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how to get the log file of aplay
by GitHub issues - opened 17 Nov '21

17 Nov '21

alsa-project/alsa-utils issue #119 was opened from MT-LeiZhao: Hi guys, I'm a new guy for audio driver. I have added logs in my ko which have the pcm function. I can see the normal call stack log using pulseaudio. But I can't see any log about my driver through aplay. I guess there was something wrong in aplay. Can anyone tell me how to get the detail log for aplay? Thank you very much~~~ Issue URL : https://github.com/alsa-project/alsa-utils/issues/119 Repository URL: https://github.com/alsa-project/alsa-utils

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HDMI driver format channel mismatch bug
by sujith kumar reddy 17 Nov '21

17 Nov '21

Hi Alsa Team, Sound is not coming in sony tv . which has below supported formats and channels. please find the attached edid information and hw_params dump. with intel platform, same monitor populating 2 channels as max in hw_params in dump. but with AMD card, its populating as 6 channels. When we digged the code, we found that snd_hdmi_eld_update_pcm_info API setting hinfo->channel_max as 8 and as channels max retrieved from sad info as 6. for LPCM , why the channel max assignment logic is not safeguared with audio format check as LPCM ? snippet code: /* update PCM info based on ELD */void snd_hdmi_eld_update_pcm_info(struct parsed_hdmi_eld *e, struct hda_pcm_stream *hinfo){ u32 rates; u64 formats; unsigned int maxbps; unsigned int channels_max; int i; /* assume basic audio support (the basic audio flag is not in ELD; * however, all audio capable sinks are required to support basic * audio) */ rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; formats = SNDRV_PCM_FMTBIT_S16_LE; maxbps = 16; channels_max = 2; for (i = 0; i < e->sad_count; i++) { struct cea_sad *a = &e->sad[i]; rates |= a->rates; if (a->channels > channels_max) channels_max = a->channels; if (a->format == AUDIO_CODING_TYPE_LPCM) { if (a->sample_bits & AC_SUPPCM_BITS_20) { formats |= SNDRV_PCM_FMTBIT_S32_LE; if (maxbps < 20) maxbps = 20; } if (a->sample_bits & AC_SUPPCM_BITS_24) { formats |= SNDRV_PCM_FMTBIT_S32_LE; if (maxbps < 24) maxbps = 24; } } } /* restrict the parameters by the values the codec provides */ hinfo->rates &= rates; hinfo->formats &= formats; hinfo->maxbps = min(hinfo->maxbps, maxbps); hinfo->channels_max = min(hinfo->channels_max, channels_max); >>>>>>>>>>the above statement channels_max is the maximum of channels supported of different formats. For example: the below attached edid is of sony tv. which supports two formats(LPCM and AC3) LPCM --->supports 2 channels AC3 ----->Supports 6 channels you can see in the attached edid info. As AMD supports LPCM ...>when we keep logs here we got channels_max =6 .in which sound is not observed on tv .when we hardcode to 2 channels the sound is heard from monitor. As you see the above code ..for loop is not distinguishing channels_max for different formats. } Thanks in Advance https://pastebin.com/t41DqErH ------edid information https://pastebin.com/4m56h10R -----dump hw-params Thanks Sujith

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[PATCH] ASoC: qcom: sdm845: only setup slim ports once
by Srinivas Kandagatla 16 Nov '21

16 Nov '21

Currently same slim channel map setup for every dai link, which is redundant. Fix this by adding a flag and conditionally setting these channel maps. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/sdm845.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 0adfc5708949..a1d5d96744b5 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -33,6 +33,7 @@ struct sdm845_snd_data { struct snd_soc_jack jack; bool jack_setup; + bool slim_port_setup; bool stream_prepared[AFE_PORT_MAX]; struct snd_soc_card *card; uint32_t pri_mi2s_clk_count; @@ -224,6 +225,7 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); + struct snd_soc_dai_link *link = rtd->dai_link; struct snd_jack *jack; /* * Codec SLIMBUS configuration @@ -276,6 +278,10 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) } break; case SLIMBUS_0_RX...SLIMBUS_6_TX: + /* setting up wcd multiple times for slim port is redundant */ + if (pdata->slim_port_setup || !link->no_pcm) + return 0; + for_each_rtd_codec_dais(rtd, i, codec_dai) { rval = snd_soc_dai_set_channel_map(codec_dai, ARRAY_SIZE(tx_ch), @@ -295,8 +301,10 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) dev_warn(card->dev, "Failed to set jack: %d\n", rval); return rval; } - } + + pdata->slim_port_setup = true; + break; default: break; -- 2.21.0

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[PATCH] ASoC: SOF: i.MX: simplify Kconfig
by Daniel Baluta 16 Nov '21

16 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Follow the Intel example and simplify the Kconfig a) start from the end-product for 'select' chains b) use 'depends on' to filter out configurations. c) use snd-sof-of as a common module without any 'select' Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/sof/Kconfig | 4 +++- sound/soc/sof/Makefile | 2 +- sound/soc/sof/imx/Kconfig | 46 +++++++++++---------------------------- 3 files changed, 17 insertions(+), 35 deletions(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 041c54639c4d..b6fa659179b6 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -40,12 +40,14 @@ config SND_SOC_SOF_ACPI_DEV config SND_SOC_SOF_OF tristate "SOF OF enumeration support" depends on OF || COMPILE_TEST - select SND_SOC_SOF help This adds support for Device Tree enumeration. This option is required to enable i.MX8 devices. Say Y if you need this option. If unsure select "N". +config SND_SOC_SOF_OF_DEV + tristate + config SND_SOC_SOF_COMPRESS bool select SND_SOC_COMPRESS diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 06e5f49f7ee8..1dac5cb4dfd6 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -17,7 +17,7 @@ obj-$(CONFIG_SND_SOC_SOF_NOCODEC) += snd-sof-nocodec.o obj-$(CONFIG_SND_SOC_SOF_ACPI_DEV) += snd-sof-acpi.o -obj-$(CONFIG_SND_SOC_SOF_OF) += snd-sof-of.o +obj-$(CONFIG_SND_SOC_SOF_OF_DEV) += snd-sof-of.o obj-$(CONFIG_SND_SOC_SOF_PCI_DEV) += snd-sof-pci.o obj-$(CONFIG_SND_SOC_SOF_INTEL_TOPLEVEL) += intel/ diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 34cf228c188f..9b8d5bb1e449 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -11,53 +11,33 @@ config SND_SOC_SOF_IMX_TOPLEVEL if SND_SOC_SOF_IMX_TOPLEVEL -config SND_SOC_SOF_IMX_OF - def_tristate SND_SOC_SOF_OF - select SND_SOC_SOF_IMX8 if SND_SOC_SOF_IMX8_SUPPORT - select SND_SOC_SOF_IMX8M if SND_SOC_SOF_IMX8M_SUPPORT - help - This option is not user-selectable but automagically handled by - 'select' statements at a higher level. - config SND_SOC_SOF_IMX_COMMON tristate + select SND_SOC_SOF_OF_DEV + select SND_SOC_SOF + select SND_SOC_SOF_XTENSA + select SND_SOC_SOF_COMPRESS help This option is not user-selectable but automagically handled by 'select' statements at a higher level. -config SND_SOC_SOF_IMX8_SUPPORT - bool "SOF support for i.MX8" - depends on IMX_SCU=y || IMX_SCU=SND_SOC_SOF_IMX_OF - depends on IMX_DSP=y || IMX_DSP=SND_SOC_SOF_IMX_OF +config SND_SOC_SOF_IMX8 + tristate "SOF support for i.MX8" + depends on IMX_SCU + depends on IMX_DSP + select SND_SOC_SOF_IMX_COMMON help This adds support for Sound Open Firmware for NXP i.MX8 platforms. Say Y if you have such a device. If unsure select "N". -config SND_SOC_SOF_IMX8 - tristate +config SND_SOC_SOF_IMX8M + tristate "SOF support for i.MX8M" + depends on IMX_DSP select SND_SOC_SOF_IMX_COMMON - select SND_SOC_SOF_XTENSA - select SND_SOC_SOF_COMPRESS - help - This option is not user-selectable but automagically handled by - 'select' statements at a higher level. - -config SND_SOC_SOF_IMX8M_SUPPORT - bool "SOF support for i.MX8M" - depends on IMX_DSP=y || IMX_DSP=SND_SOC_SOF_OF help This adds support for Sound Open Firmware for NXP i.MX8M platforms. Say Y if you have such a device. If unsure select "N". -config SND_SOC_SOF_IMX8M - tristate - select SND_SOC_SOF_IMX_COMMON - select SND_SOC_SOF_XTENSA - select SND_SOC_SOF_COMPRESS - help - This option is not user-selectable but automagically handled by - 'select' statements at a higher level. - -endif ## SND_SOC_SOF_IMX_IMX_TOPLEVEL +endif ## SND_SOC_SOF_IMX_TOPLEVEL -- 2.27.0

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[PATCH 0/5] ASoC: qdsp6: fixes and improve error reporting
by Srinivas Kandagatla 16 Nov '21

16 Nov '21

Hi Mark, During testing on a laptop we found few issues with the existing q6dsp code, first 3 patches should fix these issues. Also during debug we found that some of the errors reported are not very useful, so update error reporting in two places. Thanks, srini Srinivas Kandagatla (5): ASoC: qdsp6: qdsp6: q6prm: handle clk disable correctly ASoC: qdsp6: q6routing: Conditionally reset FrontEnd Mixer ASoC: qdsp6: q6asm: fix q6asm_dai_prepare error handling ASoC: qdsp6: q6adm: improve error reporting ASoC: qdsp6: q6routing: validate port id before setting up route sound/soc/qcom/qdsp6/audioreach.h | 4 +++ sound/soc/qcom/qdsp6/q6adm.c | 4 +-- sound/soc/qcom/qdsp6/q6asm-dai.c | 19 +++++++---- sound/soc/qcom/qdsp6/q6prm.c | 53 +++++++++++++++++++++++++++++-- sound/soc/qcom/qdsp6/q6routing.c | 12 ++++++- 5 files changed, 81 insertions(+), 11 deletions(-) -- 2.21.0

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[PATCH 0/3] ASoC: codecs: Qualcomm codecs fixes
by Srinivas Kandagatla 16 Nov '21

16 Nov '21

Hi Mark, We found few issues during runtime pm testing on this codecs. These 3 patches for wcd938x, wcd934x and lpass-rx-macro should address them. Thanks, srini Srinivas Kandagatla (3): ASoC: codecs: wcd938x: fix volatile register range ASoC: codecs: wcd934x: return error code correctly from hw_params ASoC: codecs: lpass-rx-macro: fix HPHR setting CLSH mask sound/soc/codecs/lpass-rx-macro.c | 2 +- sound/soc/codecs/wcd934x.c | 3 +-- sound/soc/codecs/wcd938x.c | 3 +++ 3 files changed, 5 insertions(+), 3 deletions(-) -- 2.21.0

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[PATCH] ASoC: topology: Add missing rwsem around snd_ctl_remove() calls
by Takashi Iwai 16 Nov '21

