and then limits it to platform_max. diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index bd88de056358..49fb34609202 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -196,7 +196,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = max; + uinfo->value.integer.max = max - mc->min; return 0; } Or should we fix the macro to set platform_max to be max - min. thanks, Srini

2 5

[PATCH 1/2] ASoC: Intel: common: add ACPI matching tables for Raptor Lake
by Kai Vehmanen 16 Aug '22

16 Aug '22

Initial support for RPL w/ RT711 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Gopal Vamshi Krishna <vamshi.krishna.gopal(a)intel.com> --- include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 2 +- .../intel/common/soc-acpi-intel-rpl-match.c | 51 +++++++++++++++++++ 3 files changed, 54 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-rpl-match.c diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index bc7fd46ec2bc..ac750afa7bc6 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -30,6 +30,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_sdw_machines[]; @@ -38,6 +39,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[]; /* diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 8ca8f872ec80..41054cf09ec9 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -9,7 +9,7 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-cml-match.o soc-acpi-intel-icl-match.o \ soc-acpi-intel-tgl-match.o soc-acpi-intel-ehl-match.o \ soc-acpi-intel-jsl-match.o soc-acpi-intel-adl-match.o \ - soc-acpi-intel-mtl-match.o \ + soc-acpi-intel-rpl-match.o soc-acpi-intel-mtl-match.o \ soc-acpi-intel-hda-match.o \ soc-acpi-intel-sdw-mockup-match.o diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c new file mode 100644 index 000000000000..0b77401e4e6f --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -0,0 +1,51 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * soc-apci-intel-rpl-match.c - tables and support for RPL ACPI enumeration. + * + * Copyright (c) 2022 Intel Corporation. + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> + +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { + { + .adr = 0x000020025D071100ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + +static const struct snd_soc_acpi_link_adr rpl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_0_adr), + .adr_d = rt711_0_adr, + }, + {} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines); + +/* this table is used when there is no I2S codec present */ +struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[] = { + { + .link_mask = 0x1, /* link0 required */ + .links = rpl_rvp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-rpl-rt711.tplg", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_sdw_machines); base-commit: efe30e2cb6ff87467389cece8ce604f6601059f9 -- 2.37.2

2 2

[PATCH 0/4] ASoC: Intel: HSW and BDW updates
by Cezary Rojewski 16 Aug '22

16 Aug '22

Sharing this PR as it touches on cross-driver subjects. Four commits yet two subject. Given the small delta, decided to combine within single PR here. I plan to send these subjects separately to the mailing list though. 1. Rename hsw_rt5640 and bdw_rt286 module names 2. Drop SND_SOC_INTEL_ACPI_MATCH dependency for catpt-driver In regard to 1), this is a followup to discussion that has already happened on upstream [1]. Goal is to align kernel module names for hsw and bdw boards with their siblings found in /intel/boards and /intel/avs/boards. In regard to 2), this is yet another followup [2]. Given the fact that /intel/avs/boards are already their on upstream, intention is to align catpt-driver to approach found in the avs-driver. This is also my suggestion for existing sof-boards: most should be listed under /sof/intel/boards/ as they're not common, really - they're are SOF specific. I believe that in time /intel/common and /intel/boards should be either limited or completely removed. As subject touches code used by the sof-driver, additional review has been conducted on thesofproject/linux [3]. [1]: https://lore.kernel.org/alsa-devel/54fcdedc-278c-06b8-5402-28880597d3f4@int… [2]: https://lore.kernel.org/alsa-devel/d9fd60e0-a157-bda7-a918-5548918d1552@int… [3]: https://github.com/thesofproject/linux/pull/3799 Cezary Rojewski (4): ASoC: Intel: hsw_rt5640: Rename module ASoC: Intel: bdw_rt286: Rename module ASoC: Intel: catpt: Drop SND_SOC_ACPI_INTEL_MATCH dependency ASoC: Intel: Drop legacy HSW/BDW board-match information include/sound/soc-acpi-intel-match.h | 1 - sound/soc/intel/Kconfig | 2 +- sound/soc/intel/boards/Makefile | 8 ++--- sound/soc/intel/catpt/device.c | 33 +++++++++++++++++-- .../common/soc-acpi-intel-hsw-bdw-match.c | 15 --------- 5 files changed, 35 insertions(+), 24 deletions(-) -- 2.25.1

2 6

Rockchip I2S commit possibly ignored
by Geraldo Nascimento 16 Aug '22

16 Aug '22

Hi Mark, I was looking at Rockchip I2S commits and it seems "ASoC: rockchip: i2s: Reset the controller if soft reset failed" was supposed to have been merged to your sound.git but never was. I don't know if this was intentional or a real miss but in any case I'm letting you know. According to 515b436be291ff197c52198282bbb19e79c9d197 'Merge series "Patches to update for rockchip i2s" from Sugar Zhang <sugar.zhang(a)rock-chips.com>:' it should have been merged to your tree. In rockchip-i2s.yaml there's even documentation that references the missing commit. However in the alsa-devel archives, https://mailman.alsa-project.org/pipermail/alsa-devel/2021-August/189050.ht… I see there was no commit info for the unmerged patch. Perhaps this caused it to be black-holed? Thanks, Geraldo Nascimento

2 4

Re: [PATCH v6 1/2] ASoC: dt-bindings: Add Everest ES8326 audio CODEC
by Mark Brown 16 Aug '22

16 Aug '22

On Tue, 16 Aug 2022 10:44:55 +0800, Zhu Ning wrote: > Add device tree binding documentation for Everest ES8326 > > ---- > v5 tested by dtschema > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/2] ASoC: dt-bindings: Add Everest ES8326 audio CODEC commit: 8c6789f4e2d4ee7d6c8c60daa88ea7a4c4cf6779 [2/2] ASoC: codecs: add support for ES8326 commit: 5c439937775d77a334696a98fb2a25dee72ffa2d 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

1 0

[RESEND v4 0/2] Add support for linked list to store acp_stream and tdm support.
by Venkata Prasad Potturu 16 Aug '22

16 Aug '22

This patch is to add support for linked list to store acp_stream instead static array and add tdm support for acp I2S stream. Ajit Kumar Pandey (1): ASoC: amd: acp: Initialize list to store acp_stream during pcm_open Venkata Prasad Potturu (1): ASoC: amd: acp: Add TDM support for acp i2s stream Changes since v3: -- Modify to spin_lock() in irq handler and spin_lock_irq() in dma open and close functions instead of spin_lock_irqsave(). -- Add tdm support patch to this series. Changes since v2: -- Add locking mechanism in irq handler, linked list updatation and deletion. Changes since v1: -- Fix compile error and remove unused variable. sound/soc/amd/acp/acp-i2s.c | 80 +++++++++++++++++++++++++++++++- sound/soc/amd/acp/acp-platform.c | 40 ++++++++-------- sound/soc/amd/acp/amd.h | 16 ++++++- 3 files changed, 113 insertions(+), 23 deletions(-) -- 2.25.1

2 3

[PATCH v5 1/2] ASoC: dt-bindings: Add Everest ES8326 audio CODEC
by Zhu Ning 16 Aug '22

16 Aug '22

Add device tree binding documentation for Everest ES8326 Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ---- v5 tested by dtschema --- .../bindings/sound/everest,es8326.yaml | 116 ++++++++++++++++++ 1 file changed, 116 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8326.yaml diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml new file mode 100755 index 000000000000..07781408e788 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8326.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8326 audio CODEC + +maintainers: + - David Yang <yangxiaohua(a)everest-semi.com> + +properties: + compatible: + const: everest,es8326 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + + everest,jack-pol: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + just the value of reg 57. Bit(3) decides whether the jack polarity is inverted. + Bit(2) decides whether the button on the headset is inverted. + Bit(1)/(0) decides the mic properity to be OMTP/CTIA or auto. + minimum: 0x00 + maximum: 0x0f + default: 0x0f + + everest,mic1-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset plugged. + minimum: 0x00 + maximum: 0x77 + default: 0x22 + + everest,mic2-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset unplugged. + minimum: 0x00 + maximum: 0x77 + default: 0x44 + + everest,jack-detect-inverted: + $ref: /schemas/types.yaml#/definitions/flag + description: + Defined to invert the jack detection. + + everest,interrupt-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x58, Defines the interrupt source. + Bit(2) 1 means button press triggers irq, 0 means not. + Bit(3) 1 means PIN9 is the irq source for jack detection. When set to 0, + bias change on PIN9 do not triggers irq. + Bit(4) 1 means PIN27 is the irq source for jack detection. + Bit(5) 1 means PIN9 is the irq source after MIC detect. + Bit(6) 1 means PIN27 is the irq source after MIC detect. + minimum: 0 + maximum: 0x3c + default: 0x08 + + everest,interrupt-clk: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x59, Defines the interrupt output behavior. + Bit(0-3) 0 means irq pulse equals 512*internal clock + 1 means irq pulse equals 1024*internal clock + 2 means ... + 7 means irq pulse equals 65536*internal clock + 8 means irq mutes PA + 9 means irq mutes PA and DAC output + Bit(4) 1 means we invert the interrupt output. + Bit(6) 1 means the chip do not detect jack type after button released. + 0 means the chip detect jack type again after button released. + minimum: 0 + maximum: 0x7f + default: 0x45 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8326: codec@19 { + compatible = "everest,es8326"; + reg = <0x19>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + everest,mic1-src = [22]; + everest,mic2-src = [44]; + everest,jack-pol = [0e]; + everest,interrupt-src = [08]; + everest,interrupt-clk = [45]; + }; + }; -- 2.36.1

4 5

[PATCH v2] ASoC: fsl_sai: fix incorrect mclk number in error message
by Pieterjan Camerlynck 16 Aug '22

16 Aug '22

In commit c3ecef21c3f26 ("ASoC: fsl_sai: add sai master mode support") the loop was changed to start iterating from 1 instead of 0. The error message however was not updated, reporting the wrong clock to the user. Signed-off-by: Pieterjan Camerlynck <pieterjan.camerlynck(a)gmail.com> --- V2: rebase against latest version --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7523bb944b21..d430eece1d6b 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1306,7 +1306,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } -- 2.25.1

3 2

[PATCH] ALSA: hda/realtek: Add quirk for Lenovo Yoga7 14IAL7
by Takashi Iwai 16 Aug '22

16 Aug '22

Lenovo Yoga7 14IAL7 requires the same quirk as Lenovo Yoga9 14IAP7 for fixing the bass speaker problems. Reported-by: Pascal Gross <baipush(a)tutanota.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/N9_CjBz--3-2@tutanota.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 f544761eb11b..b42496c01c43 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9492,6 +9492,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), + SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.35.3

1 0

[GIT PULL] ASoC fixes for v6.0-rc1
by Mark Brown 16 Aug '22

16 Aug '22

The following changes since commit 40d060b8158e26438398bf1132925f666e3b6480: ASoC: q6asm: use kcalloc() instead of kzalloc() (2022-07-28 11:59:10 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.0-rc1 for you to fetch changes up to b4b5f29a076e52181f63e45a2ad1bc88593072e3: ASoC: codec: tlv320aic32x4: fix mono playback via I2S (2022-08-10 14:52:05 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.0 A relatively large batch of fixes that came in since my pull request, none of them too major and mostly device specific apart from a series of security/robustness improvements from Takashi. ---------------------------------------------------------------- Biju Das (1): ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path Justin Stitt (1): ASoC: SOF: ipc3-topology: Fix clang -Wformat warning Mark Brown (2): ASoC: Fix theoretical buffer overflow by snprintf() TAS2770 fixes Martin Povišer (4): ASoC: tas2770: Set correct FSYNC polarity ASoC: tas2770: Allow mono streams ASoC: tas2770: Drop conflicting set_bias_level power setting ASoC: tas2770: Fix handling of mute/unmute Oder Chiou (1): ASoC: rt5640: Fix the JD voltage dropping issue Philipp Zabel (1): ASoC: codec: tlv320aic32x4: fix mono playback via I2S Pierre-Louis Bossart (1): ASoC: Intel: fix sof_es8336 probe Takashi Iwai (4): ASoC: Intel: avs: Fix potential buffer overflow by snprintf() ASoC: SOF: debug: Fix potential buffer overflow by snprintf() ASoC: SOF: Intel: hda: Fix potential buffer overflow by snprintf() ASoC: DPCM: Don't pick up BE without substream syed sabakareem (1): ASoC: amd: yc: Update DMI table entries for AMD platforms sound/soc/amd/yc/acp6x-mach.c | 28 +++++++++++ sound/soc/codecs/rt5640.c | 5 +- sound/soc/codecs/tas2770.c | 98 ++++++++++++++++--------------------- sound/soc/codecs/tas2770.h | 5 ++ sound/soc/codecs/tlv320aic32x4.c | 9 ++++ sound/soc/intel/avs/pcm.c | 4 +- sound/soc/intel/boards/sof_es8336.c | 4 +- sound/soc/sh/rz-ssi.c | 26 +++++----- sound/soc/soc-pcm.c | 3 ++ sound/soc/sof/debug.c | 6 +-- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/ipc3-topology.c | 2 +- 12 files changed, 116 insertions(+), 76 deletions(-)

2 1

[PATCH v2] ALSA: hda: cs35l41: Clarify support for CSC3551 without _DSD Properties
by Stefan Binding 16 Aug '22

16 Aug '22

For devices which use HID CSC3551, correct ACPI _DSD properties are required to be able support those systems. Add error message to clarify this. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 129bffb431c2..15e2a0009080 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1163,6 +1163,11 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; hw_cfg->gpio1.valid = true; } else { + /* + * Note: CLSA010(0/1) are special cases which use a slightly different design. + * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported. + */ + dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.\n", hid); hw_cfg->valid = false; hw_cfg->gpio1.valid = false; hw_cfg->gpio2.valid = false; -- 2.34.1