16 Nov '21

snd_ctl_remove() has to be called with card->controls_rwsem held (when called after the card instantiation). This patch add the missing rwsem calls around it. Fixes: 8a9782346dcc ("ASoC: topology: Add topology core") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-topology.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 557e22c5254c..f5b9e66ac3b8 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2700,6 +2700,7 @@ EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load); /* remove dynamic controls from the component driver */ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) { + struct snd_card *card = comp->card->snd_card; struct snd_soc_dobj *dobj, *next_dobj; int pass = SOC_TPLG_PASS_END; @@ -2707,6 +2708,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) while (pass >= SOC_TPLG_PASS_START) { /* remove mixer controls */ + down_write(&card->controls_rwsem); list_for_each_entry_safe(dobj, next_dobj, &comp->dobj_list, list) { @@ -2745,6 +2747,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp) break; } } + up_write(&card->controls_rwsem); pass--; } -- 2.31.1

2 1

[PATCH 0/4] ASoC: cs42l42: Remove redundant code
by Richard Fitzgerald 16 Nov '21

16 Nov '21

- Remove field writes in regmap_update_bits() calls where the code intends that the field value is not changed. - Remove unnecessary complexity from reporting jack unplug event - Remove a PLL config value member that was introduced in a bugfix but made redundant by a later bugfix. Richard Fitzgerald (4): ASoC: cs42l42: Remove redundant writes to DETECT_MODE ASoC: cs42l42: Remove redundant writes to RS_PLUG/RS_UNPLUG masks ASoC: cs42l42: Simplify reporting of jack unplug ASoC: cs42l42: Remove redundant pll_divout member sound/soc/codecs/cs42l42.c | 42 +++++++----------------------------------- sound/soc/codecs/cs42l42.h | 1 - 2 files changed, 7 insertions(+), 36 deletions(-) -- 2.11.0

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[PATCH 0/4] ASoC: mediatek: Update MT8195 machine driver
by Trevor Wu 16 Nov '21

16 Nov '21

Thes series of patches adds support for RT5682s headset codec and create a new machine driver for SOF support. Patches are based on broonie tree "for-next" branch. Trevor Wu (4): ASoC: mediatek: mt8195: add headset codec rt5682s support dt-bindings: mediatek: mt8195: add model property ASoC: mediatek: mt8195: separate the common code from machine driver ASoC: mediatek: mt8195: add machine driver for MT8195 SOF support .../sound/mt8195-mt6359-rt1011-rt5682.yaml | 4 + .../sound/mt8195-mt6359-rt1019-rt5682.yaml | 4 + sound/soc/mediatek/Kconfig | 2 + sound/soc/mediatek/mt8195/Makefile | 16 +- sound/soc/mediatek/mt8195/mt8195-common.c | 398 ++++++++++++ sound/soc/mediatek/mt8195/mt8195-common.h | 33 + .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 591 +----------------- .../mt8195/mt8195-mt6359-rt1019-rt5682-sof.c | 557 +++++++++++++++++ .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 551 +--------------- .../mediatek/mt8195/mt8195-realtek-common.c | 161 +++++ .../mediatek/mt8195/mt8195-realtek-common.h | 34 + 11 files changed, 1267 insertions(+), 1084 deletions(-) create mode 100644 sound/soc/mediatek/mt8195/mt8195-common.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-common.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682-sof.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-realtek-common.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-realtek-common.h -- 2.18.0

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[PATCH] ALSA: memalloc: Unify x86 SG-buffer handling (take#3)
by Takashi Iwai 16 Nov '21

16 Nov '21

This is a second attempt to unify the x86-specific SG-buffer handling code with the new standard non-contiguous page handler. The first try (in commit 2d9ea39917a4) failed due to the wrong page and address calculations, hence reverted. (And the second try failed due to a copy&paste error.) Now it's corrected with the previous fix for noncontig pages, and the proper sg page iteration by this patch. After the migration, SNDRV_DMA_TYPE_DMA_SG becomes identical with SNDRV_DMA_TYPE_NONCONTIG on x86, while others still fall back to SNDRV_DMA_TYPE_DEV. Tested-by: Alex Xu (Hello71) <alex_y_xu(a)yahoo.ca> Tested-by: Harald Arnesen <harald(a)skogtun.org> Link: https://lore.kernel.org/r/20211017074859.24112-4-tiwai@suse.de Link: https://lore.kernel.org/r/20211109062235.22310-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 14 +-- sound/core/Makefile | 1 - sound/core/memalloc.c | 53 ++++++++++- sound/core/sgbuf.c | 201 --------------------------------------- 4 files changed, 56 insertions(+), 213 deletions(-) delete mode 100644 sound/core/sgbuf.c diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 1051b84e8579..653dfffb3ac8 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -36,13 +36,6 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */ #define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */ #define SNDRV_DMA_TYPE_DEV_WC 5 /* continuous write-combined */ -#ifdef CONFIG_SND_DMA_SGBUF -#define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */ -#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ -#else -#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ -#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC -#endif #ifdef CONFIG_GENERIC_ALLOCATOR #define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ #else @@ -51,6 +44,13 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_VMALLOC 7 /* vmalloc'ed buffer */ #define SNDRV_DMA_TYPE_NONCONTIG 8 /* non-coherent SG buffer */ #define SNDRV_DMA_TYPE_NONCOHERENT 9 /* non-coherent buffer */ +#ifdef CONFIG_SND_DMA_SGBUF +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_NONCONTIG +#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ +#else +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ +#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC +#endif /* * info for buffer allocation diff --git a/sound/core/Makefile b/sound/core/Makefile index 79e1407cd0de..350d704ced98 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -19,7 +19,6 @@ snd-$(CONFIG_SND_JACK) += ctljack.o jack.o snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \ pcm_memory.o memalloc.o snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o -snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 9fc971a704a9..d1fcd1d5adae 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -620,6 +620,52 @@ static const struct snd_malloc_ops snd_dma_noncontig_ops = { .get_chunk_size = snd_dma_noncontig_get_chunk_size, }; +/* x86-specific SG-buffer with WC pages */ +#ifdef CONFIG_SND_DMA_SGBUF +#define sg_wc_address(it) ((unsigned long)page_address(sg_page_iter_page(it))) + +static void *snd_dma_sg_wc_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + void *p = snd_dma_noncontig_alloc(dmab, size); + struct sg_table *sgt = dmab->private_data; + struct sg_page_iter iter; + + if (!p) + return NULL; + for_each_sgtable_page(sgt, &iter, 0) + set_memory_wc(sg_wc_address(&iter), 1); + return p; +} + +static void snd_dma_sg_wc_free(struct snd_dma_buffer *dmab) +{ + struct sg_table *sgt = dmab->private_data; + struct sg_page_iter iter; + + for_each_sgtable_page(sgt, &iter, 0) + set_memory_wb(sg_wc_address(&iter), 1); + snd_dma_noncontig_free(dmab); +} + +static int snd_dma_sg_wc_mmap(struct snd_dma_buffer *dmab, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return dma_mmap_noncontiguous(dmab->dev.dev, area, + dmab->bytes, dmab->private_data); +} + +static const struct snd_malloc_ops snd_dma_sg_wc_ops = { + .alloc = snd_dma_sg_wc_alloc, + .free = snd_dma_sg_wc_free, + .mmap = snd_dma_sg_wc_mmap, + .sync = snd_dma_noncontig_sync, + .get_addr = snd_dma_noncontig_get_addr, + .get_page = snd_dma_noncontig_get_page, + .get_chunk_size = snd_dma_noncontig_get_chunk_size, +}; +#endif /* CONFIG_SND_DMA_SGBUF */ + /* * Non-coherent pages allocator */ @@ -679,14 +725,13 @@ static const struct snd_malloc_ops *dma_ops[] = { [SNDRV_DMA_TYPE_DEV_WC] = &snd_dma_wc_ops, [SNDRV_DMA_TYPE_NONCONTIG] = &snd_dma_noncontig_ops, [SNDRV_DMA_TYPE_NONCOHERENT] = &snd_dma_noncoherent_ops, +#ifdef CONFIG_SND_DMA_SGBUF + [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_wc_ops, +#endif #ifdef CONFIG_GENERIC_ALLOCATOR [SNDRV_DMA_TYPE_DEV_IRAM] = &snd_dma_iram_ops, #endif /* CONFIG_GENERIC_ALLOCATOR */ #endif /* CONFIG_HAS_DMA */ -#ifdef CONFIG_SND_DMA_SGBUF - [SNDRV_DMA_TYPE_DEV_SG] = &snd_dma_sg_ops, - [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_ops, -#endif }; static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab) diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c deleted file mode 100644 index 8352a5cdb19f..000000000000 --- a/sound/core/sgbuf.c +++ /dev/null @@ -1,201 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * Scatter-Gather buffer - * - * Copyright (c) by Takashi Iwai <tiwai(a)suse.de> - */ - -#include <linux/slab.h> -#include <linux/mm.h> -#include <linux/vmalloc.h> -#include <linux/export.h> -#include <sound/memalloc.h> -#include "memalloc_local.h" - -struct snd_sg_page { - void *buf; - dma_addr_t addr; -}; - -struct snd_sg_buf { - int size; /* allocated byte size */ - int pages; /* allocated pages */ - int tblsize; /* allocated table size */ - struct snd_sg_page *table; /* address table */ - struct page **page_table; /* page table (for vmap/vunmap) */ - struct device *dev; -}; - -/* table entries are align to 32 */ -#define SGBUF_TBL_ALIGN 32 -#define sgbuf_align_table(tbl) ALIGN((tbl), SGBUF_TBL_ALIGN) - -static void snd_dma_sg_free(struct snd_dma_buffer *dmab) -{ - struct snd_sg_buf *sgbuf = dmab->private_data; - struct snd_dma_buffer tmpb; - int i; - - if (!sgbuf) - return; - - vunmap(dmab->area); - dmab->area = NULL; - - tmpb.dev.type = SNDRV_DMA_TYPE_DEV; - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) - tmpb.dev.type = SNDRV_DMA_TYPE_DEV_WC; - tmpb.dev.dev = sgbuf->dev; - for (i = 0; i < sgbuf->pages; i++) { - if (!(sgbuf->table[i].addr & ~PAGE_MASK)) - continue; /* continuous pages */ - tmpb.area = sgbuf->table[i].buf; - tmpb.addr = sgbuf->table[i].addr & PAGE_MASK; - tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT; - snd_dma_free_pages(&tmpb); - } - - kfree(sgbuf->table); - kfree(sgbuf->page_table); - kfree(sgbuf); - dmab->private_data = NULL; -} - -#define MAX_ALLOC_PAGES 32 - -static void *snd_dma_sg_alloc(struct snd_dma_buffer *dmab, size_t size) -{ - struct snd_sg_buf *sgbuf; - unsigned int i, pages, chunk, maxpages; - struct snd_dma_buffer tmpb; - struct snd_sg_page *table; - struct page **pgtable; - int type = SNDRV_DMA_TYPE_DEV; - pgprot_t prot = PAGE_KERNEL; - void *area; - - dmab->private_data = sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); - if (!sgbuf) - return NULL; - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) { - type = SNDRV_DMA_TYPE_DEV_WC; -#ifdef pgprot_noncached - prot = pgprot_noncached(PAGE_KERNEL); -#endif - } - sgbuf->dev = dmab->dev.dev; - pages = snd_sgbuf_aligned_pages(size); - sgbuf->tblsize = sgbuf_align_table(pages); - table = kcalloc(sgbuf->tblsize, sizeof(*table), GFP_KERNEL); - if (!table) - goto _failed; - sgbuf->table = table; - pgtable = kcalloc(sgbuf->tblsize, sizeof(*pgtable), GFP_KERNEL); - if (!pgtable) - goto _failed; - sgbuf->page_table = pgtable; - - /* allocate pages */ - maxpages = MAX_ALLOC_PAGES; - while (pages > 0) { - chunk = pages; - /* don't be too eager to take a huge chunk */ - if (chunk > maxpages) - chunk = maxpages; - chunk <<= PAGE_SHIFT; - if (snd_dma_alloc_pages_fallback(type, dmab->dev.dev, - chunk, &tmpb) < 0) { - if (!sgbuf->pages) - goto _failed; - size = sgbuf->pages * PAGE_SIZE; - break; - } - chunk = tmpb.bytes >> PAGE_SHIFT; - for (i = 0; i < chunk; i++) { - table->buf = tmpb.area; - table->addr = tmpb.addr; - if (!i) - table->addr |= chunk; /* mark head */ - table++; - *pgtable++ = virt_to_page(tmpb.area); - tmpb.area += PAGE_SIZE; - tmpb.addr += PAGE_SIZE; - } - sgbuf->pages += chunk; - pages -= chunk; - if (chunk < maxpages) - maxpages = chunk; - } - - sgbuf->size = size; - area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, prot); - if (!area) - goto _failed; - return area; - - _failed: - snd_dma_sg_free(dmab); /* free the table */ - return NULL; -} - -static dma_addr_t snd_dma_sg_get_addr(struct snd_dma_buffer *dmab, - size_t offset) -{ - struct snd_sg_buf *sgbuf = dmab->private_data; - dma_addr_t addr; - - addr = sgbuf->table[offset >> PAGE_SHIFT].addr; - addr &= ~((dma_addr_t)PAGE_SIZE - 1); - return addr + offset % PAGE_SIZE; -} - -static struct page *snd_dma_sg_get_page(struct snd_dma_buffer *dmab, - size_t offset) -{ - struct snd_sg_buf *sgbuf = dmab->private_data; - unsigned int idx = offset >> PAGE_SHIFT; - - if (idx >= (unsigned int)sgbuf->pages) - return NULL; - return sgbuf->page_table[idx]; -} - -static unsigned int snd_dma_sg_get_chunk_size(struct snd_dma_buffer *dmab, - unsigned int ofs, - unsigned int size) -{ - struct snd_sg_buf *sg = dmab->private_data; - unsigned int start, end, pg; - - start = ofs >> PAGE_SHIFT; - end = (ofs + size - 1) >> PAGE_SHIFT; - /* check page continuity */ - pg = sg->table[start].addr >> PAGE_SHIFT; - for (;;) { - start++; - if (start > end) - break; - pg++; - if ((sg->table[start].addr >> PAGE_SHIFT) != pg) - return (start << PAGE_SHIFT) - ofs; - } - /* ok, all on continuous pages */ - return size; -} - -static int snd_dma_sg_mmap(struct snd_dma_buffer *dmab, - struct vm_area_struct *area) -{ - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); - return -ENOENT; /* continue with the default mmap handler */ -} - -const struct snd_malloc_ops snd_dma_sg_ops = { - .alloc = snd_dma_sg_alloc, - .free = snd_dma_sg_free, - .get_addr = snd_dma_sg_get_addr, - .get_page = snd_dma_sg_get_page, - .get_chunk_size = snd_dma_sg_get_chunk_size, - .mmap = snd_dma_sg_mmap, -}; -- 2.31.1

1 0

[PATCH] ALSA: hda: Fix potential deadlock at codec unbinding
by Takashi Iwai 16 Nov '21

16 Nov '21

When a codec is unbound dynamically via sysfs while its stream is in use, we may face a potential deadlock at the proc remove or a UAF. This happens since the hda_pcm linked list, as it handles the hda_pcm object release via kref. When a PCM was opened at the unbinding time, the release of hda_pcm gets delayed and it ends up with the close of the PCM stream releasing the associated hda_pcm object of its own. The hda_pcm destructor contains the PCM device release that includes the removal of procfs entries. And, this removal has the sync of the close of all in-use files -- which would never finish because it's called from the PCM file descriptor itself, i.e. it's trying to shoot its foot. For addressing the deadlock above, this patch changes the way to manage and release the hda_pcm object. The kref of hda_pcm is dropped, and instead a simple refcount is introduced in hda_codec for keeping the track of the active PCM streams, and at each PCM open and close, this refcount is adjusted accordingly. At unbinding, the driver calls snd_device_disconnect() for each PCM stream, then synchronizes with the refcount finish, and finally releases the object resources. Fixes: bbbc7e8502c9 ("ALSA: hda - Allocate hda_pcm objects dynamically") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_codec.h | 8 +++++--- sound/pci/hda/hda_bind.c | 5 +++++ sound/pci/hda/hda_codec.c | 42 ++++++++++++++++++++++++--------------- sound/pci/hda/hda_local.h | 1 + 4 files changed, 37 insertions(+), 19 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 0e45963bb767..82d9daa17851 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -8,7 +8,7 @@ #ifndef __SOUND_HDA_CODEC_H #define __SOUND_HDA_CODEC_H -#include <linux/kref.h> +#include <linux/refcount.h> #include <linux/mod_devicetable.h> #include <sound/info.h> #include <sound/control.h> @@ -166,8 +166,8 @@ struct hda_pcm { bool own_chmap; /* codec driver provides own channel maps */ /* private: */ struct hda_codec *codec; - struct kref kref; struct list_head list; + unsigned int disconnected:1; }; /* codec information */ @@ -187,6 +187,8 @@ struct hda_codec { /* PCM to create, set by patch_ops.build_pcms callback */ struct list_head pcm_list_head; + refcount_t pcm_ref; + wait_queue_head_t remove_sleep; /* codec specific info */ void *spec; @@ -420,7 +422,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec); static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm) { - kref_get(&pcm->kref); + refcount_inc(&pcm->codec->pcm_ref); } void snd_hda_codec_pcm_put(struct hda_pcm *pcm); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 1c8bffc3eec6..7153bd53e189 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -156,6 +156,11 @@ static int hda_codec_driver_remove(struct device *dev) return codec->bus->core.ext_ops->hdev_detach(&codec->core); } + refcount_dec(&codec->pcm_ref); + snd_hda_codec_disconnect_pcms(codec); + wait_event(codec->remove_sleep, !refcount_read(&codec->pcm_ref)); + snd_power_sync_ref(codec->bus->card); + if (codec->patch_ops.free) codec->patch_ops.free(codec); snd_hda_codec_cleanup_for_unbind(codec); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index eda70814369b..7016b48227bf 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -703,20 +703,10 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) /* * PCM device */ -static void release_pcm(struct kref *kref) -{ - struct hda_pcm *pcm = container_of(kref, struct hda_pcm, kref); - - if (pcm->pcm) - snd_device_free(pcm->codec->card, pcm->pcm); - clear_bit(pcm->device, pcm->codec->bus->pcm_dev_bits); - kfree(pcm->name); - kfree(pcm); -} - void snd_hda_codec_pcm_put(struct hda_pcm *pcm) { - kref_put(&pcm->kref, release_pcm); + if (refcount_dec_and_test(&pcm->codec->pcm_ref)) + wake_up(&pcm->codec->remove_sleep); } EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_put); @@ -731,7 +721,6 @@ struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, return NULL; pcm->codec = codec; - kref_init(&pcm->kref); va_start(args, fmt); pcm->name = kvasprintf(GFP_KERNEL, fmt, args); va_end(args); @@ -741,6 +730,7 @@ struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec, } list_add_tail(&pcm->list, &codec->pcm_list_head); + refcount_inc(&codec->pcm_ref); return pcm; } EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_new); @@ -748,15 +738,31 @@ EXPORT_SYMBOL_GPL(snd_hda_codec_pcm_new); /* * codec destructor */ +void snd_hda_codec_disconnect_pcms(struct hda_codec *codec) +{ + struct hda_pcm *pcm; + + list_for_each_entry(pcm, &codec->pcm_list_head, list) { + if (pcm->disconnected) + continue; + if (pcm->pcm) + snd_device_disconnect(codec->card, pcm->pcm); + snd_hda_codec_pcm_put(pcm); + pcm->disconnected = 1; + } +} + static void codec_release_pcms(struct hda_codec *codec) { struct hda_pcm *pcm, *n; list_for_each_entry_safe(pcm, n, &codec->pcm_list_head, list) { - list_del_init(&pcm->list); + list_del(&pcm->list); if (pcm->pcm) - snd_device_disconnect(codec->card, pcm->pcm); - snd_hda_codec_pcm_put(pcm); + snd_device_free(pcm->codec->card, pcm->pcm); + clear_bit(pcm->device, pcm->codec->bus->pcm_dev_bits); + kfree(pcm->name); + kfree(pcm); } } @@ -769,6 +775,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) codec->registered = 0; } + snd_hda_codec_disconnect_pcms(codec); cancel_delayed_work_sync(&codec->jackpoll_work); if (!codec->in_freeing) snd_hda_ctls_clear(codec); @@ -792,6 +799,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) remove_conn_list(codec); snd_hdac_regmap_exit(&codec->core); codec->configured = 0; + refcount_set(&codec->pcm_ref, 1); /* reset refcount */ } EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup_for_unbind); @@ -958,6 +966,8 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); INIT_LIST_HEAD(&codec->conn_list); INIT_LIST_HEAD(&codec->pcm_list_head); + refcount_set(&codec->pcm_ref, 1); + init_waitqueue_head(&codec->remove_sleep); INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); codec->depop_delay = -1; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index ea8ab8b43337..4662a47add7e 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -137,6 +137,7 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, int snd_hda_codec_reset(struct hda_codec *codec); void snd_hda_codec_register(struct hda_codec *codec); void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec); +void snd_hda_codec_disconnect_pcms(struct hda_codec *codec); #define snd_hda_regmap_sync(codec) snd_hdac_regmap_sync(&(codec)->core) -- 2.31.1

1 0

[PATCH 1/3] ALSA: jack: Add missing rwsem around snd_ctl_remove() calls
by Takashi Iwai 16 Nov '21

16 Nov '21

snd_ctl_remove() has to be called with card->controls_rwsem held (when called after the card instantiation). This patch add the missing rwsem calls around it. Fixes: 9058cbe1eed2 ("ALSA: jack: implement kctl creating for jack devices") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/jack.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/jack.c b/sound/core/jack.c index 32350c6aba84..f50a1e920e1d 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -62,10 +62,13 @@ static int snd_jack_dev_free(struct snd_device *device) struct snd_card *card = device->card; struct snd_jack_kctl *jack_kctl, *tmp_jack_kctl; + down_write(&card->controls_rwsem); list_for_each_entry_safe(jack_kctl, tmp_jack_kctl, &jack->kctl_list, list) { list_del_init(&jack_kctl->list); snd_ctl_remove(card, jack_kctl->kctl); } + up_write(&card->controls_rwsem); + if (jack->private_free) jack->private_free(jack); -- 2.31.1

1 2

[PATCH 0/3] ALSA: usb-audio: dB mapping fixes
by Takashi Iwai 16 Nov '21

16 Nov '21

Hi, here is a small patch set to slightly extend the dB mapping table in USB-audio driver to add the support for minimal-mute. The feature itself is already present in the driver, and it's a matter of addition in the mapping table. This is used in the quirk for Bose SoundLink. Takashi === Takashi Iwai (3): ALSA: usb-audio: Use int for dB map values ALSA: usb-audio: Add minimal-mute notion in dB mapping table ALSA: usb-audio: Fix dB level of Bose Revolve+ SoundLink sound/usb/mixer.c | 1 + sound/usb/mixer_maps.c | 17 +++++++++++++++-- 2 files changed, 16 insertions(+), 2 deletions(-) -- 2.26.2