2 1

[PATCH v2 0/6] ALSA: hda: Unify codec construction
by Cezary Rojewski 16 Aug '22

16 Aug '22

A follow up to the recent HDAudio fixes series [1]. Given the recently reported regression [2], before the page fault occurring on codec shutdown can be fixed, codec construction procedure needs to be updated for skylake and sof-intel drivers. Drivers: pci-hda and avs need no changes - already making use of snd_hda_codec_device_init(). As subject touches code used by the sof-driver, additional review has been conducted on thesofproject/linux [3]. Changes in v2: - dropped snd_hda_ext_core <-> snd_hda_codec dependency by calling snd_hda_codec_device_init() directly in skylake and sof drivers probe enumeration routines, as suggested by Takashi - skylake/sof portion of the change has been split into two separate patches - new functions that aim to replace hdac_ext codec init & exit functionality are added first - for skylake and sof drivers both - third patch in the series now combines the "field -> pointer" change for hdac_hda_priv->codec plus the codec-enumeration adjustments for skylake and sof drivers Both above are here to keep git bisect happy, as suggested by Pierre [1]: https://lore.kernel.org/alsa-devel/20220706120230.427296-7-cezary.rojewski@… [2]: https://lore.kernel.org/alsa-devel/3c40df55-3aee-1e08-493b-7b30cd84dc00@lin… [3]: https://github.com/thesofproject/linux/pull/3775 Cezary Rojewski (6): ASoC: Intel: Skylake: Introduce HDA codec init and exit routines ASoC: SOF: Intel: Introduce HDA codec init and exit routines ASoC: Intel: Drop hdac_ext usage for codec device creation ALSA: hda: Always free codec on the device release ALSA: hda: Remove codec init and exit routines ALSA: hda: Fix page fault in snd_hda_codec_shutdown() include/sound/hda_codec.h | 2 - include/sound/hdaudio_ext.h | 3 -- sound/hda/ext/hdac_ext_bus.c | 53 ------------------- sound/pci/hda/hda_codec.c | 49 ++++++++--------- sound/soc/codecs/hdac_hda.c | 26 ++++----- sound/soc/codecs/hdac_hda.h | 2 +- sound/soc/intel/boards/hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/skylake/skl.c | 53 ++++++++++++++----- sound/soc/sof/intel/hda-codec.c | 55 ++++++++++++++------ 10 files changed, 113 insertions(+), 134 deletions(-) -- 2.25.1

4 10

[PATCH] ASoC: Fix the include guard used for include/sound/wm8904.h
by Christophe JAILLET 16 Aug '22

16 Aug '22

__MFD_WM8994_PDATA_H__ is already used for: include/linux/mfd/wm8994/pdata.h Based on file names, use __MFD_WM8904_PDATA_H__ instead here. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Not sure of if a Fixes: tag is needed. If yes, I would use: 9e50108668a7 ("mfd: Add initial WM8994 support") which introduced the conflict, even if: a91eb199e4dc8 ("ASoC: Initial WM8904 CODEC driver") which introduced the typo is more logical. No strong opinion on it. --- include/sound/wm8904.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h index 88ac1870510e..f74293a83a17 100644 --- a/include/sound/wm8904.h +++ b/include/sound/wm8904.h @@ -7,8 +7,8 @@ * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> */ -#ifndef __MFD_WM8994_PDATA_H__ -#define __MFD_WM8994_PDATA_H__ +#ifndef __MFD_WM8904_PDATA_H__ +#define __MFD_WM8904_PDATA_H__ /* Used to enable configuration of a GPIO to all zeros */ #define WM8904_GPIO_NO_CONFIG 0x8000 -- 2.34.1

2 1

[PATCH] ASoC: intel: Replace scnprintf() calls with sysfs_emit_at()
by zhaoxiao 16 Aug '22

16 Aug '22

For sysfs outputs, it's safer to use a new helper, sysfs_emit(), instead of the raw sprintf() & co. This patch replaces the open-code with a new helper, sysfs_emit_at(), by passing the string offset. Signed-off-by: zhaoxiao <zhaoxiao(a)uniontech.com> --- sound/soc/sof/intel/hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 6d4ecbe14adf..917d639671a7 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -574,7 +574,7 @@ static void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev, const char *le chip = get_chip_info(sdev->pdata); for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) { value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + i * 0x4); - len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value); + len += sysfs_emit_at(msg, len, " 0x%x", value); } dev_printk(level, sdev->dev, "extended rom status: %s", msg); -- 2.20.1

1 0

[PATCH] ASoC: Intel: sof_rt5682: Add support for jsl_rt5682_rt1019
by Sean Hong 16 Aug '22

16 Aug '22

This patch adds the driver data for rt5682 support jsl_rt5682_rt1019. Signed-off-by: Sean Hong <sean.hong(a)quanta.corp-partner.google.com> --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 12 ++++++++++++ 2 files changed, 21 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 045965312..3a840f3a9 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1100,6 +1100,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1019", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1019_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index b95c4b2cd..e374bf163 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -29,6 +29,11 @@ static const struct snd_soc_acpi_codecs rt1015p_spk = { .codecs = {"RTL1015"} }; +static struct snd_soc_acpi_codecs rt1019p_spk = { + .num_codecs = 1, + .codecs = {"RTL1019"} +}; + static const struct snd_soc_acpi_codecs mx98360a_spk = { .num_codecs = 1, .codecs = {"MX98360A"} @@ -78,6 +83,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg", }, + { + .comp_ids = &rt5682_rt5682s_hp, + .drv_name = "jsl_rt5682_rt1019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1019p_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1019.tplg", + }, { .id = "10134242", .drv_name = "jsl_cs4242_mx98360a", -- 2.25.1

2 1

Re: [PATCH 07/13] ASoC: amd: add acp6.2 pdm driver dma ops
by Syed Saba Kareem 16 Aug '22

16 Aug '22

On 8/12/22 19:50, Amadeusz Sławiński wrote: > [CAUTION: External Email] > > On 8/12/2022 2:07 PM, Syed Saba kareem wrote: >> This patch adds PDM driver DMA operations for Pink Sardine Platform. >> >> Signed-off-by: Syed Saba Kareem <Syed.SabaKareem(a)amd.com> >> --- >> sound/soc/amd/ps/acp62.h | 41 +++++ >> sound/soc/amd/ps/ps-pdm-dma.c | 310 ++++++++++++++++++++++++++++++++++ >> 2 files changed, 351 insertions(+) >> >> diff --git a/sound/soc/amd/ps/acp62.h b/sound/soc/amd/ps/acp62.h >> index 563252834b09..525e8bd225a2 100644 >> --- a/sound/soc/amd/ps/acp62.h >> +++ b/sound/soc/amd/ps/acp62.h >> @@ -27,6 +27,31 @@ >> #define ACP_EXT_INTR_STAT_CLEAR_MASK 0xFFFFFFFF >> #define PDM_DMA_STAT 0x10 >> >> +#define PDM_DMA_INTR_MASK 0x10000 >> +#define ACP_ERROR_STAT 29 >> +#define PDM_DECIMATION_FACTOR 2 >> +#define ACP_PDM_CLK_FREQ_MASK 7 >> +#define ACP_WOV_MISC_CTRL_MASK 0x10 >> +#define ACP_PDM_ENABLE 1 >> +#define ACP_PDM_DISABLE 0 >> +#define ACP_PDM_DMA_EN_STATUS 2 >> +#define TWO_CH 2 >> +#define DELAY_US 5 >> +#define ACP_COUNTER 20000 >> + >> +#define ACP_SRAM_PTE_OFFSET 0x03800000 >> +#define PAGE_SIZE_4K_ENABLE 2 >> +#define PDM_PTE_OFFSET 0 >> +#define PDM_MEM_WINDOW_START 0x4000000 >> + >> +#define CAPTURE_MIN_NUM_PERIODS 4 >> +#define CAPTURE_MAX_NUM_PERIODS 4 >> +#define CAPTURE_MAX_PERIOD_SIZE 8192 >> +#define CAPTURE_MIN_PERIOD_SIZE 4096 >> + >> +#define MAX_BUFFER (CAPTURE_MAX_PERIOD_SIZE * CAPTURE_MAX_NUM_PERIODS) >> +#define MIN_BUFFER MAX_BUFFER >> + >> enum acp_config { >> ACP_CONFIG_0 = 0, >> ACP_CONFIG_1, >> @@ -46,6 +71,22 @@ enum acp_config { >> ACP_CONFIG_15, >> }; >> >> +struct pdm_stream_instance { >> + u16 num_pages; >> + u16 channels; >> + dma_addr_t dma_addr; >> + u64 bytescount; >> + void __iomem *acp62_base; >> +}; >> + >> +union acp_pdm_dma_count { >> + struct { >> + u32 low; >> + u32 high; >> + } bcount; > > Fix indentation. > Also you can remove struct name, and access fields directly, so instead > of accessing somevariable.bcount.low, you would use somevariable.low, it > would probably be more readable. > Okay Will fix it. >> + u64 bytescount; >> +}; >> + >> struct pdm_dev_data { >> u32 pdm_irq; >> void __iomem *acp62_base; >> diff --git a/sound/soc/amd/ps/ps-pdm-dma.c >> b/sound/soc/amd/ps/ps-pdm-dma.c >> index bca2ac3e81f5..a98a2015015d 100644 > > ... > >> + >> +static int acp62_pdm_dma_open(struct snd_soc_component *component, >> + struct snd_pcm_substream *substream) >> +{ >> + struct snd_pcm_runtime *runtime; >> + struct pdm_dev_data *adata; >> + struct pdm_stream_instance *pdm_data; >> + int ret; >> + >> + runtime = substream->runtime; >> + adata = dev_get_drvdata(component->dev); >> + pdm_data = kzalloc(sizeof(*pdm_data), GFP_KERNEL); >> + if (!pdm_data) >> + return -EINVAL; >> + >> + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) >> + runtime->hw = acp62_pdm_hardware_capture; >> + >> + ret = snd_pcm_hw_constraint_integer(runtime, >> + SNDRV_PCM_HW_PARAM_PERIODS); >> + if (ret < 0) { >> + dev_err(component->dev, "set integer constraint >> failed\n"); >> + kfree(pdm_data); >> + return ret; >> + } >> + >> + acp62_enable_pdm_interrupts(adata->acp62_base); >> + >> + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) >> + adata->capture_stream = substream; >> + >> + pdm_data->acp62_base = adata->acp62_base; >> + runtime->private_data = pdm_data; > > Should probably kfree(runtime->private_data) in acp62_pdm_dma_close(), > otherwise won't there be a memory leak? > Will fix it. >> + return ret; >> +} >> + > > ... >

1 0

[PATCH 00/13] Add Pink Sardine platform ASoC driver
by Syed Saba kareem 16 Aug '22

16 Aug '22

Pink Sardine platform is new APU series based on acp6.2 design. This patch set adds an ASoC driver for the ACP (Audio CoProcessor) block on AMD Pink Sardine APU with DMIC endpoint support. Syed Saba kareem (13): ASoC: amd: add Pink Sardine platform ACP IP register header ASoC: amd: add Pink Sardine ACP PCI driver ASoC: amd: add acp6.2 init/de-init functions ASoC: amd: add platform devices for acp6.2 pdm driver and dmic driver ASoC: amd: add acp6.2 pdm platform driver ASoC: amd: add acp6.2 irq handler ASoC: amd: add acp6.2 pdm driver dma ops ASoC: amd: add acp6.2 pci driver pm ops ASoC: amd: add acp6.2 pdm driver pm ops ASoC: amd: enable Pink Sardine acp6.2 drivers build ASoC: amd: create platform device for acp6.2 machine driver ASoC: amd: add Pink Sardine machine driver using dmic ASoC: amd: enable Pink sardine platform machine driver build. include/sound/acp62_chip_offset_byte.h | 444 ++++++++++++++++++++++++ sound/soc/amd/Kconfig | 21 ++ sound/soc/amd/Makefile | 1 + sound/soc/amd/ps/Makefile | 9 + sound/soc/amd/ps/acp62.h | 107 ++++++ sound/soc/amd/ps/pci-ps.c | 356 ++++++++++++++++++++ sound/soc/amd/ps/ps-mach.c | 79 +++++ sound/soc/amd/ps/ps-pdm-dma.c | 449 +++++++++++++++++++++++++ 8 files changed, 1466 insertions(+) create mode 100644 include/sound/acp62_chip_offset_byte.h create mode 100644 sound/soc/amd/ps/Makefile create mode 100644 sound/soc/amd/ps/acp62.h create mode 100644 sound/soc/amd/ps/pci-ps.c create mode 100644 sound/soc/amd/ps/ps-mach.c create mode 100644 sound/soc/amd/ps/ps-pdm-dma.c -- 2.25.1

4 22

6.0-RC1 new module for lenovo yoga
by Pascal Gross 16 Aug '22

16 Aug '22

Hello everybody I've seen that there's a module in the new Kernel-RC for bass speaker problems with new Lenovo (written by Philipp Jungkamp): SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN) Could you please add the exact same line for a similar model with the same Problem? SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA7_14IAL7_BASS_SPK_PIN) I'm neither a developer nor a coder so I don't know how to contribute to the Kernel, I just played around with Philipps patch, changed the lines according to my model and it works now. I would really appreciate your help and of course, if you have any questions feel free to contact me. Thanks in advance, Pascal

1 0

UAC gadget audio sample rates
by Matt Flax 16 Aug '22

16 Aug '22

Hi there, Is it possible to have a USB audio gadget which supports multiple/arbitrary sample rates ? Matt