1 3

[PATCH] firmware: cs_dsp: tidy includes in cs_dsp.c and cs_dsp.h
by Simon Trimmer 16 Nov '21

16 Nov '21

This patch removes unused included header files and moves others into cs_dsp.h to ensure that types referenced in the header file are properly described to prevent compiler warnings. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- drivers/firmware/cirrus/cs_dsp.c | 6 ------ include/linux/firmware/cirrus/cs_dsp.h | 5 +++++ 2 files changed, 5 insertions(+), 6 deletions(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 948dd8382686..1a0c6c793f6a 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -12,16 +12,10 @@ #include <linux/ctype.h> #include <linux/debugfs.h> #include <linux/delay.h> -#include <linux/device.h> -#include <linux/firmware.h> -#include <linux/interrupt.h> -#include <linux/list.h> #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/regmap.h> #include <linux/slab.h> #include <linux/vmalloc.h> -#include <linux/workqueue.h> #include <linux/firmware/cirrus/cs_dsp.h> #include <linux/firmware/cirrus/wmfw.h> diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h index 9ad9eaaaa552..3a54b1afc48f 100644 --- a/include/linux/firmware/cirrus/cs_dsp.h +++ b/include/linux/firmware/cirrus/cs_dsp.h @@ -11,6 +11,11 @@ #ifndef __CS_DSP_H #define __CS_DSP_H +#include <linux/device.h> +#include <linux/firmware.h> +#include <linux/list.h> +#include <linux/regmap.h> + #define CS_ADSP2_REGION_0 BIT(0) #define CS_ADSP2_REGION_1 BIT(1) #define CS_ADSP2_REGION_2 BIT(2) -- 2.33.0

3 2

[PATCH] ASoC: adau1701: Replace legacy gpio interface for gpiod
by Maíra Canal 16 Nov '21

16 Nov '21

Considering the current transition of the GPIO subsystem, remove all dependencies of the legacy GPIO interface (linux/gpio.h and linux /of_gpio.h) and replace it with the descriptor-based GPIO approach. Signed-off-by: Maíra Canal <maira.canal(a)usp.br> --- sound/soc/codecs/adau1701.c | 94 ++++++++++++------------------------- 1 file changed, 31 insertions(+), 63 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 5ce74697564a..1ef071533b06 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -13,8 +13,8 @@ #include <linux/delay.h> #include <linux/slab.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/of_device.h> +#include <linux/gpio/consumer.h> #include <linux/regulator/consumer.h> #include <linux/regmap.h> #include <sound/core.h> @@ -106,8 +106,8 @@ static const char * const supply_names[] = { }; struct adau1701 { - int gpio_nreset; - int gpio_pll_mode[2]; + struct gpio_desc *gpio_nreset; + struct gpio_descs *gpio_pll_mode; unsigned int dai_fmt; unsigned int pll_clkdiv; unsigned int sysclk; @@ -303,39 +303,41 @@ static int adau1701_reset(struct snd_soc_component *component, unsigned int clkd struct adau1701 *adau1701 = snd_soc_component_get_drvdata(component); int ret; + DECLARE_BITMAP(values, 2); sigmadsp_reset(adau1701->sigmadsp); - if (clkdiv != ADAU1707_CLKDIV_UNSET && - gpio_is_valid(adau1701->gpio_pll_mode[0]) && - gpio_is_valid(adau1701->gpio_pll_mode[1])) { + if (clkdiv != ADAU1707_CLKDIV_UNSET && adau1701->gpio_pll_mode) { switch (clkdiv) { case 64: - gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); - gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); + __assign_bit(0, values, 0); + __assign_bit(1, values, 0); break; case 256: - gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); - gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); + __assign_bit(0, values, 0); + __assign_bit(1, values, 1); break; case 384: - gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); - gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); + __assign_bit(0, values, 1); + __assign_bit(1, values, 0); break; - case 0: /* fallback */ + case 0: /* fallback */ case 512: - gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); - gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); + __assign_bit(0, values, 1); + __assign_bit(1, values, 1); break; } + gpiod_set_array_value_cansleep(adau1701->gpio_pll_mode->ndescs, + adau1701->gpio_pll_mode->desc, adau1701->gpio_pll_mode->info, + values); } adau1701->pll_clkdiv = clkdiv; - if (gpio_is_valid(adau1701->gpio_nreset)) { - gpio_set_value_cansleep(adau1701->gpio_nreset, 0); + if (adau1701->gpio_nreset) { + gpiod_set_value_cansleep(adau1701->gpio_nreset, 0); /* minimum reset time is 20ns */ udelay(1); - gpio_set_value_cansleep(adau1701->gpio_nreset, 1); + gpiod_set_value_cansleep(adau1701->gpio_nreset, 1); /* power-up time may be as long as 85ms */ mdelay(85); } @@ -719,8 +721,8 @@ static void adau1701_remove(struct snd_soc_component *component) { struct adau1701 *adau1701 = snd_soc_component_get_drvdata(component); - if (gpio_is_valid(adau1701->gpio_nreset)) - gpio_set_value_cansleep(adau1701->gpio_nreset, 0); + if (adau1701->gpio_nreset) + gpiod_set_value_cansleep(adau1701->gpio_nreset, 0); regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); } @@ -788,8 +790,6 @@ static int adau1701_i2c_probe(struct i2c_client *client, { struct adau1701 *adau1701; struct device *dev = &client->dev; - int gpio_nreset = -EINVAL; - int gpio_pll_mode[2] = { -EINVAL, -EINVAL }; int ret, i; adau1701 = devm_kzalloc(dev, sizeof(*adau1701), GFP_KERNEL); @@ -823,26 +823,6 @@ static int adau1701_i2c_probe(struct i2c_client *client, if (dev->of_node) { - gpio_nreset = of_get_named_gpio(dev->of_node, "reset-gpio", 0); - if (gpio_nreset < 0 && gpio_nreset != -ENOENT) { - ret = gpio_nreset; - goto exit_regulators_disable; - } - - gpio_pll_mode[0] = of_get_named_gpio(dev->of_node, - "adi,pll-mode-gpios", 0); - if (gpio_pll_mode[0] < 0 && gpio_pll_mode[0] != -ENOENT) { - ret = gpio_pll_mode[0]; - goto exit_regulators_disable; - } - - gpio_pll_mode[1] = of_get_named_gpio(dev->of_node, - "adi,pll-mode-gpios", 1); - if (gpio_pll_mode[1] < 0 && gpio_pll_mode[1] != -ENOENT) { - ret = gpio_pll_mode[1]; - goto exit_regulators_disable; - } - of_property_read_u32(dev->of_node, "adi,pll-clkdiv", &adau1701->pll_clkdiv); @@ -851,32 +831,20 @@ static int adau1701_i2c_probe(struct i2c_client *client, ARRAY_SIZE(adau1701->pin_config)); } - if (gpio_is_valid(gpio_nreset)) { - ret = devm_gpio_request_one(dev, gpio_nreset, GPIOF_OUT_INIT_LOW, - "ADAU1701 Reset"); - if (ret < 0) - goto exit_regulators_disable; + adau1701->gpio_nreset = devm_gpiod_get_optional(dev, "reset", GPIOD_IN); + + if (IS_ERR(adau1701->gpio_nreset)) { + ret = PTR_ERR(adau1701->gpio_nreset); + goto exit_regulators_disable; } - if (gpio_is_valid(gpio_pll_mode[0]) && - gpio_is_valid(gpio_pll_mode[1])) { - ret = devm_gpio_request_one(dev, gpio_pll_mode[0], - GPIOF_OUT_INIT_LOW, - "ADAU1701 PLL mode 0"); - if (ret < 0) - goto exit_regulators_disable; + adau1701->gpio_pll_mode = devm_gpiod_get_array_optional(dev, "adi,pll-mode", GPIOD_OUT_LOW); - ret = devm_gpio_request_one(dev, gpio_pll_mode[1], - GPIOF_OUT_INIT_LOW, - "ADAU1701 PLL mode 1"); - if (ret < 0) - goto exit_regulators_disable; + if (IS_ERR(adau1701->gpio_pll_mode)) { + ret = PTR_ERR(adau1701->gpio_pll_mode); + goto exit_regulators_disable; } - adau1701->gpio_nreset = gpio_nreset; - adau1701->gpio_pll_mode[0] = gpio_pll_mode[0]; - adau1701->gpio_pll_mode[1] = gpio_pll_mode[1]; - i2c_set_clientdata(client, adau1701); adau1701->sigmadsp = devm_sigmadsp_init_i2c(client, -- 2.31.1

2 1

[PATCH] [RESEND,v2] ASoC: Intel: add sof-nau8825 machine driver
by Mac Chiang 16 Nov '21

16 Nov '21

The machine driver is a generic machine driver for SOF with nau8825 codec w or w/o speaker additionally. Depending on the SOC HDMI, DMIC, Bluetooth offload support are added dynamically. Only add information related to SOF since the machine driver was only tested with SOF. There are currently 4 i2s machine variants of ADL. This supports the headphone NUA8825(SSP0) alone or with smart or dumb speakers. Board 2,3,4 use SSP2 for Bluetooth offload support except board 1. Board 1 : NAU8825 + RT1019P(SSP2) Board 2 : NAU8825 + MAX98373(SSP1) Board 3 : NAU8825 + MAX98360A(SSP1) Board 4 : NAU8825 Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> Co-developed-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- 1. remove hdac_hdmi relevant codes support 2. remove duplicated max_98360a_dai_link calls. Instead of checking no_amp condition 3. alter module license name to "GPL" 4. due to rt1019 non-i2c mode, so change to SND_SOC_RT1015P in KConfig --- sound/soc/intel/boards/Kconfig | 19 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/sof_nau8825.c | 651 ++++++++++++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 35 + 4 files changed, 707 insertions(+) create mode 100644 sound/soc/intel/boards/sof_nau8825.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 2dd5ff7e35ce..849445fcc05d 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -527,6 +527,25 @@ config SND_SOC_INTEL_SOF_ES8336_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_SOF_NAU8825_MACH + tristate "SOF with nau8825 codec in I2S Mode" + depends on I2C && ACPI && GPIOLIB + depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ + (MFD_INTEL_LPSS || COMPILE_TEST)) + select SND_SOC_NAU8825 + select SND_SOC_RT1015P + select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_HDA_DSP_COMMON + select SND_SOC_INTEL_SOF_MAXIM_COMMON + help + This adds support for ASoC machine driver for SOF platforms + with nau8825 codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL if (SND_SOC_SOF_COMETLAKE && SND_SOC_SOF_HDA_LINK) diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 9ee8ed864f5d..3ea273d27168 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -22,6 +22,7 @@ snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-sof_rt5682-objs := sof_rt5682.o sof_realtek_common.o snd-soc-sof_cs42l42-objs := sof_cs42l42.o snd-soc-sof_es8336-objs := sof_es8336.o +snd-soc-sof_nau8825-objs := sof_nau8825.o sof_realtek_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o @@ -44,6 +45,7 @@ snd-soc-sof-sdw-objs += sof_sdw.o \ obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH) += snd-soc-sof_cs42l42.o obj-$(CONFIG_SND_SOC_INTEL_SOF_ES8336_MACH) += snd-soc-sof_es8336.o +obj-$(CONFIG_SND_SOC_INTEL_SOF_NAU8825_MACH) += snd-soc-sof_nau8825.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c new file mode 100644 index 000000000000..33de043b66c6 --- /dev/null +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -0,0 +1,651 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright(c) 2021 Intel Corporation. +// Copyright(c) 2021 Nuvoton Corporation. + +/* + * Intel SOF Machine Driver with Nuvoton headphone codec NAU8825 + * and speaker codec RT1019P MAX98360a or MAX98373 + */ +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/dmi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/sof.h> +#include <sound/soc-acpi.h> +#include "../../codecs/nau8825.h" +#include "../common/soc-intel-quirks.h" +#include "hda_dsp_common.h" +#include "sof_realtek_common.h" +#include "sof_maxim_common.h" + +#define NAME_SIZE 32 + +#define SOF_NAU8825_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) +#define SOF_NAU8825_SSP_CODEC_MASK (GENMASK(2, 0)) +#define SOF_SPEAKER_AMP_PRESENT BIT(3) +#define SOF_NAU8825_SSP_AMP_SHIFT 4 +#define SOF_NAU8825_SSP_AMP_MASK (GENMASK(6, 4)) +#define SOF_NAU8825_SSP_AMP(quirk) \ + (((quirk) << SOF_NAU8825_SSP_AMP_SHIFT) & SOF_NAU8825_SSP_AMP_MASK) +#define SOF_NAU8825_NUM_HDMIDEV_SHIFT 7 +#define SOF_NAU8825_NUM_HDMIDEV_MASK (GENMASK(9, 7)) +#define SOF_NAU8825_NUM_HDMIDEV(quirk) \ + (((quirk) << SOF_NAU8825_NUM_HDMIDEV_SHIFT) & SOF_NAU8825_NUM_HDMIDEV_MASK) + +/* BT audio offload: reserve 3 bits for future */ +#define SOF_BT_OFFLOAD_SSP_SHIFT 10 +#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(12, 10)) +#define SOF_BT_OFFLOAD_SSP(quirk) \ + (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) +#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(13) +#define SOF_RT1019P_SPEAKER_AMP_PRESENT BIT(14) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) + +static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0); + +struct sof_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct sof_card_private { + struct clk *mclk; + struct snd_soc_jack sof_headset; + struct list_head hdmi_pcm_list; +}; + +static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); + struct sof_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + /* dai_link id is 1:1 mapped to the PCM device */ + pcm->device = rtd->dai_link->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int sof_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + + struct snd_soc_jack *jack; + int ret; + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->sof_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static void sof_nau8825_codec_exit(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + + snd_soc_component_set_jack(component, NULL, NULL); +} + +static int sof_nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + int clk_freq, ret; + + clk_freq = sof_dai_get_bclk(rtd); /* BCLK freq */ + + if (clk_freq <= 0) { + dev_err(rtd->dev, "get bclk freq failed: %d\n", clk_freq); + return -EINVAL; + } + + /* Configure clock for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_BLK, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set BCLK clock %d\n", ret); + return ret; + } + + /* Configure pll for codec */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, clk_freq, + params_rate(params) * 256); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set BCLK: %d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops sof_nau8825_ops = { + .hw_params = sof_nau8825_hw_params, +}; + +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + +static int sof_card_late_probe(struct snd_soc_card *card) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_dapm_context *dapm = &card->dapm; + 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"); + snd_soc_dapm_disable_pin(dapm, "Right Spk"); + err = snd_soc_dapm_sync(dapm); + if (err < 0) + return err; + } + + return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); +} + +static const struct snd_kcontrol_new sof_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_kcontrol_new speaker_controls[] = { + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget sof_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_widget speaker_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_widget dmic_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route sof_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* other jacks */ + { "MIC", NULL, "Headset Mic" }, +}; + +static const struct snd_soc_dapm_route speaker_map[] = { + /* speaker */ + { "Spk", NULL, "Speaker" }, +}; + +static const struct snd_soc_dapm_route dmic_map[] = { + /* digital mics */ + {"DMic", NULL, "SoC DMIC"}, +}; + +static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, speaker_widgets, + ARRAY_SIZE(speaker_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, speaker_controls, + ARRAY_SIZE(speaker_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map, + ARRAY_SIZE(speaker_map)); + + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +static int dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets, + ARRAY_SIZE(dmic_widgets)); + if (ret) { + dev_err(card->dev, "DMic widget addition failed: %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map, + ARRAY_SIZE(dmic_map)); + + if (ret) + dev_err(card->dev, "DMic map addition failed: %d\n", ret); + + return ret; +} + +/* sof audio machine driver for nau8825 codec */ +static struct snd_soc_card sof_audio_card_nau8825 = { + .name = "nau8825", /* the sof- prefix is added by the core */ + .owner = THIS_MODULE, + .controls = sof_controls, + .num_controls = ARRAY_SIZE(sof_controls), + .dapm_widgets = sof_widgets, + .num_dapm_widgets = ARRAY_SIZE(sof_widgets), + .dapm_routes = sof_map, + .num_dapm_routes = ARRAY_SIZE(sof_map), + .fully_routed = true, + .late_probe = sof_card_late_probe, +}; + +static struct snd_soc_dai_link_component nau8825_component[] = { + { + .name = "i2c-10508825:00", + .dai_name = "nau8825-hifi", + } +}; + +static struct snd_soc_dai_link_component dmic_component[] = { + { + .name = "dmic-codec", + .dai_name = "dmic-hifi", + } +}; + +static struct snd_soc_dai_link_component rt1019p_component[] = { + { + .name = "RTL1019:00", + .dai_name = "HiFi", + } +}; + +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + +static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, + int ssp_codec, + int ssp_amp, + int dmic_be_num, + int hdmi_num) +{ + struct snd_soc_dai_link_component *idisp_components; + struct snd_soc_dai_link_component *cpus; + struct snd_soc_dai_link *links; + int i, id = 0; + + links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * + sof_audio_card_nau8825.num_links, GFP_KERNEL); + cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component) * + sof_audio_card_nau8825.num_links, GFP_KERNEL); + if (!links || !cpus) + goto devm_err; + + /* codec SSP */ + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_codec); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].codecs = nau8825_component; + links[id].num_codecs = ARRAY_SIZE(nau8825_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_nau8825_codec_init; + links[id].exit = sof_nau8825_codec_exit; + links[id].ops = &sof_nau8825_ops; + links[id].dpcm_playback = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_codec); + if (!links[id].cpus->dai_name) + goto devm_err; + + id++; + + /* dmic */ + if (dmic_be_num > 0) { + /* at least we have dmic01 */ + links[id].name = "dmic01"; + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = "DMIC01 Pin"; + links[id].init = dmic_init; + if (dmic_be_num > 1) { + /* set up 2 BE links at most */ + links[id + 1].name = "dmic16k"; + links[id + 1].cpus = &cpus[id + 1]; + links[id + 1].cpus->dai_name = "DMIC16k Pin"; + dmic_be_num = 2; + } + } + + for (i = 0; i < dmic_be_num; i++) { + links[id].id = id; + links[id].num_cpus = 1; + links[id].codecs = dmic_component; + links[id].num_codecs = ARRAY_SIZE(dmic_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].ignore_suspend = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + id++; + } + + /* HDMI */ + if (hdmi_num > 0) { + idisp_components = devm_kzalloc(dev, + sizeof(struct snd_soc_dai_link_component) * + hdmi_num, GFP_KERNEL); + if (!idisp_components) + goto devm_err; + } + for (i = 1; i <= hdmi_num; i++) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d", i); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d Pin", i); + if (!links[id].cpus->dai_name) + goto devm_err; + + idisp_components[i - 1].name = "ehdaudio0D2"; + idisp_components[i - 1].dai_name = devm_kasprintf(dev, + GFP_KERNEL, + "intel-hdmi-hifi%d", + i); + if (!idisp_components[i - 1].dai_name) + goto devm_err; + + links[id].codecs = &idisp_components[i - 1]; + links[id].num_codecs = 1; + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_hdmi_init; + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + id++; + } + + /* speaker amp */ + if (sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_amp); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + if (sof_nau8825_quirk & SOF_RT1019P_SPEAKER_AMP_PRESENT) { + links[id].codecs = rt1019p_component; + links[id].num_codecs = ARRAY_SIZE(rt1019p_component); + links[id].init = speaker_codec_init; + } else if (sof_nau8825_quirk & + SOF_MAX98373_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98373_components; + links[id].num_codecs = ARRAY_SIZE(max_98373_components); + links[id].init = max_98373_spk_codec_init; + links[id].ops = &max_98373_ops; + /* feedback stream */ + links[id].dpcm_capture = 1; + } else if (sof_nau8825_quirk & + SOF_MAX98360A_SPEAKER_AMP_PRESENT) { + max_98360a_dai_link(&links[id]); + } else { + goto devm_err; + } + + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_amp); + if (!links[id].cpus->dai_name) + goto devm_err; + id++; + } + + /* BT audio offload */ + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { + int port = (sof_nau8825_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> + SOF_BT_OFFLOAD_SSP_SHIFT; + + links[id].id = id; + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", port); + if (!links[id].cpus->dai_name) + goto devm_err; + links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); + if (!links[id].name) + goto devm_err; + links[id].codecs = dummy_component; + links[id].num_codecs = ARRAY_SIZE(dummy_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].dpcm_playback = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + links[id].num_cpus = 1; + } + + return links; +devm_err: + return NULL; +} + +static int sof_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *dai_links; + struct snd_soc_acpi_mach *mach; + struct sof_card_private *ctx; + int dmic_be_num, hdmi_num; + int ret, ssp_amp, ssp_codec; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + if (pdev->id_entry && pdev->id_entry->driver_data) + sof_nau8825_quirk = (unsigned long)pdev->id_entry->driver_data; + + mach = pdev->dev.platform_data; + + /* A speaker amp might not be present when the quirk claims one is. + * Detect this via whether the machine driver match includes quirk_data. + */ + if ((sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) + sof_nau8825_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + + dev_dbg(&pdev->dev, "sof_nau8825_quirk = %lx\n", sof_nau8825_quirk); + + /* default number of DMIC DAI's */ + dmic_be_num = 2; + hdmi_num = (sof_nau8825_quirk & SOF_NAU8825_NUM_HDMIDEV_MASK) >> + SOF_NAU8825_NUM_HDMIDEV_SHIFT; + /* default number of HDMI DAI's */ + if (!hdmi_num) + hdmi_num = 3; + + ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >> + SOF_NAU8825_SSP_AMP_SHIFT; + + ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK; + + /* compute number of dai links */ + sof_audio_card_nau8825.num_links = 1 + dmic_be_num + hdmi_num; + + if (sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) + sof_audio_card_nau8825.num_links++; + + if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) + max_98373_set_codec_conf(&sof_audio_card_nau8825); + + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) + sof_audio_card_nau8825.num_links++; + + dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, + dmic_be_num, hdmi_num); + if (!dai_links) + return -ENOMEM; + + sof_audio_card_nau8825.dai_link = dai_links; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + sof_audio_card_nau8825.dev = &pdev->dev; + + /* set platform name for each dailink */ + ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_nau8825, + mach->mach_params.platform); + if (ret) + return ret; + + snd_soc_card_set_drvdata(&sof_audio_card_nau8825, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &sof_audio_card_nau8825); +} + +static const struct platform_device_id board_ids[] = { + { + .name = "sof_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, + { + .name = "adl_rt1019p_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1019P_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(2) | + SOF_NAU8825_NUM_HDMIDEV(4)), + }, + { + .name = "adl_max98373_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98373_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { + /* The limitation of length of char array, shorten the name */ + .name = "adl_mx98360a_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, + { } +}; +MODULE_DEVICE_TABLE(platform, board_ids); + +static struct platform_driver sof_audio = { + .probe = sof_audio_probe, + .driver = { + .name = "sof_nau8825", + .pm = &snd_soc_pm_ops, + }, + .id_table = board_ids, +}; +module_platform_driver(sof_audio) + +/* Module information */ +MODULE_DESCRIPTION("SOF Audio Machine driver for NAU8825"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 06f503452aa5..4a3955473271 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -290,6 +290,11 @@ static const struct snd_soc_acpi_codecs adl_rt5682_rt5682s_hp = { .codecs = {"10EC5682", "RTL5682"}, }; +static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { + .num_codecs = 1, + .codecs = {"RTL1019"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .comp_ids = &adl_rt5682_rt5682s_hp, @@ -315,6 +320,36 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", }, + { + .id = "10508825", + .drv_name = "adl_rt1019p_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_rt1019p_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "adl_max98373_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98373_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98373-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "adl_mx98360a_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-mx98360a-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "sof_nau8825", + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-nau8825.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.20.1