1 0

[PATCH v6 2/2] ASoC: codecs: add support for ES8326
by Zhu Ning 16 Aug '22

16 Aug '22

The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ------ v6 remove rate 96000 remove HEX suffix --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8326.c | 905 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8326.h | 182 ++++++++ 4 files changed, 1094 insertions(+) create mode 100755 sound/soc/codecs/es8326.c create mode 100755 sound/soc/codecs/es8326.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d16b4efb88a7..9465992c2d68 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -913,6 +914,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 92fd441d426a..81e00863d992 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -455,6 +456,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..975302b2a61d --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,905 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua(a)everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct delayed_work button_press_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling or during an irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + u8 interrupt_src; + u8 interrupt_clk; + bool jd_inverted; + unsigned int sysclk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL, ES8326_ADC2_VOL, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PGAGAIN, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL, + 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL, + 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DETECT, ES8326_HP_DETECT), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFP: + snd_soc_component_update_bits(component, ES8326_RESET, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_component_update_bits(component, ES8326_FMT, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR, + coeff_div[coeff].regb); + } else { + dev_warn(component->dev, "Clock coefficients do not match"); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + regmap_write(es8326->regmap, ES8326_RESET, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x11); + regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; + +static void es8326_enable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_disable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +/* + * For button detection, set the following in soundcard + * snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + * snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + * snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + */ +static void es8326_jack_button_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, button_press_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + static int button_to_report, press_count; + static int prev_button, cur_button; + + if (!(es8326->jack->status & SND_JACK_HEADSET)) /* Jack unplugged */ + return; + + mutex_lock(&es8326->lock); + iface = snd_soc_component_read(comp, ES8326_HP_DETECT); + switch (iface) { + case 0x93: + /* pause button detected */ + cur_button = SND_JACK_BTN_0; + break; + case 0x6f: + /* button volume up */ + cur_button = SND_JACK_BTN_1; + break; + case 0x27: + /* button volume down */ + cur_button = SND_JACK_BTN_2; + break; + case 0x1e: + /* button released or not pressed */ + cur_button = 0; + break; + default: + break; + } + + if ((prev_button == cur_button) && (cur_button != 0)) { + press_count++; + if (press_count > 10) { + /* report a press every 500ms */ + snd_soc_jack_report(es8326->jack, cur_button, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + press_count = 0; + } + button_to_report = cur_button; + queue_delayed_work(system_wq, &es8326->button_press_work, + msecs_to_jiffies(50)); + } else if (prev_button != cur_button) { + /* mismatch, detect again */ + prev_button = cur_button; + queue_delayed_work(system_wq, &es8326->button_press_work, + msecs_to_jiffies(50)); + } else { + /* released or no pressed */ + if (button_to_report != 0) { + snd_soc_jack_report(es8326->jack, button_to_report, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + snd_soc_jack_report(es8326->jack, 0, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + button_to_report = 0; + } + } + mutex_unlock(&es8326->lock); +} + +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + mutex_lock(&es8326->lock); + iface = snd_soc_component_read(comp, ES8326_HP_DETECT); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + /* Jack unplugged or spurious IRQ */ + dev_dbg(comp->dev, "No headset detected"); + if (es8326->jack->status & SND_JACK_HEADPHONE) { + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC, es8326->mic2_src); + es8326_disable_micbias(comp); + } + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if (es8326->jack->status & SND_JACK_HEADSET) { + /* detect button */ + queue_delayed_work(system_wq, &es8326->button_press_work, 10); + } else { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, + SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, + ES8326_ADC1_SRC, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, + SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } + } + mutex_unlock(&es8326->lock); +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + if (!es8326->jack) + goto out; + + es8326_enable_micbias(comp); + + if (es8326->jack->status & SND_JACK_HEADSET) + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(10)); + else + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + +out: + return IRQ_HANDLED; +} + +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + unsigned int reg; + + regcache_cache_only(es8326->regmap, false); + regcache_sync(es8326->regmap); + + regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, es8326->interrupt_src); + regmap_write(es8326->regmap, ES8326_INTOUT_IO, es8326->interrupt_clk); + regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PGAGAIN, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + + regmap_read(es8326->regmap, ES8326_CHIP_VERSION, &reg); + if ((reg & ES8326_VERSION_B) == 1) { + regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xDD); + regmap_write(es8326->regmap, ES8326_ANA_VSEL, 0x7F); + regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x0F); + /* enable button detect */ + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xA0); + } + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_suspend(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + cancel_delayed_work_sync(&es8326->jack_detect_work); + es8326_disable_micbias(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); + regcache_cache_only(es8326->regmap, true); + regcache_mark_dirty(es8326->regmap); + + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + ret = device_property_read_u8(component->dev, "everest,interrupt-src", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "interrupt-src return %d", ret); + es8326->interrupt_src = ES8326_HP_DET_SRC_PIN9; + } + dev_dbg(component->dev, "interrupt-src %x", es8326->interrupt_src); + + ret = device_property_read_u8(component->dev, "everest,interrupt-clk", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "interrupt-clk return %d", ret); + es8326->interrupt_clk = 0x45; + } + dev_dbg(component->dev, "interrupt-clk %x", es8326->interrupt_clk); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) { + es8326_disable_micbias(component); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + } + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .suspend = es8326_suspend, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + INIT_DELAYED_WORK(&es8326->button_press_work, + es8326_jack_button_handler); + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) { + dev_warn(&i2c->dev, "Failed to request IRQ: %d: %d\n", + es8326->irq, ret); + es8326->irq = -ENXIO; + } + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, + .id_table = es8326_i2c_id, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..8e5ffe5ee10d --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,182 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.h -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ +#define ES8326_RESET 0x00 +#define ES8326_CLK_CTL 0x01 +#define ES8326_CLK_INV 0x02 +#define ES8326_CLK_RESAMPLE 0x03 +#define ES8326_CLK_DIV1 0x04 +#define ES8326_CLK_DIV2 0x05 +#define ES8326_CLK_DLL 0x06 +#define ES8326_CLK_MUX 0x07 +#define ES8326_CLK_ADC_SEL 0x08 +#define ES8326_CLK_DAC_SEL 0x09 +#define ES8326_CLK_ADC_OSR 0x0a +#define ES8326_CLK_DAC_OSR 0x0b +#define ES8326_CLK_DIV_CPC 0x0c +#define ES8326_CLK_DIV_BCLK 0x0d +#define ES8326_CLK_TRI 0x0e +#define ES8326_CLK_DIV_LRCK 0x0f +#define ES8326_CLK_VMIDS1 0x10 +#define ES8326_CLK_VMIDS2 0x11 +#define ES8326_CLK_CAL_TIME 0x12 +#define ES8326_FMT 0x13 + +#define ES8326_DAC_MUTE 0x14 +#define ES8326_ADC_MUTE 0x15 +#define ES8326_ANA_PDN 0x16 +#define ES8326_PGA_PDN 0x17 +#define ES8326_VMIDSEL 0x18 +#define ES8326_ANA_LP 0x19 +#define ES8326_ANA_DMS 0x1a +#define ES8326_ANA_MICBIAS 0x1b +#define ES8326_ANA_VSEL 0x1c +#define ES8326_SYS_BIAS 0x1d +#define ES8326_BIAS_SW1 0x1e +#define ES8326_BIAS_SW2 0x1f +#define ES8326_BIAS_SW3 0x20 +#define ES8326_BIAS_SW4 0x21 +#define ES8326_VMIDLOW 0x22 +#define ES8326_PGAGAIN 0x23 +#define ES8326_HP_DRIVER 0x24 +#define ES8326_DAC2HPMIX 0x25 +#define ES8326_HP_VOL 0x26 +#define ES8326_HP_CAL 0x27 +#define ES8326_HP_DRIVER_REF 0x28 +#define ES8326_ADC_SCALE 0x29 +#define ES8326_ADC1_SRC 0x2a +#define ES8326_ADC2_SRC 0x2b +#define ES8326_ADC1_VOL 0x2c +#define ES8326_ADC2_VOL 0x2d +#define ES8326_ADC_RAMPRATE 0x2e +#define ES8326_ALC_RECOVERY 0x32 +#define ES8326_ALC_LEVEL 0x33 +#define ES8326_ADC_HPFS1 0x34 +#define ES8326_ADC_HPFS2 0x35 +#define ES8326_ADC_EQ 0x36 +#define ES8326_HP_OFFSET_CAL 0x4A +#define ES8326_HPL_OFFSET_INI 0x4B +#define ES8326_HPR_OFFSET_INI 0x4C +#define ES8326_DAC_DSM 0x4D +#define ES8326_DAC_RAMPRATE 0x4E +#define ES8326_DAC_VPPSCALE 0x4F +#define ES8326_DAC_VOL 0x50 +#define ES8326_DRC_RECOVERY 0x53 +#define ES8326_DRC_WINSIZE 0x54 +#define ES8326_HPJACK_TIMER 0x56 +#define ES8326_HP_DET 0x57 +#define ES8326_INT_SOURCE 0x58 +#define ES8326_INTOUT_IO 0x59 +#define ES8326_SDINOUT1_IO 0x5A +#define ES8326_SDINOUT23_IO 0x5B +#define ES8326_JACK_PULSE 0x5C + +#define ES8326_PULLUP_CTL 0xF9 +#define ES8326_HP_DETECT 0xFB +#define ES8326_CHIP_ID1 0xFD +#define ES8326_CHIP_ID2 0xFE +#define ES8326_CHIP_VERSION 0xFF + +/* ES8326_RESET */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PGAGAIN */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DETECT */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif -- 2.36.1

1 0

Re: [PATCH v2 3/6] ASoC: Intel: Drop hdac_ext usage for codec device creation
by Mark Brown 16 Aug '22

16 Aug '22

On Mon, Aug 15, 2022 at 07:42:24PM +0200, Cezary Rojewski wrote: > To make snd_hda_codec_device_init() the only constructor for struct > hda_codec instances remaining tasks are: Acked-by: Mark Brown <broonie(a)kernel.org>

1 0

[PATCH 5.19 0894/1157] ASoC: max98390: use linux/gpio/consumer.h to fix build
by Greg Kroah-Hartman 15 Aug '22

15 Aug '22

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit aa7407f807b250eca7697e5fe9a699bc6c2fab71 ] Change the header file to fix build errors in max98390.c: ../sound/soc/codecs/max98390.c: In function 'max98390_i2c_probe': ../sound/soc/codecs/max98390.c:1076:22: error: implicit declaration of function 'devm_gpiod_get_optional'; did you mean 'devm_regulator_get_optional'? [-Werror=implicit-function-declaration] 1076 | reset_gpio = devm_gpiod_get_optional(&i2c->dev, ../sound/soc/codecs/max98390.c:1077:55: error: 'GPIOD_OUT_HIGH' undeclared (first use in this function); did you mean 'GPIOF_INIT_HIGH'? 1077 | "reset", GPIOD_OUT_HIGH); ../sound/soc/codecs/max98390.c:1077:55: note: each undeclared identifier is reported only once for each function it appears in ../sound/soc/codecs/max98390.c:1083:17: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1083 | gpiod_set_value_cansleep(reset_gpio, 0); Fixes: 397ff0249606 ("ASoC: max98390: Add reset gpio control") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Steve Lee <steve.lee.analog(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Link: https://lore.kernel.org/r/20220605163123.23537-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 2a6b1648c884..d83f81d9ff4e 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -10,7 +10,7 @@ #include <linux/cdev.h> #include <linux/dmi.h> #include <linux/firmware.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_gpio.h> -- 2.35.1

1 0

[PATCH] ASoC: cs42l42: Only report button state if there was a button interrupt
by Richard Fitzgerald 15 Aug '22

15 Aug '22

Only report a button state change if the interrupt status shows that there was a button event. Previously the code would always drop into the button reporting at the end of interrupt handling if the jack was present. If neither of the button report interrupts were pending it would report all buttons released. This could then lead to a button being reported as released while it is still pressed. Fixes: c5b8ee0879bc ("ASoC: cs42l42: Report jack and button detection") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d545a593a251..daafd4251ce6 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1617,7 +1617,6 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) unsigned int current_plug_status; unsigned int current_button_status; unsigned int i; - int report = 0; mutex_lock(&cs42l42->irq_lock); if (cs42l42->suspended) { @@ -1711,13 +1710,15 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) if (current_button_status & CS42L42_M_DETECT_TF_MASK) { dev_dbg(cs42l42->dev, "Button released\n"); - report = 0; + snd_soc_jack_report(cs42l42->jack, 0, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); } else if (current_button_status & CS42L42_M_DETECT_FT_MASK) { - report = cs42l42_handle_button_press(cs42l42); - + snd_soc_jack_report(cs42l42->jack, + cs42l42_handle_button_press(cs42l42), + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); } - snd_soc_jack_report(cs42l42->jack, report, SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3); } } -- 2.30.2

2 1

trying to open ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf for PinePhonePro
by GitHub issues - edited 15 Aug '22

15 Aug '22

alsa-project/alsa-ucm-conf issue #199 was edited from ollieparanoid: Hi, we are trying to narrow down why alsa-lib tries to open `/usr/share/alsa/ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf` instead of the proper path `/usr/share/alsa/ucm2/PinePhonePro/PINE64-PinePhonePro-.conf` with custom (to-be-upstreamed to alsa-ucm-conf.git) ucm configurations we are using in postmarketOS for the PinePhonePro: https://gitlab.com/postmarketOS/pmaports/-/tree/master/device/community/dev… It only happens for some users, but not for everybody. I've taken a quick look at the code but didn't see right away where the wrong path is built, `git grep UnknownProduct` gives no results. Could somebody give us a hint where to look? I don't really have a clue how this works. Thanks! Downstream discussion: https://gitlab.com/postmarketOS/pmaports/-/merge_requests/3356 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/199 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

trying to open ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf for PinePhonePro
by GitHub issues - opened 15 Aug '22

15 Aug '22

alsa-project/alsa-ucm-conf issue #199 was opened from ollieparanoid: Hi, we are trying to narrow down why alsa-lib tries to open `/usr/share/alsa/ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf` instead of the proper path `/usr/share/alsa/ucm2/PinePhonePro/PINE64-PinePhonePro-.conf` with custom (to-be-upstreamed to alsa-ucm-conf.git) ucm configurations we are using in postmarketOS for the PinePhonePro: https://gitlab.com/postmarketOS/pmaports/-/tree/master/device/community/dev… It only happens for some users, but not for everybody. I've taken a quick look at the code but didn't see right away where the wrong path is built, `git grep UnknownProduct` gives no results. Could somebody give us a hint where to look? I don't really have a clue how this works. Thanks! Downstream discussion: https://gitlab.com/postmarketOS/pmaports/-/merge_requests/3356 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/199 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

trying to open ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf for PinePhonePro
by GitHub issues - opened 15 Aug '22

15 Aug '22

alsa-project/alsa-lib issue #260 was opened from ollieparanoid: Hi, we are trying to narrow down why alsa-lib tries to open `/usr/share/alsa/ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf` instead of the proper path `/usr/share/alsa/ucm2/PinePhonePro/PINE64-PinePhonePro-.conf` with custom (to-be-upstreamed to alsa-ucm-conf.git) ucm configurations we are using in postmarketOS for the PinePhonePro: https://gitlab.com/postmarketOS/pmaports/-/tree/master/device/community/dev… It only happens for some users, but not for everybody. I've taken a quick look at the code but didn't see right away where the wrong path is built, `git grep UnknownProduct` gives no results. Could somebody give us a hint where to look? I don't really have a clue how this works. Thanks! Downstream discussion: https://gitlab.com/postmarketOS/pmaports/-/merge_requests/3356 Issue URL : https://github.com/alsa-project/alsa-lib/issues/260 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v4] ASoC: codecs: dt bind. doc for the new TI SRC4392 codec
by Matt Flax 15 Aug '22