2 1

[PATCH 0/3] ASoC: cs42l42: Fix definition and handling of jack switch invert
by Richard Fitzgerald 16 Nov '21

16 Nov '21

Summary: The driver applied the opposite of the DT setting to the wrong register bit. The jack plug detect hardware in cs42l42 is somewhat confusing, compounded by an unclear description in the datasheet. This is most likely the reason that the driver implemented a DT property for the wrong register bit, that had the opposite effect of what was described in the binding. Changing the meaning of the property values isn't feasible; the driver dates from 2016 and the risk of breaking out-of-tree configs is too high (the property is also available to ACPI systems). So the fix is to make the binding doc match the actual behaviour and then fix the driver to apply it to the correct register bit. As a bonus, patch #3 converts the binding to yaml. Richard Fitzgerald (3): ASoC: dt-bindings: cs42l42: Correct description of ts-inv ASoC: cs42l42: Correct configuring of switch inversion from ts-inv ASoC: dt-bindings: cs42l42: Convert binding to yaml .../devicetree/bindings/sound/cirrus,cs42l42.yaml | 225 +++++++++++++++++++++ .../devicetree/bindings/sound/cs42l42.txt | 114 ----------- MAINTAINERS | 1 + sound/soc/codecs/cs42l42.c | 9 +- 4 files changed, 230 insertions(+), 119 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l42.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs42l42.txt -- 2.11.0

3 6

[PATCH v2] dt-bindings: sound: wlf, wm8903: Convert txt bindings to yaml
by David Heidelberg 16 Nov '21

16 Nov '21

Convert the Wolfson WM8903 Ultra-Low Power Stereo CODEC Device Tree binding documentation to json-schema. Signed-off-by: David Heidelberg <david(a)ixit.cz> --- .../devicetree/bindings/sound/wlf,wm8903.yaml | 116 ++++++++++++++++++ .../devicetree/bindings/sound/wm8903.txt | 82 ------------- 2 files changed, 116 insertions(+), 82 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8903.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8903.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml new file mode 100644 index 000000000000..7105ed5fd6c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8903.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/wlf,wm8903.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: WM8903 audio codec + +description: | + This device supports I2C only. + Pins on the device (for linking into audio routes): + * IN1L + * IN1R + * IN2L + * IN2R + * IN3L + * IN3R + * DMICDAT + * HPOUTL + * HPOUTR + * LINEOUTL + * LINEOUTR + * LOP + * LON + * ROP + * RON + * MICBIAS + +maintainers: + - patches(a)opensource.cirrus.com + +properties: + compatible: + const: wlf,wm8903 + + reg: + maxItems: 1 + + gpio-controller: true + '#gpio-cells': + const: 2 + + interrupts: + maxItems: 1 + + micdet-cfg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 0 + description: Default register value for R6 (Mic Bias). + + micdet-delay: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 100 + description: The debounce delay for microphone detection in mS. + + gpio-cfg: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + minItems: 5 + maxItems: 5 + A list of GPIO configuration register values. + If absent, no configuration of these registers is performed. + If any entry has the value 0xffffffff, that GPIO's + configuration will not be modified. + + AVDD-supply: + description: Analog power supply regulator on the AVDD pin. + + CPVDD-supply: + description: Charge pump supply regulator on the CPVDD pin. + + DBVDD-supply: + description: Digital buffer supply regulator for the DBVDD pin. + + DCVDD-supply: + description: Digital core supply regulator for the DCVDD pin. + + +required: + - compatible + - reg + - gpio-controller + - '#gpio-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + wm8903: codec@1a { + compatible = "wlf,wm8903"; + reg = <0x1a>; + interrupts = <347>; + + AVDD-supply = <&fooreg_a>; + CPVDD-supply = <&fooreg_b>; + DBVDD-supply = <&fooreg_c>; + DCVDD-supply = <&fooreg_d>; + + gpio-controller; + #gpio-cells = <2>; + + micdet-cfg = <0>; + micdet-delay = <100>; + gpio-cfg = < + 0x0600 /* DMIC_LR, output */ + 0x0680 /* DMIC_DAT, input */ + 0x0000 /* GPIO, output, low */ + 0x0200 /* Interrupt, output */ + 0x01a0 /* BCLK, input, active high */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8903.txt b/Documentation/devicetree/bindings/sound/wm8903.txt deleted file mode 100644 index 6371c2434afe..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8903.txt +++ /dev/null @@ -1,82 +0,0 @@ -WM8903 audio CODEC - -This device supports I2C only. - -Required properties: - - - compatible : "wlf,wm8903" - - - reg : the I2C address of the device. - - - gpio-controller : Indicates this device is a GPIO controller. - - - #gpio-cells : Should be two. The first cell is the pin number and the - second cell is used to specify optional parameters (currently unused). - -Optional properties: - - - interrupts : The interrupt line the codec is connected to. - - - micdet-cfg : Default register value for R6 (Mic Bias). If absent, the - default is 0. - - - micdet-delay : The debounce delay for microphone detection in mS. If - absent, the default is 100. - - - gpio-cfg : A list of GPIO configuration register values. The list must - be 5 entries long. If absent, no configuration of these registers is - performed. If any entry has the value 0xffffffff, that GPIO's - configuration will not be modified. - - - AVDD-supply : Analog power supply regulator on the AVDD pin. - - - CPVDD-supply : Charge pump supply regulator on the CPVDD pin. - - - DBVDD-supply : Digital buffer supply regulator for the DBVDD pin. - - - DCVDD-supply : Digital core supply regulator for the DCVDD pin. - -Pins on the device (for linking into audio routes): - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * DMICDAT - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * LOP - * LON - * ROP - * RON - * MICBIAS - -Example: - -wm8903: codec@1a { - compatible = "wlf,wm8903"; - reg = <0x1a>; - interrupts = < 347 >; - - AVDD-supply = <&fooreg_a>; - CPVDD-supply = <&fooreg_b>; - DBVDD-supply = <&fooreg_c>; - DCVDC-supply = <&fooreg_d>; - - gpio-controller; - #gpio-cells = <2>; - - micdet-cfg = <0>; - micdet-delay = <100>; - gpio-cfg = < - 0x0600 /* DMIC_LR, output */ - 0x0680 /* DMIC_DAT, input */ - 0x0000 /* GPIO, output, low */ - 0x0200 /* Interrupt, output */ - 0x01a0 /* BCLK, input, active high */ - >; -}; -- 2.33.0

4 4

[PATCH] ASoC: codecs: Axe some dead code in 'wcd_mbhc_adc_hs_rem_irq()'
by Christophe JAILLET 16 Nov '21

16 Nov '21

'hphpa_on' is know to be false, so this is just dead code that should be removed. Suggested-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- This patch is a follow-up of: https://lore.kernel.org/kernel-janitors/988948f7f266aa00698704687537335b7e6… --- sound/soc/codecs/wcd-mbhc-v2.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index 405128ccb4b0..b905eb8f3c67 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -1176,7 +1176,6 @@ static irqreturn_t wcd_mbhc_adc_hs_rem_irq(int irq, void *data) struct wcd_mbhc *mbhc = data; unsigned long timeout; int adc_threshold, output_mv, retry = 0; - bool hphpa_on = false; mutex_lock(&mbhc->lock); timeout = jiffies + msecs_to_jiffies(WCD_FAKE_REMOVAL_MIN_PERIOD_MS); @@ -1210,10 +1209,6 @@ static irqreturn_t wcd_mbhc_adc_hs_rem_irq(int irq, void *data) wcd_mbhc_elec_hs_report_unplug(mbhc); wcd_mbhc_write_field(mbhc, WCD_MBHC_BTN_ISRC_CTL, 0); - if (hphpa_on) { - hphpa_on = false; - wcd_mbhc_write_field(mbhc, WCD_MBHC_HPH_PA_EN, 3); - } exit: mutex_unlock(&mbhc->lock); return IRQ_HANDLED; -- 2.30.2

3 2

[PATCH] ASoC: codecs: MBHC: Add support for special headset
by Srinivasa Rao Mandadapu 16 Nov '21