15 Aug '22

Signed-off-by: Matt Flax <flatmax(a)flatmax.com> --- Notes: v4: * updated the id to have the correct path v3: * Renamed to include ti, * fixed license to include BSD-2-Clause * removed generic description * renamed the node in the example v2: * Changed the maintainer .../devicetree/bindings/sound/ti,src4xxx.yaml | 48 +++++++++++++++++++ 1 file changed, 48 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,src4xxx.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml new file mode 100644 index 000000000000..9681b72b4918 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,src4xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments SRC4392 Device Tree Bindings + +description: | + The SRC4392 is a digital audio codec that can be connected via + I2C or SPI. Currently, only I2C bus is supported. + +maintainers: + - Matt Flax <flatmax(a)flatmax.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: ti,src4392 + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@70 { + #sound-dai-cells = <0>; + compatible = "ti,src4392"; + reg = <0x70>; + }; + }; +... -- 2.34.1

2 1

[PATCH] ASoC: mt6359: fix tests for platform_get_irq() failure
by Dan Carpenter 15 Aug '22

15 Aug '22

The platform_get_irq() returns negative error codes. It can't actually return zero, but if it did that should be treated as success. Fixes: eef07b9e0925 ("ASoC: mediatek: mt6359: add MT6359 accdet jack driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/mt6359-accdet.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c index c190628e2905..7f624854948c 100644 --- a/sound/soc/codecs/mt6359-accdet.c +++ b/sound/soc/codecs/mt6359-accdet.c @@ -965,7 +965,7 @@ static int mt6359_accdet_probe(struct platform_device *pdev) mutex_init(&priv->res_lock); priv->accdet_irq = platform_get_irq(pdev, 0); - if (priv->accdet_irq) { + if (priv->accdet_irq >= 0) { ret = devm_request_threaded_irq(&pdev->dev, priv->accdet_irq, NULL, mt6359_accdet_irq, IRQF_TRIGGER_HIGH | IRQF_ONESHOT, @@ -979,7 +979,7 @@ static int mt6359_accdet_probe(struct platform_device *pdev) if (priv->caps & ACCDET_PMIC_EINT0) { priv->accdet_eint0 = platform_get_irq(pdev, 1); - if (priv->accdet_eint0) { + if (priv->accdet_eint0 >= 0) { ret = devm_request_threaded_irq(&pdev->dev, priv->accdet_eint0, NULL, mt6359_accdet_irq, @@ -994,7 +994,7 @@ static int mt6359_accdet_probe(struct platform_device *pdev) } } else if (priv->caps & ACCDET_PMIC_EINT1) { priv->accdet_eint1 = platform_get_irq(pdev, 2); - if (priv->accdet_eint1) { + if (priv->accdet_eint1 >= 0) { ret = devm_request_threaded_irq(&pdev->dev, priv->accdet_eint1, NULL, mt6359_accdet_irq, -- 2.35.1

2 1

[PATCH v3] ASoC: codecs: add support for the TI SRC4392 codec
by Matt Flax 15 Aug '22