16 Nov '21

Update MBHC driver to support special headset such as apple and huwawei headsets. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Co-developed-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> Signed-off-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> --- sound/soc/codecs/wcd-mbhc-v2.c | 78 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 74 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index 405128c..3b4cd39 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -1022,6 +1022,57 @@ static int wcd_mbhc_get_plug_from_adc(struct wcd_mbhc *mbhc, int adc_result) return plug_type; } +static int wcd_mbhc_get_spl_hs_thres(struct wcd_mbhc *mbhc) +{ + int hs_threshold, micbias_mv; + + micbias_mv = wcd_mbhc_get_micbias(mbhc); + if (mbhc->cfg->hs_thr && mbhc->cfg->micb_mv != WCD_MBHC_ADC_MICBIAS_MV) { + if (mbhc->cfg->micb_mv == micbias_mv) + hs_threshold = mbhc->cfg->hs_thr; + else + hs_threshold = (mbhc->cfg->hs_thr * micbias_mv) / + mbhc->cfg->micb_mv; + } else { + hs_threshold = ((WCD_MBHC_ADC_HS_THRESHOLD_MV * micbias_mv) / + WCD_MBHC_ADC_MICBIAS_MV); + } + return hs_threshold; +} + +static bool wcd_mbhc_check_for_spl_headset(struct wcd_mbhc *mbhc) +{ + bool is_spl_hs = false; + int output_mv, hs_threshold, hph_threshold; + + if (!mbhc->mbhc_cb->mbhc_micb_ctrl_thr_mic) + return false; + + /* Bump up MIC_BIAS2 to 2.7V */ + mbhc->mbhc_cb->mbhc_micb_ctrl_thr_mic(mbhc->component, MIC_BIAS_2, true); + usleep_range(10000, 10100); + + output_mv = wcd_measure_adc_once(mbhc, MUX_CTL_IN2P); + hs_threshold = wcd_mbhc_get_spl_hs_thres(mbhc); + hph_threshold = wcd_mbhc_adc_get_hph_thres(mbhc); + + if (output_mv > hs_threshold || output_mv < hph_threshold) { + if (mbhc->force_linein == true) + is_spl_hs = false; + } else { + is_spl_hs = true; + } + + /* Back MIC_BIAS2 to 1.8v if the type is not special headset */ + if (!is_spl_hs) { + mbhc->mbhc_cb->mbhc_micb_ctrl_thr_mic(mbhc->component, MIC_BIAS_2, false); + /* Add 10ms delay for micbias to settle */ + usleep_range(10000, 10100); + } + + return is_spl_hs; +} + static void wcd_correct_swch_plug(struct work_struct *work) { struct wcd_mbhc *mbhc; @@ -1029,12 +1080,14 @@ static void wcd_correct_swch_plug(struct work_struct *work) enum wcd_mbhc_plug_type plug_type = MBHC_PLUG_TYPE_INVALID; unsigned long timeout; int pt_gnd_mic_swap_cnt = 0; - int output_mv, cross_conn, hs_threshold, try = 0; + int output_mv, cross_conn, hs_threshold, try = 0, micbias_mv; + bool is_spl_hs = false; bool is_pa_on; mbhc = container_of(work, struct wcd_mbhc, correct_plug_swch); component = mbhc->component; + micbias_mv = wcd_mbhc_get_micbias(mbhc); hs_threshold = wcd_mbhc_adc_get_hs_thres(mbhc); /* Mask ADC COMPLETE interrupt */ @@ -1097,6 +1150,18 @@ static void wcd_correct_swch_plug(struct work_struct *work) plug_type = wcd_mbhc_get_plug_from_adc(mbhc, output_mv); is_pa_on = wcd_mbhc_read_field(mbhc, WCD_MBHC_HPH_PA_EN); + + if ((output_mv > hs_threshold) && + (!is_spl_hs)) { + is_spl_hs = wcd_mbhc_check_for_spl_headset(mbhc); + output_mv = wcd_measure_adc_once(mbhc, MUX_CTL_IN2P); + + if (is_spl_hs) { + hs_threshold = (hs_threshold * wcd_mbhc_get_micbias(mbhc)) / + micbias_mv; + } + } + if ((output_mv <= hs_threshold) && !is_pa_on) { /* Check for cross connection*/ cross_conn = wcd_check_cross_conn(mbhc); @@ -1122,14 +1187,19 @@ static void wcd_correct_swch_plug(struct work_struct *work) } } - if (output_mv > hs_threshold) /* cable is extension cable */ + /* cable is extension cable */ + if (output_mv > hs_threshold || mbhc->force_linein == true) plug_type = MBHC_PLUG_TYPE_HIGH_HPH; } wcd_mbhc_bcs_enable(mbhc, plug_type, true); - if (plug_type == MBHC_PLUG_TYPE_HIGH_HPH) - wcd_mbhc_write_field(mbhc, WCD_MBHC_ELECT_ISRC_EN, 1); + if (plug_type == MBHC_PLUG_TYPE_HIGH_HPH) { + if (is_spl_hs) + plug_type = MBHC_PLUG_TYPE_HEADSET; + else + wcd_mbhc_write_field(mbhc, WCD_MBHC_ELECT_ISRC_EN, 1); + } wcd_mbhc_write_field(mbhc, WCD_MBHC_ADC_MODE, 0); wcd_mbhc_write_field(mbhc, WCD_MBHC_ADC_EN, 0); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 2

[PATCH 1/2] ASoC: cs35l41: Change monitor widgets to siggens
by David Rhodes 16 Nov '21

16 Nov '21

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Currently the internal monitor sources are input widgets, which means if the card is set to fully routed these will not enable unless connected to something in the machine driver. However, all these are internal monitor signals so it makes no sense to connect them to something in the machine driver. As such switch them to siggen widgets which will have the same behaviour except not require external linkage on a fully routed card. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: David Rhodes <drhodes(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l41.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 94ed21d7676f..9d0530dde996 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -612,6 +612,12 @@ static const struct snd_soc_dapm_widget cs35l41_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("ASPTX3", NULL, 0, CS35L41_SP_ENABLES, 2, 0), SND_SOC_DAPM_AIF_OUT("ASPTX4", NULL, 0, CS35L41_SP_ENABLES, 3, 0), + SND_SOC_DAPM_SIGGEN("VSENSE"), + SND_SOC_DAPM_SIGGEN("ISENSE"), + SND_SOC_DAPM_SIGGEN("VP"), + SND_SOC_DAPM_SIGGEN("VBST"), + SND_SOC_DAPM_SIGGEN("TEMP"), + SND_SOC_DAPM_ADC("VMON ADC", NULL, CS35L41_PWR_CTRL2, 12, 0), SND_SOC_DAPM_ADC("IMON ADC", NULL, CS35L41_PWR_CTRL2, 13, 0), SND_SOC_DAPM_ADC("VPMON ADC", NULL, CS35L41_PWR_CTRL2, 8, 0), @@ -623,12 +629,6 @@ static const struct snd_soc_dapm_widget cs35l41_dapm_widgets[] = { cs35l41_main_amp_event, SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_INPUT("VP"), - SND_SOC_DAPM_INPUT("VBST"), - SND_SOC_DAPM_INPUT("ISENSE"), - SND_SOC_DAPM_INPUT("VSENSE"), - SND_SOC_DAPM_INPUT("TEMP"), - SND_SOC_DAPM_MUX("ASP TX1 Source", SND_SOC_NOPM, 0, 0, &asp_tx1_mux), SND_SOC_DAPM_MUX("ASP TX2 Source", SND_SOC_NOPM, 0, 0, &asp_tx2_mux), SND_SOC_DAPM_MUX("ASP TX3 Source", SND_SOC_NOPM, 0, 0, &asp_tx3_mux), @@ -674,8 +674,8 @@ static const struct snd_soc_dapm_route cs35l41_audio_map[] = { {"VMON ADC", NULL, "VSENSE"}, {"IMON ADC", NULL, "ISENSE"}, {"VPMON ADC", NULL, "VP"}, - {"TEMPMON ADC", NULL, "TEMP"}, {"VBSTMON ADC", NULL, "VBST"}, + {"TEMPMON ADC", NULL, "TEMP"}, {"ASPRX1", NULL, "AMP Playback"}, {"ASPRX2", NULL, "AMP Playback"}, -- 2.25.1

3 3

[PATCH] ASoC: SOF: core: Unregister machine driver before IPC and debugfs
by Peter Ujfalusi 16 Nov '21

16 Nov '21

To ensure clean unload of the machine driver, components and topology, do the unregister before we free IPC and debugfs. It is a possibility that part of the unregister we would have IPC communication with the firmware. Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> 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/core.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 2c3de295f11f..af5bbc861297 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -362,6 +362,13 @@ int snd_sof_device_remove(struct device *dev) if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); + /* + * Unregister machine driver. This will unbind the snd_card which + * will remove the component driver and unload the topology + * before freeing the snd_card. + */ + snd_sof_machine_unregister(sdev, pdata); + if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) { ret = snd_sof_dsp_power_down_notify(sdev); if (ret < 0) @@ -373,13 +380,6 @@ int snd_sof_device_remove(struct device *dev) snd_sof_free_trace(sdev); } - /* - * Unregister machine driver. This will unbind the snd_card which - * will remove the component driver and unload the topology - * before freeing the snd_card. - */ - snd_sof_machine_unregister(sdev, pdata); - /* * Unregistering the machine driver results in unloading the topology. * Some widgets, ex: scheduler, attempt to power down the core they are -- 2.33.1

2 1

[PATCH v2] ASoC: cs42l42: Add control for audio slow-start switch
by Richard Fitzgerald 16 Nov '21

16 Nov '21

This adds an ALSA control so that the slow-start audio ramp feature can be disabled. This is useful for high-definition audio applications. The register field is unusual in that it is a 3-bit field with only two valid values, 000=off and 111=on. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 30 +++++++++++++++++++++++++++++- sound/soc/codecs/cs42l42.h | 3 +++ 2 files changed, 32 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 27a1c4c73074..56804a3f285e 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -42,6 +42,7 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_SRC_CTL, 0x10 }, { CS42L42_MCLK_CTL, 0x02 }, { CS42L42_SFTRAMP_RATE, 0xA4 }, + { CS42L42_SLOW_START_ENABLE, 0x70 }, { CS42L42_I2C_DEBOUNCE, 0x88 }, { CS42L42_I2C_STRETCH, 0x03 }, { CS42L42_I2C_TIMEOUT, 0xB7 }, @@ -177,6 +178,7 @@ static bool cs42l42_readable_register(struct device *dev, unsigned int reg) case CS42L42_MCLK_STATUS: case CS42L42_MCLK_CTL: case CS42L42_SFTRAMP_RATE: + case CS42L42_SLOW_START_ENABLE: case CS42L42_I2C_DEBOUNCE: case CS42L42_I2C_STRETCH: case CS42L42_I2C_TIMEOUT: @@ -387,6 +389,28 @@ static const struct regmap_config cs42l42_regmap = { static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true); static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); +static int cs42l42_slow_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + u8 val; + + /* all bits of SLOW_START_EN much change together */ + switch (ucontrol->value.integer.value[0]) { + case 0: + val = 0; + break; + case 1: + val = CS42L42_SLOW_START_EN_MASK; + break; + default: + return -EINVAL; + } + + return snd_soc_component_update_bits(component, CS42L42_SLOW_START_ENABLE, + CS42L42_SLOW_START_EN_MASK, val); +} + static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" }; @@ -431,7 +455,11 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DAC_HPF_EN_SHIFT, true, false), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, - 0x3f, 1, mixer_tlv) + 0x3f, 1, mixer_tlv), + + SOC_SINGLE_EXT("Slow Start Switch", CS42L42_SLOW_START_ENABLE, + CS42L42_SLOW_START_EN_SHIFT, true, false, + snd_soc_get_volsw, cs42l42_slow_start_put), }; static int cs42l42_hp_adc_ev(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index f45bcc9a3a62..c8b3267a318b 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -62,6 +62,9 @@ #define CS42L42_INTERNAL_FS_MASK (1 << CS42L42_INTERNAL_FS_SHIFT) #define CS42L42_SFTRAMP_RATE (CS42L42_PAGE_10 + 0x0A) +#define CS42L42_SLOW_START_ENABLE (CS42L42_PAGE_10 + 0x0B) +#define CS42L42_SLOW_START_EN_MASK GENMASK(6, 4) +#define CS42L42_SLOW_START_EN_SHIFT 4 #define CS42L42_I2C_DEBOUNCE (CS42L42_PAGE_10 + 0x0E) #define CS42L42_I2C_STRETCH (CS42L42_PAGE_10 + 0x0F) #define CS42L42_I2C_TIMEOUT (CS42L42_PAGE_10 + 0x10) -- 2.11.0

2 1

[PATCH] ASoC: wm_adsp: wm_adsp_control_add() error: uninitialized symbol 'ret'
by Simon Trimmer 16 Nov '21

16 Nov '21

This patch fixes the static analysis warning as it is correctly indicating a possible code path, it cannot know that for the affected firmware versions subname would always be NULL. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d4f0d72cbcc8..6cb01a8e08fb 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -617,8 +617,9 @@ static int wm_adsp_control_add(struct cs_dsp_coeff_ctl *cs_ctl) switch (cs_dsp->fw_ver) { case 0: case 1: - snprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "%s %s %x", - cs_dsp->name, region_name, cs_ctl->alg_region.alg); + ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + "%s %s %x", cs_dsp->name, region_name, + cs_ctl->alg_region.alg); break; case 2: ret = scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, -- 2.33.0