15 Aug '22

The src4xxx keyword is used for future capability to integrate other codecs similar to the src4392 to the same code base. Signed-off-by: Matt Flax <flatmax(a)flatmax.com> --- Notes: V3 changes : * converted header to C++ comments * added break to default switch cases * updated SRC vol to SRC volume, scanned for other name issues * now using yaml for the dt binding documentation (separate patch) * Fixed Kconfig to select the _I2C module in menuconfig V2 changes : * Fixed indents in Kconfig * Comment header of source code is now one block * Documented the devicetree binding * sample rate converter MCLK src is non-user settable * hushed spurious dev_info output * src4xxx_remove was empty is removed. * Using switch in hw_params to select port reg. for clarity * Removed interrupt reg. settings - will re-intro. properly later * probe function is now EXPORT_SYMBOL_GPL sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/src4xxx-i2c.c | 47 +++ sound/soc/codecs/src4xxx.c | 513 +++++++++++++++++++++++++++++++++ sound/soc/codecs/src4xxx.h | 113 ++++++++ 5 files changed, 690 insertions(+) create mode 100644 sound/soc/codecs/src4xxx-i2c.c create mode 100644 sound/soc/codecs/src4xxx.c create mode 100644 sound/soc/codecs/src4xxx.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d16b4efb88a7..fe445465f877 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -205,6 +205,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_SIMPLE_AMPLIFIER imply SND_SOC_SIMPLE_MUX imply SND_SOC_SPDIF + imply SND_SOC_SRC4XXX_I2C imply SND_SOC_SSM2305 imply SND_SOC_SSM2518 imply SND_SOC_SSM2602_SPI @@ -1471,6 +1472,18 @@ config SND_SOC_SIMPLE_MUX config SND_SOC_SPDIF tristate "S/PDIF CODEC" +config SND_SOC_SRC4XXX_I2C + tristate "Texas Instruments SRC4XXX DIR/DIT and SRC codecs" + depends on I2C + select SND_SOC_SRC4XXX + help + Enable support for the TI SRC4XXX family of codecs. These include the + scr4392 which has digital receivers, transmitters, and + a sample rate converter, including numerous ports. + +config SND_SOC_SRC4XXX + tristate + config SND_SOC_SSM2305 tristate "Analog Devices SSM2305 Class-D Amplifier" help diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 92fd441d426a..a599b727c65b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -231,6 +231,8 @@ snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o snd-soc-si476x-objs := si476x.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-src4xxx-objs := src4xxx.o +snd-soc-src4xxx-i2c-objs := src4xxx-i2c.o snd-soc-ssm2305-objs := ssm2305.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o @@ -579,6 +581,8 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SRC4XXX) += snd-soc-src4xxx.o +obj-$(CONFIG_SND_SOC_SRC4XXX_I2C) += snd-soc-src4xxx-i2c.o obj-$(CONFIG_SND_SOC_SSM2305) += snd-soc-ssm2305.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/src4xxx-i2c.c b/sound/soc/codecs/src4xxx-i2c.c new file mode 100644 index 000000000000..43daa9dc8ab5 --- /dev/null +++ b/sound/soc/codecs/src4xxx-i2c.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Driver for SRC4XXX codecs +// +// Copyright 2021-2022 Deqx Pty Ltd +// Author: Matt Flax <flatmax(a)flatmax.com> + +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "src4xxx.h" + +static int src4xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + return src4xxx_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &src4xxx_regmap_config), NULL); +} + +static const struct i2c_device_id src4xxx_i2c_ids[] = { + { "src4392", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, src4xxx_i2c_ids); + +static const struct of_device_id src4xxx_of_match[] = { + { .compatible = "ti,src4392", }, + { } +}; +MODULE_DEVICE_TABLE(of, src4xxx_of_match); + + +static struct i2c_driver src4xxx_i2c_driver = { + .driver = { + .name = "src4xxx", + .of_match_table = of_match_ptr(src4xxx_of_match), + }, + .probe = src4xxx_i2c_probe, + .id_table = src4xxx_i2c_ids, +}; +module_i2c_driver(src4xxx_i2c_driver); + +MODULE_DESCRIPTION("ASoC SRC4392 CODEC I2C driver"); +MODULE_AUTHOR("Matt Flax <flatmax(a)flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.c b/sound/soc/codecs/src4xxx.c new file mode 100644 index 000000000000..a8f143057b41 --- /dev/null +++ b/sound/soc/codecs/src4xxx.c @@ -0,0 +1,513 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TI SRC4xxx Audio Codec driver +// +// Copyright 2021-2022 Deqx Pty Ltd +// Author: Matt Flax <flatmax(a)flatmax.com> + +#include <linux/module.h> + +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "src4xxx.h" + +struct src4xxx { + struct regmap *regmap; + bool master[2]; + int mclk_hz; + struct device *dev; +}; + +enum {SRC4XXX_PORTA, SRC4XXX_PORTB}; + +/* SRC attenuation */ +static const DECLARE_TLV_DB_SCALE(src_tlv, -12750, 50, 0); + +static const struct snd_kcontrol_new src4xxx_controls[] = { + SOC_DOUBLE_R_TLV("SRC Volume", + SRC4XXX_SCR_CTL_30, SRC4XXX_SCR_CTL_31, 0, 255, 1, src_tlv), +}; + +/* I2S port control */ +static const char * const port_out_src_text[] = { + "loopback", "other_port", "DIR", "SRC" +}; +static SOC_ENUM_SINGLE_DECL(porta_out_src_enum, SRC4XXX_PORTA_CTL_03, 4, + port_out_src_text); +static SOC_ENUM_SINGLE_DECL(portb_out_src_enum, SRC4XXX_PORTB_CTL_05, 4, + port_out_src_text); +static const struct snd_kcontrol_new porta_out_control = + SOC_DAPM_ENUM("Port A source select", porta_out_src_enum); +static const struct snd_kcontrol_new portb_out_control = + SOC_DAPM_ENUM("Port B source select", portb_out_src_enum); + +/* Digital audio transmitter control */ +static const char * const dit_mux_text[] = {"Port A", "Port B", "DIR", "SRC"}; +static SOC_ENUM_SINGLE_DECL(dit_mux_enum, SRC4XXX_TX_CTL_07, 3, dit_mux_text); +static const struct snd_kcontrol_new dit_mux_control = + SOC_DAPM_ENUM("DIT source", dit_mux_enum); + +/* SRC control */ +static const char * const src_in_text[] = {"Port A", "Port B", "DIR"}; +static SOC_ENUM_SINGLE_DECL(src_in_enum, SRC4XXX_SCR_CTL_2D, 0, src_in_text); +static const struct snd_kcontrol_new src_in_control = + SOC_DAPM_ENUM("SRC source select", src_in_enum); + +/* DIR control */ +static const char * const dir_in_text[] = {"Ch 1", "Ch 2", "Ch 3", "Ch 4"}; +static SOC_ENUM_SINGLE_DECL(dir_in_enum, SRC4XXX_RCV_CTL_0D, 0, dir_in_text); +static const struct snd_kcontrol_new dir_in_control = + SOC_DAPM_ENUM("Digital Input", dir_in_enum); + +static const struct snd_soc_dapm_widget src4xxx_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("loopback_A"), + SND_SOC_DAPM_INPUT("other_port_A"), + SND_SOC_DAPM_INPUT("DIR_A"), + SND_SOC_DAPM_INPUT("SRC_A"), + SND_SOC_DAPM_MUX("Port A source", + SND_SOC_NOPM, 0, 0, &porta_out_control), + + SND_SOC_DAPM_INPUT("loopback_B"), + SND_SOC_DAPM_INPUT("other_port_B"), + SND_SOC_DAPM_INPUT("DIR_B"), + SND_SOC_DAPM_INPUT("SRC_B"), + SND_SOC_DAPM_MUX("Port B source", + SND_SOC_NOPM, 0, 0, &portb_out_control), + + SND_SOC_DAPM_INPUT("Port_A"), + SND_SOC_DAPM_INPUT("Port_B"), + SND_SOC_DAPM_INPUT("DIR_"), + + /* Digital audio receivers and transmitters */ + SND_SOC_DAPM_OUTPUT("DIR_OUT"), + SND_SOC_DAPM_OUTPUT("SRC_OUT"), + SND_SOC_DAPM_MUX("DIT Out Src", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIT_SHIFT, 1, &dit_mux_control), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF_A_RX", "Playback A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_A_TX", "Capture A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_IN("AIF_B_RX", "Playback B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_B_TX", "Capture B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + + SND_SOC_DAPM_MUX("SRC source", SND_SOC_NOPM, 0, 0, &src_in_control), + + SND_SOC_DAPM_INPUT("MCLK"), + SND_SOC_DAPM_INPUT("RXMCLKI"), + SND_SOC_DAPM_INPUT("RXMCLKO"), + + SND_SOC_DAPM_INPUT("RX1"), + SND_SOC_DAPM_INPUT("RX2"), + SND_SOC_DAPM_INPUT("RX3"), + SND_SOC_DAPM_INPUT("RX4"), + SND_SOC_DAPM_MUX("Digital Input", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIR_SHIFT, 1, &dir_in_control), +}; + +static const struct snd_soc_dapm_route src4xxx_audio_routes[] = { + /* I2S Input to Output Routing */ + {"Port A source", "loopback", "loopback_A"}, + {"Port A source", "other_port", "other_port_A"}, + {"Port A source", "DIR", "DIR_A"}, + {"Port A source", "SRC", "SRC_A"}, + {"Port B source", "loopback", "loopback_B"}, + {"Port B source", "other_port", "other_port_B"}, + {"Port B source", "DIR", "DIR_B"}, + {"Port B source", "SRC", "SRC_B"}, + /* DIT muxing */ + {"DIT Out Src", "Port A", "Capture A"}, + {"DIT Out Src", "Port B", "Capture B"}, + {"DIT Out Src", "DIR", "DIR_OUT"}, + {"DIT Out Src", "SRC", "SRC_OUT"}, + + /* SRC input selection */ + {"SRC source", "Port A", "Port_A"}, + {"SRC source", "Port B", "Port_B"}, + {"SRC source", "DIR", "DIR_"}, + /* SRC mclk selection */ + {"SRC mclk source", "Master (MCLK)", "MCLK"}, + {"SRC mclk source", "Master (RXCLKI)", "RXMCLKI"}, + {"SRC mclk source", "Recovered receiver clk", "RXMCLKO"}, + /* DIR input selection */ + {"Digital Input", "Ch 1", "RX1"}, + {"Digital Input", "Ch 2", "RX2"}, + {"Digital Input", "Ch 3", "RX3"}, + {"Digital Input", "Ch 4", "RX4"}, +}; + + +static const struct snd_soc_component_driver src4xxx_driver = { + .controls = src4xxx_controls, + .num_controls = ARRAY_SIZE(src4xxx_controls), + + .dapm_widgets = src4xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(src4xxx_dapm_widgets), + .dapm_routes = src4xxx_audio_routes, + .num_dapm_routes = ARRAY_SIZE(src4xxx_audio_routes), +}; + +static int src4xxx_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int ctrl; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl = SRC4XXX_BUS_MASTER; + src4xxx->master[dai->id] = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl = 0; + src4xxx->master[dai->id] = false; + break; + default: + return -EINVAL; + break; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl |= SRC4XXX_BUS_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl |= SRC4XXX_BUS_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl |= SRC4XXX_BUS_RIGHT_J_24; + break; + default: + return -EINVAL; + break; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + break; + } + + regmap_update_bits(src4xxx->regmap, SRC4XXX_BUS_FMT(dai->id), + SRC4XXX_BUS_FMT_MS_MASK, ctrl); + + return 0; +} + +static int src4xxx_set_mclk_hz(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + + dev_info(component->dev, "changing mclk rate from %d to %d Hz\n", + src4xxx->mclk_hz, freq); + src4xxx->mclk_hz = freq; + + return 0; +} + +static int src4xxx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int mclk_div; + int val, pj, jd, d; + int reg; + int ret; + + switch (dai->id) { + case SRC4XXX_PORTB: + reg = SRC4XXX_PORTB_CTL_06; + break; + default: + reg = SRC4XXX_PORTA_CTL_04; + break; + } + + if (src4xxx->master[dai->id]) { + mclk_div = src4xxx->mclk_hz/params_rate(params); + if (src4xxx->mclk_hz != mclk_div*params_rate(params)) { + dev_err(component->dev, + "mclk %d / rate %d has a remainder.\n", + src4xxx->mclk_hz, params_rate(params)); + return -EINVAL; + } + + val = ((int)mclk_div - 128) / 128; + if ((val < 0) | (val > 3)) { + dev_err(component->dev, + "div register setting %d is out of range\n", + val); + dev_err(component->dev, + "unsupported sample rate %d Hz for the master clock of %d Hz\n", + params_rate(params), src4xxx->mclk_hz); + return -EINVAL; + } + + /* set the TX DIV */ + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + /* set the PLL for the digital receiver */ + switch (src4xxx->mclk_hz) { + case 24576000: + pj = 0x22; + jd = 0x00; + d = 0x00; + break; + case 22579200: + pj = 0x22; + jd = 0x1b; + d = 0xa3; + break; + default: + /* don't error out here, + * other parts of the chip are still functional + */ + dev_info(component->dev, + "Couldn't set the RCV PLL as this master clock rate is unknown\n"); + break; + } + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_0F); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_10); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_11); + + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret < 0) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + return regmap_update_bits(src4xxx->regmap, reg, + SRC4XXX_MCLK_DIV_MASK, val); + } else { + dev_info(dai->dev, "not setting up MCLK as not master\n"); + } + + return 0; +}; + +static const struct snd_soc_dai_ops src4xxx_dai_ops = { + .hw_params = src4xxx_hw_params, + .set_sysclk = src4xxx_set_mclk_hz, + .set_fmt = src4xxx_set_dai_fmt, +}; + +#define SRC4XXX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) +#define SRC4XXX_RATES (SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000|\ + SNDRV_PCM_RATE_88200|\ + SNDRV_PCM_RATE_96000|\ + SNDRV_PCM_RATE_176400|\ + SNDRV_PCM_RATE_192000) + +static struct snd_soc_dai_driver src4xxx_dai_driver[] = { + { + .id = SRC4XXX_PORTA, + .name = "src4xxx-portA", + .playback = { + .stream_name = "Playback A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, + { + .id = SRC4XXX_PORTB, + .name = "src4xxx-portB", + .playback = { + .stream_name = "Playback B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, +}; + +static const struct reg_default src4xxx_reg_defaults[] = { + { SRC4XXX_PWR_RST_01, 0x00 }, /* all powered down intially */ + { SRC4XXX_PORTA_CTL_03, 0x00 }, + { SRC4XXX_PORTA_CTL_04, 0x00 }, + { SRC4XXX_PORTB_CTL_05, 0x00 }, + { SRC4XXX_PORTB_CTL_06, 0x00 }, + { SRC4XXX_TX_CTL_07, 0x00 }, + { SRC4XXX_TX_CTL_08, 0x00 }, + { SRC4XXX_TX_CTL_09, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MSK_0B, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MODE_0C, 0x00 }, + { SRC4XXX_RCV_CTL_0D, 0x00 }, + { SRC4XXX_RCV_CTL_0E, 0x00 }, + { SRC4XXX_RCV_PLL_0F, 0x00 }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_10, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_11, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RVC_IRQ_MSK_16, 0x00 }, + { SRC4XXX_RVC_IRQ_MSK_17, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_18, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_19, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_1A, 0x00 }, + { SRC4XXX_GPIO_1_1B, 0x00 }, + { SRC4XXX_GPIO_2_1C, 0x00 }, + { SRC4XXX_GPIO_3_1D, 0x00 }, + { SRC4XXX_GPIO_4_1E, 0x00 }, + { SRC4XXX_SCR_CTL_2D, 0x00 }, + { SRC4XXX_SCR_CTL_2E, 0x00 }, + { SRC4XXX_SCR_CTL_2F, 0x00 }, + { SRC4XXX_SCR_CTL_30, 0x00 }, + { SRC4XXX_SCR_CTL_31, 0x00 }, +}; + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)) +{ + struct src4xxx *src4xxx; + int ret; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + src4xxx = devm_kzalloc(dev, sizeof(*src4xxx), GFP_KERNEL); + if (!src4xxx) + return -ENOMEM; + + src4xxx->regmap = regmap; + src4xxx->dev = dev; + src4xxx->mclk_hz = 0; /* mclk has not been configured yet */ + dev_set_drvdata(dev, src4xxx); + + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_RESET); + if (ret < 0) + dev_err(dev, "Failed to issue reset: %d\n", ret); + usleep_range(1, 500); /* sleep for more then 500 ns */ + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_POWER_DOWN); + if (ret < 0) + dev_err(dev, "Failed to decommission reset: %d\n", ret); + usleep_range(500, 1000); /* sleep for 500 us or more */ + + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_PWR_RST_01, + SRC4XXX_POWER_ENABLE, SRC4XXX_POWER_ENABLE); + if (ret < 0) + dev_err(dev, "Failed to port A and B : %d\n", ret); + + /* set receiver to use master clock (rcv mclk is most likely jittery) */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0D, + SRC4XXX_RXCLK_MCLK, SRC4XXX_RXCLK_MCLK); + if (ret < 0) + dev_err(dev, + "Failed to enable mclk as the PLL1 DIR reference : %d\n", ret); + + /* default to leaving the PLL2 running on loss of lock, divide by 8 */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0E, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL); + if (ret < 0) + dev_err(dev, "Failed to enable mclk rec and div : %d\n", ret); + + ret = devm_snd_soc_register_component(dev, &src4xxx_driver, + src4xxx_dai_driver, ARRAY_SIZE(src4xxx_dai_driver)); + if (ret == 0) + dev_info(dev, "src4392 probe ok %d\n", ret); + return ret; +} +EXPORT_SYMBOL_GPL(src4xxx_probe); + +static bool src4xxx_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SRC4XXX_RES_00: + case SRC4XXX_GLOBAL_ITR_STS_02: + case SRC4XXX_SRC_DIT_STS_0A: + case SRC4XXX_NON_AUDIO_D_12: + case SRC4XXX_RVC_STS_13: + case SRC4XXX_RVC_STS_14: + case SRC4XXX_RVC_STS_15: + case SRC4XXX_SUB_CODE_1F: + case SRC4XXX_SUB_CODE_20: + case SRC4XXX_SUB_CODE_21: + case SRC4XXX_SUB_CODE_22: + case SRC4XXX_SUB_CODE_23: + case SRC4XXX_SUB_CODE_24: + case SRC4XXX_SUB_CODE_25: + case SRC4XXX_SUB_CODE_26: + case SRC4XXX_SUB_CODE_27: + case SRC4XXX_SUB_CODE_28: + case SRC4XXX_PC_PREAMBLE_HI_29: + case SRC4XXX_PC_PREAMBLE_LO_2A: + case SRC4XXX_PD_PREAMBLE_HI_2B: + case SRC4XXX_PC_PREAMBLE_LO_2C: + case SRC4XXX_IO_RATIO_32: + case SRC4XXX_IO_RATIO_33: + return true; + } + + if (reg > SRC4XXX_IO_RATIO_33 && reg < SRC4XXX_PAGE_SEL_7F) + return true; + + return false; +} + +const struct regmap_config src4xxx_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + .max_register = SRC4XXX_IO_RATIO_33, + + .reg_defaults = src4xxx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(src4xxx_reg_defaults), + .volatile_reg = src4xxx_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(src4xxx_regmap_config); + +MODULE_DESCRIPTION("ASoC SRC4XXX CODEC driver"); +MODULE_AUTHOR("Matt Flax <flatmax(a)flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.h b/sound/soc/codecs/src4xxx.h new file mode 100644 index 000000000000..5bf778fb9945 --- /dev/null +++ b/sound/soc/codecs/src4xxx.h @@ -0,0 +1,113 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// src4xxx.h -- SRC4XXX ALSA SoC audio driver +// +// Copyright 2021-2022 Deqx Pty Ltd +// Author: Matt R Flax <flatmax(a)flatmax.com> + +#ifndef __SRC4XXX_H__ +#define __SRC4XXX_H__ + +#define SRC4XXX_RES_00 0x00 +#define SRC4XXX_PWR_RST_01 0x01 +#define SRC4XXX_RESET 0x80 +#define SRC4XXX_POWER_DOWN 0x00 +#define SRC4XXX_POWER_ENABLE 0x20 +#define SRC4XXX_ENABLE_SRC 0x1 +#define SRC4XXX_ENABLE_SRC_SHIFT 0 +#define SRC4XXX_ENABLE_DIR 0x2 +#define SRC4XXX_ENABLE_DIR_SHIFT 1 +#define SRC4XXX_ENABLE_DIT 0x4 +#define SRC4XXX_ENABLE_DIT_SHIFT 2 +#define SRC4XXX_ENABLE_PORT_B 0x8 +#define SRC4XXX_ENABLE_PORT_B_SHIFT 3 +#define SRC4XXX_ENABLE_PORT_A 0x10 +#define SRC4XXX_ENABLE_PORT_A_SHIFT 4 + +#define SRC4XXX_PORTA_CTL_03 0x03 +#define SRC4XXX_BUS_MASTER 0x8 +#define SRC4XXX_BUS_LEFT_J 0x0 +#define SRC4XXX_BUS_I2S 0x1 +#define SRC4XXX_BUS_RIGHT_J_16 0x4 +#define SRC4XXX_BUS_RIGHT_J_18 0x5 +#define SRC4XXX_BUS_RIGHT_J_20 0x6 +#define SRC4XXX_BUS_RIGHT_J_24 0x7 +#define SRC4XXX_BUS_FMT_MS_MASK 0xf + +#define SRC4XXX_PORTA_CTL_04 0x04 +#define SRC4XXX_MCLK_DIV_MASK 0x3 + +#define SRC4XXX_BUS_FMT(id) (SRC4XXX_PORTA_CTL_03+2*id) +#define SRC4XXX_BUS_CLK(id) (SRC4XXX_PORTA_CTL_04+2*id) + +#define SRC4XXX_PORTB_CTL_05 0x05 +#define SRC4XXX_PORTB_CTL_06 0x06 + +#define SRC4XXX_TX_CTL_07 0x07 +#define SRC4XXX_TX_MCLK_DIV_MASK 0x60 +#define SRC4XXX_TX_MCLK_DIV_SHIFT 5 + +#define SRC4XXX_TX_CTL_08 0x08 +#define SRC4XXX_TX_CTL_09 0x09 +#define SRC4XXX_SRC_DIT_IRQ_MSK_0B 0x0B +#define SRC4XXX_SRC_BTI_EN 0x01 +#define SRC4XXX_SRC_TSLIP_EN 0x02 +#define SRC4XXX_SRC_DIT_IRQ_MODE_0C 0x0C +#define SRC4XXX_RCV_CTL_0D 0x0D +#define SRC4XXX_RXCLK_RXCKI 0x0 +#define SRC4XXX_RXCLK_MCLK 0x8 +#define SRC4XXX_RCV_CTL_0E 0x0E +#define SRC4XXX_REC_MCLK_EN 0x1 +#define SRC4XXX_PLL2_DIV_0 (0x0<<1) +#define SRC4XXX_PLL2_DIV_2 (0x1<<1) +#define SRC4XXX_PLL2_DIV_4 (0x2<<1) +#define SRC4XXX_PLL2_DIV_8 (0x3<<1) +#define SRC4XXX_PLL2_LOL 0x8 +#define SRC4XXX_RCV_PLL_0F 0x0F +#define SRC4XXX_RCV_PLL_10 0x10 +#define SRC4XXX_RCV_PLL_11 0x11 +#define SRC4XXX_RVC_IRQ_MSK_16 0x16 +#define SRC4XXX_RVC_IRQ_MSK_17 0x17 +#define SRC4XXX_RVC_IRQ_MODE_18 0x18 +#define SRC4XXX_RVC_IRQ_MODE_19 0x19 +#define SRC4XXX_RVC_IRQ_MODE_1A 0x1A +#define SRC4XXX_GPIO_1_1B 0x1B +#define SRC4XXX_GPIO_2_1C 0x1C +#define SRC4XXX_GPIO_3_1D 0x1D +#define SRC4XXX_GPIO_4_1E 0x1E +#define SRC4XXX_SCR_CTL_2D 0x2D +#define SRC4XXX_SCR_CTL_2E 0x2E +#define SRC4XXX_SCR_CTL_2F 0x2F +#define SRC4XXX_SCR_CTL_30 0x30 +#define SRC4XXX_SCR_CTL_31 0x31 +#define SRC4XXX_PAGE_SEL_7F 0x7F + +// read only registers +#define SRC4XXX_GLOBAL_ITR_STS_02 0x02 +#define SRC4XXX_SRC_DIT_STS_0A 0x0A +#define SRC4XXX_NON_AUDIO_D_12 0x12 +#define SRC4XXX_RVC_STS_13 0x13 +#define SRC4XXX_RVC_STS_14 0x14 +#define SRC4XXX_RVC_STS_15 0x15 +#define SRC4XXX_SUB_CODE_1F 0x1F +#define SRC4XXX_SUB_CODE_20 0x20 +#define SRC4XXX_SUB_CODE_21 0x21 +#define SRC4XXX_SUB_CODE_22 0x22 +#define SRC4XXX_SUB_CODE_23 0x23 +#define SRC4XXX_SUB_CODE_24 0x24 +#define SRC4XXX_SUB_CODE_25 0x25 +#define SRC4XXX_SUB_CODE_26 0x26 +#define SRC4XXX_SUB_CODE_27 0x27 +#define SRC4XXX_SUB_CODE_28 0x28 +#define SRC4XXX_PC_PREAMBLE_HI_29 0x29 +#define SRC4XXX_PC_PREAMBLE_LO_2A 0x2A +#define SRC4XXX_PD_PREAMBLE_HI_2B 0x2B +#define SRC4XXX_PC_PREAMBLE_LO_2C 0x2C +#define SRC4XXX_IO_RATIO_32 0x32 +#define SRC4XXX_IO_RATIO_33 0x33 + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)); +extern const struct regmap_config src4xxx_regmap_config; + +#endif /* __SRC4XXX_H__ */ -- 2.34.1

2 2

[PATCH v2] ASoC: codecs: dt bind. doc for the new TI SRC4392 codec
by Matt Flax 15 Aug '22

15 Aug '22

Signed-off-by: Matt Flax <flatmax(a)flatmax.com> --- Notes: v2: * Added myself as the maintainer .../devicetree/bindings/sound/src4xxx.yaml | 50 +++++++++++++++++++ 1 file changed, 50 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/src4xxx.yaml diff --git a/Documentation/devicetree/bindings/sound/src4xxx.yaml b/Documentation/devicetree/bindings/sound/src4xxx.yaml new file mode 100644 index 000000000000..368bb4ba1a77 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/src4xxx.yaml @@ -0,0 +1,50 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/src4xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments SRC4392 Device Tree Bindings + +description: | + The SRC4392 is a digital audio codec that can be connected via + I2C or SPI. Currently, only I2C bus is supported. + +maintainers: + - Matt Flax <flatmax(a)flatmax.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: ti,src4392 + + "#sound-dai-cells": + const: 0 + + reg: + description: + The I2C address of the device + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + src4392@70 { + #sound-dai-cells = <0>; + compatible = "ti,src4392"; + reg = <0x70>; + }; + }; +... -- 2.34.1

3 2

[PATCH v3 0/3] DT binding for sample format conversion
by Sameer Pujar 15 Aug '22

15 Aug '22

DT binding properties are available to fixup rate and channel parameters of a DAI. This series extends this to sample format conversion as well. With this now DAI PCM parameters (channels, sample rate and sample format) can be fixed up as necessary in an audio path. Changelog: ========== v2->v3: ------- * Fix DT binding errors in simple-card.yaml * Drop simple-card binding changes from the series since there is no usage requirement. It can be extended later when necessary. * Use definitions instead of properties in the common schema and re-use this in audio-graph related bindings. v1->v2: ------- * Move DAI params properties to a new schema and re-use this for simple-card and audio-graph-card. * Use string type for DAI format binding as suggested by Krzysztof. Sameer Pujar (3): ASoC: dt-bindings: Definitions for DAI params ASoC: dt-bindings: Add sample format conversion ASoC: simple-card-utils: Fixup DAI sample format .../bindings/sound/audio-graph-port.yaml | 17 ++++----- .../devicetree/bindings/sound/audio-graph.yaml | 9 ++--- .../devicetree/bindings/sound/dai-params.yaml | 40 ++++++++++++++++++++++ include/sound/simple_card_utils.h | 1 + sound/soc/generic/simple-card-utils.c | 34 ++++++++++++++++++ 5 files changed, 89 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/dai-params.yaml -- 2.7.4

3 5

[PATCH] ASoC: Variable type completion
by Xin Gao 15 Aug '22

15 Aug '22

'unsigned int' is better than 'unsigned'. Signed-off-by: Xin Gao <gaoxin(a)cdjrlc.com> --- sound/soc/codecs/uda134x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 037833c509f7..607897f1f357 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -450,7 +450,7 @@ static int uda134x_soc_probe(struct snd_soc_component *component) struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); struct uda134x_platform_data *pd = uda134x->pd; const struct snd_soc_dapm_widget *widgets; - unsigned num_widgets; + unsigned int num_widgets; int ret; printk(KERN_INFO "UDA134X SoC Audio Codec\n"); -- 2.30.2

2 1

[PATCH] ASoC: sam9g20_wm8731: Simplify some error message
by Christophe JAILLET 15 Aug '22

15 Aug '22

dev_err_probe() already prints the error code in a human readable way, so there is no need to duplicate it as a numerical value at the end of the message. Fixes: 29f4078f777f ("ASoC: sam9g20_wm8731: Use dev_err_probe() for snd_soc_register_card()") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 4d25fb61c652..1430642c8433 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -172,7 +172,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { dev_err_probe(&pdev->dev, ret, - "snd_soc_register_card() failed: %d\n", ret); + "snd_soc_register_card() failed\n"); goto err; } -- 2.34.1

2 1

[PATCH] ASoC: tlv320adcx140: Fix a typo in a comment
by Christophe JAILLET 15 Aug '22

15 Aug '22

s/TLV320ADCX104/TLV320ADCX140/ Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/tlv320adcx140.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index d7d4e3a88b5c..795b5f7194e6 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0 -// TLV320ADCX104 Sound driver +// TLV320ADCX140 Sound driver // Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com/ #ifndef _TLV320ADCX140_H -- 2.34.1

2 1

[PATCH 0/8] ASoC: Replace sprintf() with sysfs_emit()
by Takashi Iwai 15 Aug '22

15 Aug '22

Hi, this is a patch set for rather simple conversions from the plain sprintf() & co to the new helpers, sysfs_emit() and sysfs_emit_at(). No functional changes are expected. Takashi === Takashi Iwai (8): ASoC: cs43130: Replace scnprintf() with sysfs_emit() ASoC: tlv320aic26: Replace sprintf() with sysfs_emit() ASoC: Intel: sst: Replace sprintf() with sysfs_emit() ASoC: Intel: catpt: Replace sprintf() with sysfs_emit() ASoC: Intel: skylake: Replace sprintf() with sysfs_emit() ASoC: core: Replace sprintf() with sysfs_emit() ASoC: DAPM: Replace sprintf() calls with sysfs_emit_at() ASoC: omap: Replace sprintf() with sysfs_emit() sound/soc/codecs/cs43130.c | 11 +++++------ sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/intel/atom/sst/sst.c | 8 ++++---- sound/soc/intel/catpt/sysfs.c | 6 +++--- sound/soc/intel/skylake/skl-nhlt.c | 2 +- sound/soc/soc-core.c | 2 +- sound/soc/soc-dapm.c | 9 ++++----- sound/soc/ti/omap-mcbsp-st.c | 6 +++--- sound/soc/ti/omap-mcbsp.c | 8 ++++---- 9 files changed, 26 insertions(+), 28 deletions(-) -- 2.35.3

4 12

[PATCH v2] ASoC: imx-rpmsg: Support configure sysclk for codec dai
by Shengjiu Wang 15 Aug '22

15 Aug '22

Some codecs need to configure the sysclk even with slave mode, otherwise it may not work properly with some case. wm8960 is the one that need sysclk be configured, so add late_probe() to call the snd_soc_dai_set_sysclk() of codec Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - use unsigned long for sysclk sound/soc/fsl/imx-rpmsg.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 2e117311e582..4d99f4858a14 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -19,6 +19,7 @@ struct imx_rpmsg { struct snd_soc_dai_link dai; struct snd_soc_card card; + unsigned long sysclk; }; static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { @@ -28,6 +29,27 @@ static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { SND_SOC_DAPM_MIC("Main MIC", NULL), }; +static int imx_rpmsg_late_probe(struct snd_soc_card *card) +{ + struct imx_rpmsg *data = snd_soc_card_get_drvdata(card); + struct snd_soc_pcm_runtime *rtd = list_first_entry(&card->rtd_list, + struct snd_soc_pcm_runtime, list); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct device *dev = card->dev; + int ret; + + if (!data->sysclk) + return 0; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, data->sysclk, SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + return 0; +} + static int imx_rpmsg_probe(struct platform_device *pdev) { struct snd_soc_dai_link_component *dlc; @@ -72,12 +94,18 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->dai.codecs->dai_name = "snd-soc-dummy-dai"; data->dai.codecs->name = "snd-soc-dummy"; } else { + struct clk *clk; + data->dai.codecs->of_node = args.np; ret = snd_soc_get_dai_name(&args, &data->dai.codecs->dai_name); if (ret) { dev_err(&pdev->dev, "Unable to get codec_dai_name\n"); goto fail; } + + clk = devm_get_clk_from_child(&pdev->dev, args.np, NULL); + if (!IS_ERR(clk)) + data->sysclk = clk_get_rate(clk); } data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev); @@ -103,6 +131,7 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->card.owner = THIS_MODULE; data->card.dapm_widgets = imx_rpmsg_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_rpmsg_dapm_widgets); + data->card.late_probe = imx_rpmsg_late_probe; /* * Inoder to use common api to get card name and audio routing. * Use parent of_node for this device, revert it after finishing using -- 2.34.1

2 1

[PATCH] ASoC: mediatek: mt8186: remove unnecessary NULL check
by Dan Carpenter 15 Aug '22

15 Aug '22

The "i2s_priv" pointer cannot be NULL. Some NULL checks were deleted in commit d7bffbe9cbd3 ("ASoC: mediatek: mt8186: remove unnecessary judgments") but this one was accidentally left behind. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c index ec79e2f2a54d..d7a227169548 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c @@ -968,7 +968,7 @@ static int mtk_dai_i2s_config(struct mtk_base_afe *afe, } /* set share i2s */ - if (i2s_priv && i2s_priv->share_i2s_id >= 0) { + if (i2s_priv->share_i2s_id >= 0) { ret = mtk_dai_i2s_config(afe, params, i2s_priv->share_i2s_id); if (ret) return ret; -- 2.35.1

2 1

[PATCH v4] ASoC: dt-bindings: fsl,sai: Convert format to json-schema
by Shengjiu Wang 15 Aug '22

15 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. Beside conversion, 'fsl,shared-interrupt' and '#sound-dai-cells' are added for they are already used by some dts. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v4 - remove $ref for boolean properties changes in v3 - address comments from Krzysztof changes in v2 - fix exclusive property issue - fix order issue of compatible, clock-names, dma-names .../devicetree/bindings/sound/fsl,sai.yaml | 216 ++++++++++++++++++ .../devicetree/bindings/sound/fsl-sai.txt | 95 -------- 2 files changed, 216 insertions(+), 95 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,sai.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..70c4111d59c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,216 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$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: + oneOf: + - enum: + - fsl,vf610-sai + - fsl,imx6sx-sai + - fsl,imx6ul-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - fsl,imx8ulp-sai + - items: + - enum: + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - const: fsl,imx8mq-sai + + reg: + maxItems: 1 + + interrupts: + items: + - description: receive and transmit interrupt + + dmas: + maxItems: 2 + + dma-names: + maxItems: 2 + + clocks: + items: + - description: The ipg clock for register access + - description: master clock source 0 (obsoleted) + - 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 + minItems: 4 + + clock-names: + oneOf: + - items: + - const: bus + - const: mclk0 + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + - items: + - const: bus + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + + lsb-first: + 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. + type: boolean + + big-endian: + description: | + required if all the SAI registers are big-endian rather than little-endian. + type: boolean + + fsl,sai-synchronous-rx: + description: | + 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. + type: boolean + + fsl,sai-asynchronous: + description: | + 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. + type: boolean + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + description: | + Configure the dataline. It has 3 value for each configuration + maxItems: 16 + items: + items: + - description: format Default(0), I2S(1) or PDM(2) + enum: [0, 1, 2] + - description: dataline mask for 'rx' + - description: dataline mask for 'tx' + + fsl,sai-mclk-direction-output: + description: SAI will output the SAI MCLK clock. + type: boolean + + fsl,shared-interrupt: + description: Interrupt is shared with other modules. + type: boolean + + "#sound-dai-cells": + const: 0 + description: optional, some dts node didn't add it. + +allOf: + - if: + properties: + compatible: + contains: + const: fsl,vf610-sai + then: + properties: + dmas: + items: + - description: DMA controller phandle and request line for TX + - description: DMA controller phandle and request line for RX + dma-names: + items: + - const: tx + - const: rx + else: + properties: + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + dma-names: + items: + - const: rx + - const: tx + - if: + required: + - fsl,sai-asynchronous + then: + properties: + fsl,sai-synchronous-rx: false + +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 = "tx", "rx"; + dmas = <&edma0 0 21>, + <&edma0 0 20>; + big-endian; + lsb-first; + }; + + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mm-clock.h> + sai1: sai@30010000 { + compatible = "fsl,imx8mm-sai", "fsl,imx8mq-sai"; + reg = <0x30010000 0x10000>; + interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_SAI1_IPG>, + <&clk IMX8MM_CLK_DUMMY>, + <&clk IMX8MM_CLK_SAI1_ROOT>, + <&clk IMX8MM_CLK_DUMMY>, <&clk IMX8MM_CLK_DUMMY>; + clock-names = "bus", "mclk0", "mclk1", "mclk2", "mclk3"; + dmas = <&sdma2 0 2 0>, <&sdma2 1 2 0>; + dma-names = "rx", "tx"; + fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; + #sound-dai-cells = <0>; + }; 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

3 2

[PATCH v3] ASoC: Intel: cirrus-common: Use UID to map correct amp to prefix
by Stefan Binding 15 Aug '22

15 Aug '22

Since the order of the amps in the ACPI determines the device name, and the ACPI order may change depending on hardware configuration, use UID to dynamically compute the dai links, allowing dynamic assignment of the name_prefix. The UIDs for these amps in ACPI are fixed, and map to a name_prefix, where: UID 0x0 -> WL UID 0x1 -> WR UID 0x2 -> TL UID 0x3 -> TR Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 92 ++++++++++++---------- 1 file changed, 49 insertions(+), 43 deletions(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index f4192df962d6..6e39eda77385 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -10,6 +10,9 @@ #include "../../codecs/cs35l41.h" #include "sof_cirrus_common.h" +#define CS35L41_HID "CSC3541" +#define CS35L41_MAX_AMPS 4 + /* * Cirrus Logic CS35L41/CS35L53 */ @@ -35,50 +38,12 @@ static const struct snd_soc_dapm_route cs35l41_dapm_routes[] = { {"TR Spk", NULL, "TR SPK"}, }; -static struct snd_soc_dai_link_component cs35l41_components[] = { - { - .name = CS35L41_DEV0_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV1_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV2_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV3_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, -}; +static struct snd_soc_dai_link_component cs35l41_components[CS35L41_MAX_AMPS]; /* * Mapping between ACPI instance id and speaker position. - * - * Four speakers: - * 0: Tweeter left, 1: Woofer left - * 2: Tweeter right, 3: Woofer right */ -static struct snd_soc_codec_conf cs35l41_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TR", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "WR", - }, -}; +static struct snd_soc_codec_conf cs35l41_codec_conf[CS35L41_MAX_AMPS]; static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) { @@ -117,10 +82,10 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) static const struct { unsigned int rx[2]; } cs35l41_channel_map[] = { - {.rx = {0, 1}}, /* TL */ {.rx = {0, 1}}, /* WL */ - {.rx = {1, 0}}, /* TR */ {.rx = {1, 0}}, /* WR */ + {.rx = {0, 1}}, /* TL */ + {.rx = {1, 0}}, /* TR */ }; static int cs35l41_hw_params(struct snd_pcm_substream *substream, @@ -175,10 +140,51 @@ static const struct snd_soc_ops cs35l41_ops = { .hw_params = cs35l41_hw_params, }; +static const char * const cs35l41_name_prefixes[] = { "WL", "WR", "TL", "TR" }; + +/* + * Expected UIDs are integers (stored as strings). + * UID Mapping is fixed: + * UID 0x0 -> WL + * UID 0x1 -> WR + * UID 0x2 -> TL + * UID 0x3 -> TR + * Note: If there are less than 4 Amps, UIDs still map to WL/WR/TL/TR. Dynamic code will only create + * dai links for UIDs which exist, and ignore non-existant ones. Only 2 or 4 amps are expected. + * Return number of codecs found. + */ +static int cs35l41_compute_codec_conf(void) +{ + const char * const uid_strings[] = { "0", "1", "2", "3" }; + unsigned int uid, sz = 0; + struct acpi_device *adev; + struct device *physdev; + + for (uid = 0; uid < CS35L41_MAX_AMPS; uid++) { + adev = acpi_dev_get_first_match_dev(CS35L41_HID, uid_strings[uid], -1); + if (!adev) { + pr_devel("Cannot find match for HID %s UID %u (%s)\n", CS35L41_HID, uid, + cs35l41_name_prefixes[uid]); + continue; + } + physdev = get_device(acpi_get_first_physical_node(adev)); + cs35l41_components[sz].name = dev_name(physdev); + cs35l41_components[sz].dai_name = CS35L41_CODEC_DAI; + cs35l41_codec_conf[sz].dlc.name = dev_name(physdev); + cs35l41_codec_conf[sz].name_prefix = cs35l41_name_prefixes[uid]; + acpi_dev_put(adev); + sz++; + } + + if (sz != 2 && sz != 4) + pr_warn("Invalid number of cs35l41 amps found: %d, expected 2 or 4\n", sz); + return sz; +} + void cs35l41_set_dai_link(struct snd_soc_dai_link *link) { + link->num_codecs = cs35l41_compute_codec_conf(); link->codecs = cs35l41_components; - link->num_codecs = ARRAY_SIZE(cs35l41_components); link->init = cs35l41_init; link->ops = &cs35l41_ops; } -- 2.34.1

3 2

[PATCH] ASoC: imx-rpmsg: Support configure sysclk for codec dai
by Shengjiu Wang 15 Aug '22

15 Aug '22

Some codecs need to configure the sysclk even with slave mode, otherwise it may not work properly with some case. wm8960 is the one that need sysclk be configured, so add late_probe() to call the snd_soc_dai_set_sysclk() of codec Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-rpmsg.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 2e117311e582..bef4bd997c98 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -19,6 +19,7 @@ struct imx_rpmsg { struct snd_soc_dai_link dai; struct snd_soc_card card; + unsigned int sysclk; }; static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { @@ -28,6 +29,27 @@ static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { SND_SOC_DAPM_MIC("Main MIC", NULL), }; +static int imx_rpmsg_late_probe(struct snd_soc_card *card) +{ + struct imx_rpmsg *data = snd_soc_card_get_drvdata(card); + struct snd_soc_pcm_runtime *rtd = list_first_entry(&card->rtd_list, + struct snd_soc_pcm_runtime, list); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct device *dev = card->dev; + int ret; + + if (!data->sysclk) + return 0; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, data->sysclk, SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + return 0; +} + static int imx_rpmsg_probe(struct platform_device *pdev) { struct snd_soc_dai_link_component *dlc; @@ -72,12 +94,18 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->dai.codecs->dai_name = "snd-soc-dummy-dai"; data->dai.codecs->name = "snd-soc-dummy"; } else { + struct clk *clk; + data->dai.codecs->of_node = args.np; ret = snd_soc_get_dai_name(&args, &data->dai.codecs->dai_name); if (ret) { dev_err(&pdev->dev, "Unable to get codec_dai_name\n"); goto fail; } + + clk = devm_get_clk_from_child(&pdev->dev, args.np, NULL); + if (!IS_ERR(clk)) + data->sysclk = clk_get_rate(clk); } data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev); @@ -103,6 +131,7 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->card.owner = THIS_MODULE; data->card.dapm_widgets = imx_rpmsg_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_rpmsg_dapm_widgets); + data->card.late_probe = imx_rpmsg_late_probe; /* * Inoder to use common api to get card name and audio routing. * Use parent of_node for this device, revert it after finishing using -- 2.34.1

4 4

[PATCH v1] ALSA: hda: cs35l41: Clarify support for CSC3551 without _DSD Properties
by Stefan Binding 15 Aug '22

15 Aug '22

For devices which use HID CSC3551, correct ACPI _DSD properties are required to be able support those systems. Add error message to clarify this. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 129bffb431c2..b9d94f3a5b99 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1163,6 +1163,11 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; hw_cfg->gpio1.valid = true; } else { + /* + * Note: CLSA010(0/1) are special cases which use a slightly different design. + * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported. + */ + dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.", hid); hw_cfg->valid = false; hw_cfg->gpio1.valid = false; hw_cfg->gpio2.valid = false; -- 2.34.1

1 0

Re: [PATCH v1] ALSA: hda/realtek: Add quirks for ASUS Zenbooks using CS35L41
by Takashi Iwai 15 Aug '22

15 Aug '22

On Mon, 15 Aug 2022 16:19:53 +0200, Stefan Binding wrote: > > These Asus Zenbook laptop use Realtek HDA codec combined with > 2xCS35L41 Amplifiers using SPI. > > Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Thanks, applied. Takashi

1 0

[RFC PATCH] Soundwire: Initialize multi_link with fwnode props
by Khalid Masum 15 Aug '22

15 Aug '22

According to the TODO, In sw_bus_master_add, bus->multi_link is to be populated with properties from FW node props. Make this happen by creating a new fwnode_handle flag FWNODE_FLAG_MULTI_LINKED and use the flag to store the multi_link value from intel_link_startup. Use this flag to initialize bus->multi_link. Signed-off-by: Khalid Masum <khalid.masum.92(a)gmail.com> --- I do not think adding a new flag for fwnode_handle is a good idea. So, what would be the best way to initialize bus->multilink with fwnode props? -- Khalid Masum drivers/soundwire/bus.c | 4 ++-- drivers/soundwire/intel.c | 1 + include/linux/fwnode.h | 1 + 3 files changed, 4 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index a2bfb0434a67..80df1672c60b 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -74,9 +74,9 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, /* * Initialize multi_link flag - * TODO: populate this flag by reading property from FW node */ - bus->multi_link = false; + bus->multi_link = (fwnode->flags & FWNODE_FLAG_MULTI_LINKED) + == FWNODE_FLAG_MULTI_LINKED; if (bus->ops->read_prop) { ret = bus->ops->read_prop(bus); if (ret < 0) { diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 505c5ef061e3..034d1c523ddf 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1347,6 +1347,7 @@ int intel_link_startup(struct auxiliary_device *auxdev) */ bus->multi_link = true; bus->hw_sync_min_links = 1; + dev->fwnode->flags |= FWNODE_FLAG_MULTI_LINKED; } /* Initialize shim, controller */ diff --git a/include/linux/fwnode.h b/include/linux/fwnode.h index 9a81c4410b9f..446a52744953 100644 --- a/include/linux/fwnode.h +++ b/include/linux/fwnode.h @@ -32,6 +32,7 @@ struct device; #define FWNODE_FLAG_NOT_DEVICE BIT(1) #define FWNODE_FLAG_INITIALIZED BIT(2) #define FWNODE_FLAG_NEEDS_CHILD_BOUND_ON_ADD BIT(3) +#define FWNODE_FLAG_MULTI_LINKED BIT(4) struct fwnode_handle { struct fwnode_handle *secondary; -- 2.37.1

5 8

[PATCH] ASoC: soc-utils: Improve kerneldoc for snd_soc_tdm_params_to_bclk()
by Richard Fitzgerald 15 Aug '22

15 Aug '22

The statement that snd_soc_tdm_params_to_bclk() is equivalent to snd_soc_params_to_bclk() if tdm_width==tdm_slots==0 is not accurate, it is only true is slot_multiple is also <2. However, the description of special-case behaviour in terms of pairs of tdm_width and tdm_slot values is not particularly helpful so we might as well take the opportunity to rework the description to say the same thing in a simpler way. The behaviour of a pair of values is obvious from a description of each argument. At the same time make a few edits to clarify the rest of the description. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/soc-utils.c | 23 ++++++++++++----------- 1 file changed, 12 insertions(+), 11 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 70c380c0ac7b..a3b6df2378b4 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -56,23 +56,24 @@ EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); /** * snd_soc_tdm_params_to_bclk - calculate bclk from params and tdm slot info. * - * Calculate the bclk from the params sample rate and the tdm slot count and - * tdm slot width. Either or both of tdm_width and tdm_slots can be 0. + * Calculate the bclk from the params sample rate, the tdm slot count and the + * tdm slot width. Optionally round-up the slot count to a given multiple. + * Either or both of tdm_width and tdm_slots can be 0. * - * If tdm_width == 0 and tdm_slots > 0: the params_width will be used. - * If tdm_width > 0 and tdm_slots == 0: the params_channels will be used - * as the slot count. - * Both tdm_width and tdm_slots are 0: this is equivalent to calling - * snd_soc_params_to_bclk(). + * If tdm_width == 0: use params_width() as the slot width. + * If tdm_slots == 0: use params_channels() as the slot count. * - * If slot_multiple > 1 the slot count (or params_channels if tdm_slots == 0) - * will be rounded up to a multiple of this value. This is mainly useful for + * If slot_multiple > 1 the slot count (or params_channels() if tdm_slots == 0) + * will be rounded up to a multiple of slot_multiple. This is mainly useful for * I2S mode, which has a left and right phase so the number of slots is always * a multiple of 2. * + * If tdm_width == 0 && tdm_slots == 0 && slot_multiple < 2, this is equivalent + * to calling snd_soc_params_to_bclk(). + * * @params: Pointer to struct_pcm_hw_params. - * @tdm_width: Width in bits of the tdm slots. - * @tdm_slots: Number of tdm slots per frame. + * @tdm_width: Width in bits of the tdm slots. Must be >= 0. + * @tdm_slots: Number of tdm slots per frame. Must be >= 0. * @slot_multiple: If >1 roundup slot count to a multiple of this value. * * Return: bclk frequency in Hz, else a negative error code if params format -- 2.30.2

1 0

[PATCH] ASoC: Intel: sof_rt5682: Add support for jsl_rt5682_rt1019
by Sean Hong 15 Aug '22

15 Aug '22

This patch adds the driver data for rt5682s on SSP0 for ADL platform Signed-off-by: Sean Hong <sean.hong(a)quanta.corp-partner.google.com> --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 13 +++++++++++++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 045965312245b..3a840f3a9f5d2 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1100,6 +1100,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1019", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1019_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index b95c4b2cda947..139d2468f5f70 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -29,6 +29,11 @@ static const struct snd_soc_acpi_codecs rt1015p_spk = { .codecs = {"RTL1015"} }; +static struct snd_soc_acpi_codecs rt1019p_spk = { + .num_codecs = 1, + .codecs = {"RTL1019"} +} + static const struct snd_soc_acpi_codecs mx98360a_spk = { .num_codecs = 1, .codecs = {"MX98360A"} @@ -78,6 +83,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg", }, + { + .comp_ids = &rt5682_rt5682s_hp, + .drv_name = "jsl_rt5682_rt1019", + .sof_fw_filename = "sof-jsl.ri", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1019p_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", + }, { .id = "10134242", .drv_name = "jsl_cs4242_mx98360a", -- 2.25.1

4 3

[PATCH v3] ASoC: codecs: dt bind. doc for the new TI SRC4392 codec
by Matt Flax 15 Aug '22

15 Aug '22

Signed-off-by: Matt Flax <flatmax(a)flatmax.com> --- Notes: v3: * Renamed to include ti, * fixed license to include BSD-2-Clause * removed generic description * renamed the node in the example v2: * Changed the maintainer .../devicetree/bindings/sound/ti,src4xxx.yaml | 48 +++++++++++++++++++ 1 file changed, 48 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,src4xxx.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml new file mode 100644 index 000000000000..807b9e35b9b9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/src4xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments SRC4392 Device Tree Bindings + +description: | + The SRC4392 is a digital audio codec that can be connected via + I2C or SPI. Currently, only I2C bus is supported. + +maintainers: + - Matt Flax <flatmax(a)flatmax.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: ti,src4392 + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@70 { + #sound-dai-cells = <0>; + compatible = "ti,src4392"; + reg = <0x70>; + }; + }; +... -- 2.34.1

1 0

[PATCH AUTOSEL 4.9 4/8] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 31079ea5e484..d5d0e5e53920 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -457,4 +457,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index f2e8226c88fb..efe26b8ca57f 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -25,6 +25,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -153,3 +154,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 5/9] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index c333ceb80d5f..720d63574bfb 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -75,7 +75,7 @@ struct snd_timer_user { unsigned int filter; struct timespec tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1295,7 +1295,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1332,7 +1332,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1399,7 +1399,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1465,6 +1465,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) if (tu->timeri) snd_timer_close(tu->timeri); mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2017,7 +2018,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 4/9] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 4104a9d1001f..9d04e700b855 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -442,4 +442,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 0f818d593c9e..d100feba26b5 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -25,6 +25,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -160,3 +161,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 09/14] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 4920ec4f4594..f0e8b98f346e 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -75,7 +75,7 @@ struct snd_timer_user { unsigned int filter; struct timespec tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1306,7 +1306,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1343,7 +1343,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1410,7 +1410,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1476,6 +1476,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) if (tu->timeri) snd_timer_close(tu->timeri); mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2027,7 +2028,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 08/14] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 36a5934cf4b1..b5a8cc4d02cc 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -444,4 +444,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 0f818d593c9e..d100feba26b5 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -25,6 +25,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -160,3 +161,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 11/16] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index d684aa4150aa..420cc07a7f88 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -61,7 +61,7 @@ struct snd_timer_user { unsigned int filter; struct timespec tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1317,7 +1317,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1354,7 +1354,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1421,7 +1421,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1487,6 +1487,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) if (tu->timeri) snd_timer_close(tu->timeri); mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2050,7 +2051,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 10/16] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index ee238f100f73..8a80121811d9 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -440,4 +440,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 3579dd7a161f..c3f3d94b5197 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -10,6 +10,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -145,3 +146,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 13/19] ALSA: control: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4a971e84a7ae10a38d875cd2d4e487c8d1682ca3 ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from the control API. Note that it's merely a workaround. Another note: although we haven't received reports about the deadlock with the control API, the deadlock is still potentially possible, and it's better to align the behavior with other core APIs (PCM and timer); so let's move altogether. Link: https://lore.kernel.org/r/20220728125945.29533-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/control.h | 2 +- sound/core/control.c | 7 ++++--- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 77d9fa10812d..41bd72ffd232 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -103,7 +103,7 @@ struct snd_ctl_file { int preferred_subdevice[SND_CTL_SUBDEV_ITEMS]; wait_queue_head_t change_sleep; spinlock_t read_lock; - struct fasync_struct *fasync; + struct snd_fasync *fasync; int subscribed; /* read interface is activated */ struct list_head events; /* waiting events for read */ }; diff --git a/sound/core/control.c b/sound/core/control.c index 3b44378b9dec..732eb515d2f5 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -121,6 +121,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) if (control->vd[idx].owner == ctl) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); + snd_fasync_free(ctl->fasync); snd_ctl_empty_read_queue(ctl); put_pid(ctl->pid); kfree(ctl); @@ -175,7 +176,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, _found: wake_up(&ctl->change_sleep); spin_unlock(&ctl->read_lock); - kill_fasync(&ctl->fasync, SIGIO, POLL_IN); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_IN); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } @@ -1941,7 +1942,7 @@ static int snd_ctl_fasync(int fd, struct file * file, int on) struct snd_ctl_file *ctl; ctl = file->private_data; - return fasync_helper(fd, file, on, &ctl->fasync); + return snd_fasync_helper(fd, file, on, &ctl->fasync); } /* return the preferred subdevice number if already assigned; @@ -2015,7 +2016,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); - kill_fasync(&ctl->fasync, SIGIO, POLL_ERR); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_ERR); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 12/19] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 04cd8953605a..764d2b19344e 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -83,7 +83,7 @@ struct snd_timer_user { unsigned int filter; struct timespec64 tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1345,7 +1345,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1383,7 +1383,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1453,7 +1453,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1521,6 +1521,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) snd_timer_instance_free(tu->timeri); } mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2135,7 +2136,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 11/19] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 0462c577d7a3..85610ede9ea0 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -446,4 +446,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 3579dd7a161f..c3f3d94b5197 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -10,6 +10,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -145,3 +146,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 21/28] ALSA: control: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4a971e84a7ae10a38d875cd2d4e487c8d1682ca3 ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from the control API. Note that it's merely a workaround. Another note: although we haven't received reports about the deadlock with the control API, the deadlock is still potentially possible, and it's better to align the behavior with other core APIs (PCM and timer); so let's move altogether. Link: https://lore.kernel.org/r/20220728125945.29533-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/control.h | 2 +- sound/core/control.c | 7 ++++--- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 985c51a8fb74..a1fc7e0a47d9 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -109,7 +109,7 @@ struct snd_ctl_file { int preferred_subdevice[SND_CTL_SUBDEV_ITEMS]; wait_queue_head_t change_sleep; spinlock_t read_lock; - struct fasync_struct *fasync; + struct snd_fasync *fasync; int subscribed; /* read interface is activated */ struct list_head events; /* waiting events for read */ }; diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..f66fe4be30d3 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -127,6 +127,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) if (control->vd[idx].owner == ctl) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); + snd_fasync_free(ctl->fasync); snd_ctl_empty_read_queue(ctl); put_pid(ctl->pid); kfree(ctl); @@ -181,7 +182,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, _found: wake_up(&ctl->change_sleep); spin_unlock(&ctl->read_lock); - kill_fasync(&ctl->fasync, SIGIO, POLL_IN); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_IN); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } @@ -2002,7 +2003,7 @@ static int snd_ctl_fasync(int fd, struct file * file, int on) struct snd_ctl_file *ctl; ctl = file->private_data; - return fasync_helper(fd, file, on, &ctl->fasync); + return snd_fasync_helper(fd, file, on, &ctl->fasync); } /* return the preferred subdevice number if already assigned; @@ -2170,7 +2171,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); - kill_fasync(&ctl->fasync, SIGIO, POLL_ERR); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_ERR); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 20/28] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index b3214baa8919..e08a37c23add 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -83,7 +83,7 @@ struct snd_timer_user { unsigned int filter; struct timespec64 tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1345,7 +1345,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1383,7 +1383,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1453,7 +1453,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1521,6 +1521,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) snd_timer_instance_free(tu->timeri); } mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2135,7 +2136,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 19/28] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 6d4cc49584c6..39cee40ac22e 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -501,4 +501,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 50e4aaa6270d..d32a19976a2b 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -10,6 +10,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -145,3 +146,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 08/28] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_irq_ctrl()
by Sasha Levin 14 Aug '22

14 Aug '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit ef30911d3c39fd57884c348c29b9cbff88def155 ] Before, ssiu.c didn't care SSI5-8, thus, commit b1384d4c95088d0 ("ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()") cares it for status clear. But we should care it for error irq handling, too. This patch cares it. Reported-by: Nguyen Bao Nguyen <nguyen.nguyen.yj(a)renesas.com> Reported-by: Nishiyama Kunihiko <kunihiko.nishiyama.dn(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/871quocio1.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4b8a63e336c7..d7f4646ee029 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -67,6 +67,8 @@ static void rsnd_ssiu_busif_err_irq_ctrl(struct rsnd_mod *mod, int enable) shift = 1; offset = 1; break; + default: + return; } for (i = 0; i < 4; i++) { -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 32/39] ALSA: control: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4a971e84a7ae10a38d875cd2d4e487c8d1682ca3 ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from the control API. Note that it's merely a workaround. Another note: although we haven't received reports about the deadlock with the control API, the deadlock is still potentially possible, and it's better to align the behavior with other core APIs (PCM and timer); so let's move altogether. Link: https://lore.kernel.org/r/20220728125945.29533-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/control.h | 2 +- sound/core/control.c | 7 ++++--- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 985c51a8fb74..a1fc7e0a47d9 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -109,7 +109,7 @@ struct snd_ctl_file { int preferred_subdevice[SND_CTL_SUBDEV_ITEMS]; wait_queue_head_t change_sleep; spinlock_t read_lock; - struct fasync_struct *fasync; + struct snd_fasync *fasync; int subscribed; /* read interface is activated */ struct list_head events; /* waiting events for read */ }; diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..f66fe4be30d3 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -127,6 +127,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) if (control->vd[idx].owner == ctl) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); + snd_fasync_free(ctl->fasync); snd_ctl_empty_read_queue(ctl); put_pid(ctl->pid); kfree(ctl); @@ -181,7 +182,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, _found: wake_up(&ctl->change_sleep); spin_unlock(&ctl->read_lock); - kill_fasync(&ctl->fasync, SIGIO, POLL_IN); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_IN); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } @@ -2002,7 +2003,7 @@ static int snd_ctl_fasync(int fd, struct file * file, int on) struct snd_ctl_file *ctl; ctl = file->private_data; - return fasync_helper(fd, file, on, &ctl->fasync); + return snd_fasync_helper(fd, file, on, &ctl->fasync); } /* return the preferred subdevice number if already assigned; @@ -2170,7 +2171,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); - kill_fasync(&ctl->fasync, SIGIO, POLL_ERR); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_ERR); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 31/39] ALSA: pcm: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 96b097091c66df4f6fbf5cbff21df6cc02a2f055 ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from timer API. Note that it's merely a workaround. Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Reported-by: syzbot+669c9abf11a6a011dd09(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/pcm.h | 2 +- sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 2 +- 4 files changed, 4 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 6b99310b5b88..6987110843f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -399,7 +399,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t twake; /* do transfer (!poll) wakeup if non-zero */ wait_queue_head_t sleep; /* poll sleep */ wait_queue_head_t tsleep; /* transfer sleep */ - struct fasync_struct *fasync; + struct snd_fasync *fasync; bool stop_operating; /* sync_stop will be called */ struct mutex buffer_mutex; /* protect for buffer changes */ atomic_t buffer_accessing; /* >0: in r/w operation, <0: blocked */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 977d54320a5c..c917ac84a7e5 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1005,6 +1005,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) substream->runtime = NULL; } mutex_destroy(&runtime->buffer_mutex); + snd_fasync_free(runtime->fasync); kfree(runtime); put_pid(substream->pid); substream->pid = NULL; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 1fc7c50ffa62..40751e5aff09 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1822,7 +1822,7 @@ void snd_pcm_period_elapsed_under_stream_lock(struct snd_pcm_substream *substrea snd_timer_interrupt(substream->timer, 1); #endif _end: - kill_fasync(&runtime->fasync, SIGIO, POLL_IN); + snd_kill_fasync(runtime->fasync, SIGIO, POLL_IN); } EXPORT_SYMBOL(snd_pcm_period_elapsed_under_stream_lock); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4adaee62ef33..16fcf57c6f03 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3945,7 +3945,7 @@ static int snd_pcm_fasync(int fd, struct file * file, int on) runtime = substream->runtime; if (runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; - return fasync_helper(fd, file, on, &runtime->fasync); + return snd_fasync_helper(fd, file, on, &runtime->fasync); } /* -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 30/39] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index b3214baa8919..e08a37c23add 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -83,7 +83,7 @@ struct snd_timer_user { unsigned int filter; struct timespec64 tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1345,7 +1345,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1383,7 +1383,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1453,7 +1453,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1521,6 +1521,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) snd_timer_instance_free(tu->timeri); } mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2135,7 +2136,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 29/39] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 6d4cc49584c6..39cee40ac22e 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -501,4 +501,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 50e4aaa6270d..d32a19976a2b 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -10,6 +10,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -145,3 +146,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 26/39] ASoC: codecs: va-macro: use fsgen as clock
by Sasha Levin 14 Aug '22

14 Aug '22

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 30097967e0566cac817273ef76add100f6b0f463 ] VA Macro fsgen clock is supplied to other LPASS Macros using proper clock apis, however the internal user uses the registers directly without clk apis. This approch has race condition where in external users of the clock might cut the clock while VA macro is actively using this. Moving the internal usage to clk apis would provide a proper refcounting and avoid such race conditions. This issue was noticed while headset was pulled out while recording is in progress and shifting record patch to DMIC. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Tested-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Link: https://lore.kernel.org/r/20220727124749.4604-1-srinivas.kandagatla@linaro.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-va-macro.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index d18b56e60433..1ea10dc70748 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -199,6 +199,7 @@ struct va_macro { struct clk *mclk; struct clk *macro; struct clk *dcodec; + struct clk *fsgen; struct clk_hw hw; struct lpass_macro *pds; @@ -467,9 +468,9 @@ static int va_macro_mclk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - return va_macro_mclk_enable(va, true); + return clk_prepare_enable(va->fsgen); case SND_SOC_DAPM_POST_PMD: - return va_macro_mclk_enable(va, false); + clk_disable_unprepare(va->fsgen); } return 0; @@ -1473,6 +1474,12 @@ static int va_macro_probe(struct platform_device *pdev) if (ret) goto err_clkout; + va->fsgen = clk_hw_get_clk(&va->hw, "fsgen"); + if (IS_ERR(va->fsgen)) { + ret = PTR_ERR(va->fsgen); + goto err_clkout; + } + ret = devm_snd_soc_register_component(dev, &va_macro_component_drv, va_macro_dais, ARRAY_SIZE(va_macro_dais)); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 20/39] ASoC: Intel: sof_nau8825: Move quirk check to the front in late probe
by Sasha Levin 14 Aug '22

14 Aug '22

From: Yong Zhi <yong.zhi(a)intel.com> [ Upstream commit 5b56db90bbaf9d8581e5e6268727d8ad706555e4 ] The sof_rt5682_quirk check was placed in the middle of hdmi handling code, move it to the front to be consistent with sof_rt5682.c/sof_card_late_probe(). Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220725194909.145418-11-pierre-louis.bossart@lin… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_nau8825.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 33de043b66c6..b6826d6572b8 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -177,11 +177,6 @@ static int sof_card_late_probe(struct snd_soc_card *card) struct sof_hdmi_pcm *pcm; int err; - if (list_empty(&ctx->hdmi_pcm_list)) - return -EINVAL; - - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); - if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { /* Disable Left and Right Spk pin after boot */ snd_soc_dapm_disable_pin(dapm, "Left Spk"); @@ -191,6 +186,11 @@ static int sof_card_late_probe(struct snd_soc_card *card) return err; } + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; + + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); + return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); } -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 19/39] ASoC: Intel: sof_es8336: ignore GpioInt when looking for speaker/headset GPIO lines
by Sasha Levin 14 Aug '22

14 Aug '22

From: Andrey Turkin <andrey.turkin(a)gmail.com> [ Upstream commit 751e77011f7a43a204bf2a5d02fbf5f8219bc531 ] This fixes speaker GPIO detection on machines those ACPI tables list their jack detection GpioInt before output GpioIo. GpioInt entry can never be the speaker/headphone amplifier control so it makes sense to only look for GpioIo entries when looking for them. Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Andrey Turkin <andrey.turkin(a)gmail.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220725194909.145418-5-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_es8336.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 81e12f03ec97..dade1749c444 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -57,23 +57,23 @@ static const struct acpi_gpio_params enable_gpio0 = { 0, 0, true }; static const struct acpi_gpio_params enable_gpio1 = { 1, 0, true }; static const struct acpi_gpio_mapping acpi_speakers_enable_gpio0[] = { - { "speakers-enable-gpios", &enable_gpio0, 1 }, + { "speakers-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; static const struct acpi_gpio_mapping acpi_speakers_enable_gpio1[] = { - { "speakers-enable-gpios", &enable_gpio1, 1 }, + { "speakers-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, }; static const struct acpi_gpio_mapping acpi_enable_both_gpios[] = { - { "speakers-enable-gpios", &enable_gpio0, 1 }, - { "headphone-enable-gpios", &enable_gpio1, 1 }, + { "speakers-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, + { "headphone-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; static const struct acpi_gpio_mapping acpi_enable_both_gpios_rev_order[] = { - { "speakers-enable-gpios", &enable_gpio1, 1 }, - { "headphone-enable-gpios", &enable_gpio0, 1 }, + { "speakers-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, + { "headphone-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; -- 2.35.1

1 0