3 2

[PATCH 0/2] Add tfa9897 rcv-gpios support
by Vincent Knecht 15 Nov '21

15 Nov '21

This is the continuation of a previous series [1] where - patch 1/4 is removed in favor of using pin switch This will be posted independently of tfa989x support, since it mainly require changes to sound/soc/qcom/common.c and device DTS. - patch 2/4 is already merged so here are reworked patch 3/4 (bindings fixed and example added) and patch 4/4 unchanged. [1] https://patchwork.kernel.org/project/alsa-devel/cover/20211024085840.153643… Vincent Knecht (2): ASoC: dt-bindings: nxp, tfa989x: Add rcv-gpios property for tfa9897 ASoC: codecs: tfa989x: Add support for tfa9897 optional rcv-gpios .../bindings/sound/nxp,tfa989x.yaml | 41 +++++++++++++++++++ sound/soc/codecs/tfa989x.c | 20 ++++++++- 2 files changed, 60 insertions(+), 1 deletion(-) -- 2.31.1

3 4

[PATCH 00/10] ASoC: Intel: sof_sdw: Use fixed DAI link id
by Bard Liao 15 Nov '21

15 Nov '21

This series provides a way to use constant dailink numbers for different devices. So that we don't need to renumber them in topologies. Some patches with different purpose are sent together in this series since they are dependent. Bard Liao (8): ASoC: intel: sof_sdw: return the original error number ASoC: intel: sof_sdw: rename be_index/link_id to link_index ASoC: intel: sof_sdw: Use a fixed DAI link id for AMP ASoC: intel: sof_sdw: move DMIC link id overwrite to create_sdw_dailink ASoC: intel: sof_sdw: remove SOF_RT715_DAI_ID_FIX quirk ASoC: intel: sof_sdw: remove sof_sdw_mic_codec_mockup_init ASoC: intel: sof_sdw: remove get_next_be_id ASoC: intel: sof_sdw: add link adr order check Pierre-Louis Bossart (2): ASoC: Intel: sof_sdw: fix jack detection on HP Spectre x360 convertible ASoC: Intel: sof_sdw: add SKU for Dell Latitude 9520 sound/soc/intel/boards/sof_sdw.c | 152 ++++++++++---------- sound/soc/intel/boards/sof_sdw_common.h | 7 +- sound/soc/intel/boards/sof_sdw_rt715.c | 7 - sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 7 - 4 files changed, 85 insertions(+), 88 deletions(-) -- 2.17.1

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[PATCH 0/3] ASoC: SOF: Implement support for DMA trace free IPC command
by Peter Ujfalusi 15 Nov '21

15 Nov '21

Hi, This series adds support for the new DMA_TRACE_FREE IPC message to gracefully stop and free the DMA trace related resources on the firmware side. Regards, Peter --- Ranjani Sridharan (3): ASoC: SOF: IPC: Add new IPC command to free trace DMA ASoC: SOF: IPC: update ipc_log_header() ASoC: SOF: trace: send DMA_TRACE_FREE IPC during release include/sound/sof/header.h | 1 + sound/soc/sof/core.c | 2 +- sound/soc/sof/ipc.c | 17 ++++++++++++++++- sound/soc/sof/trace.c | 18 ++++++++++++++++++ 4 files changed, 36 insertions(+), 2 deletions(-) -- 2.33.1

2 4

[PATCH] ASoC: sunxi: sun4i-spdif: Implement IEC958 control
by Jernej Skrabec 15 Nov '21

15 Nov '21

SPDIF core is capable of sending custom status. Implement IEC958 control handling. Signed-off-by: Jernej Skrabec <jernej.skrabec(a)gmail.com> --- sound/soc/sunxi/sun4i-spdif.c | 99 +++++++++++++++++++++++++++++++++++ 1 file changed, 99 insertions(+) diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index a10949bf0ca1..f7438e3ece65 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -21,6 +21,7 @@ #include <linux/platform_device.h> #include <linux/pm_runtime.h> #include <linux/reset.h> +#include <sound/asoundef.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -385,11 +386,109 @@ static int sun4i_spdif_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } +static int sun4i_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int sun4i_spdif_get_status_mask(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 *status = ucontrol->value.iec958.status; + + status[0] = 0xff; + status[1] = 0xff; + status[2] = 0xff; + status[3] = 0xff; + status[4] = 0xff; + status[5] = 0x03; + + return 0; +} + +static int sun4i_spdif_get_status(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + u8 *status = ucontrol->value.iec958.status; + unsigned int reg; + + regmap_read(host->regmap, SUN4I_SPDIF_TXCHSTA0, &reg); + + status[0] = reg & 0xff; + status[1] = (reg >> 8) & 0xff; + status[2] = (reg >> 16) & 0xff; + status[3] = (reg >> 24) & 0xff; + + regmap_read(host->regmap, SUN4I_SPDIF_TXCHSTA1, &reg); + + status[4] = reg & 0xff; + status[5] = (reg >> 8) & 0x3; + + return 0; +} + +static int sun4i_spdif_set_status(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + u8 *status = ucontrol->value.iec958.status; + unsigned int reg; + + reg = (u32)status[3] << 24; + reg |= (u32)status[2] << 16; + reg |= (u32)status[1] << 8; + reg |= (u32)status[0]; + + regmap_write(host->regmap, SUN4I_SPDIF_TXCHSTA0, reg); + + reg = (u32)(status[5] & 3) << 8; + reg |= (u32)status[4]; + + regmap_write(host->regmap, SUN4I_SPDIF_TXCHSTA1, reg); + + reg = SUN4I_SPDIF_TXCFG_CHSTMODE; + if (status[0] & IEC958_AES0_NONAUDIO) + reg |= SUN4I_SPDIF_TXCFG_NONAUDIO; + + regmap_update_bits(host->regmap, SUN4I_SPDIF_TXCFG, + SUN4I_SPDIF_TXCFG_CHSTMODE | + SUN4I_SPDIF_TXCFG_NONAUDIO, reg); + + return 0; +} + +static struct snd_kcontrol_new sun4i_spdif_controls[] = { + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK), + .info = sun4i_spdif_info, + .get = sun4i_spdif_get_status_mask + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = sun4i_spdif_info, + .get = sun4i_spdif_get_status, + .put = sun4i_spdif_set_status + } +}; + static int sun4i_spdif_soc_dai_probe(struct snd_soc_dai *dai) { struct sun4i_spdif_dev *host = snd_soc_dai_get_drvdata(dai); snd_soc_dai_init_dma_data(dai, &host->dma_params_tx, NULL); + snd_soc_add_dai_controls(dai, sun4i_spdif_controls, + ARRAY_SIZE(sun4i_spdif_controls)); + return 0; } -- 2.33.1

2 1

[PATCH v5 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 15 Nov '21

15 Nov '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..c4fed6335230 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..59ed510e6cf1 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[PATCH v5 0/2] Codec driver for TLV320ADC3001 and 3101
by Ricard Wanderlof 15 Nov '21

15 Nov '21

This is a codec driver for the TLV320ADC3001 and TLV320ADC3101 2 channel audio ADC chips from Texas Instruments. Based on an old driver for these chips that I was referred to by Texas Instruments, but which apparently was never upstreamed, I've taken patches from various incarnations, ported them and augmented them with various new features in order to make a working driver. There is currently no support for the on-chip "miniDSP". Although the TLV320ADC3001 and TLV320ADC3101 have similarities with other TI codec chips for which there are existing drivers, upon closer inspection I decided the differences were too great to warrant attempting to modify an existing driver, especially without access to all supported chips and resources for testing them all. There are certainly things that could be improved, among others in the area of PLL register value calculations, but it has been tested on the Axis ARTPEC-8 platform and has been found to work satisfactorily at least with a TLV320ADC3101 operating as an I2S slave at 48 kHz with 32 bit bit depth, 1 or 2 channels, both with the on-chip PLL enabled and disabled. I have not tested this on a TLV320ADC3001 chip; in the original driver the only differences were related to the now-removed initialization code, so I've just retained the compatible strings for both chips. Limitations: - GPIO pins on chip can only be set up as outputs. - Using BCLK as an input to clock chain is supported by the hw but not by driver. Changes v4->v5: Yet another fix based on kernel test robot (lkp(a)intel.com), this time a legitimate issue: the return value from of_get_named_gpio_flags() was crammed into an unsigned int so any potential negative error codes were lost. No changes to the devicetree bindings. Changes v3->v4: More fixes reported by kernel test robot (lkp(a)intel.com). Don't know why these weren't reported the first time. Changes v2->v3: Fix a couple of issues reported by the kernel test robot (lkp(a)intel.com). Changes v1->v2: Large portions of the driver have been reworked from v1 based on remarks from Mark. More specifically: - Remove an unneeded #ifdef and EXPORT_SYMBOL_GPL. - Add ADC3XXX_ prefix to all macro names to avoid namespace contention. - Clean up certain macro names with mixed lower case - upper case names. - Do away with the D0..D7 macros representing bit values. - Fix general formatting: spaces before and after {, }, replace multi spaces with tabs, maximum 2 blank lines anywhere, and other minor issues. - Page register does not need to be volatile. - Rather than use 0 / -6dB enum for input attenuator, use standard TLV. - The dither offset control contains reserved values so use VALUE_ENUM instead of straight ENUM. - Replace vector of SOC_ENUMs with individual variables. - Fix ALSA control names: - Use Volume at the end of all gain controls, and remove "Gain" where applicable. - Use "Switch" at the end of all switch controls, either adding it or changing "switch" into "Switch" as applicable. - Switches and Volumes related to capture should be set up as such, so add "Capture" to widget names where appropriate. This is mostly important for the PGA Capture and Switch widgets, not just for consistency, but also because amixer simple Switch controls expect different names (cap/nocap instead of on/off) for control values for capture. - Make widget names have capital first letters for all consitutent words. - When resuming, the registers will not have changed, so no need to call snd_soc_component_cache_sync(). - Move reset gpio init to I2C probe - Remove needless _priv at end of private struct name. - Make hex constants lower case - _CBM_CFM/_CBS_CFS -> _CBP_CFP/_CBC_CFC - Instead of using device specific ALSA controls to control the two GPIO pins, use GPIOLIB. GPIO pin functions still can be set up in the devicetree as the pins can be used for other functions which are not GPIO related. - Limitation: The GPIO pins can only be used as outputs, even though the hardware alternatively supports using them as inputs. - Handle GPIO configs in a vector rather than separately. - Use more appropriate module_i2c_driver instead of generic module_init/module_exit. - Remove useless initialization data. - As the ADC mute function had no control, add it. - Clean up GPIO initialization which was done both in the I2C probe and in the component probe. - Remove probe, remove, suspend and resume callbacks, which either do not do anything at all anymore, or uselessly just do set_bias_level. - The PLL should not be configured in the devicetree, so move the corresponding definitions to tlv320adc3xxx.h where they can be picked up by a machine driver if needed. - Default to PLL AUTO mode. - Use "reset-gpios" instead of too-generic "gpios" for dt property for configuring codec reset pin. - Use function for printing PLL mode rather than macro. - dev_info is too verbose for most of the printouts, change them to dev_dbg. - Add support for configuring micbias voltage in DT. - Remove ALSA control for mic bias. - Move PLL and NADC, MADC divisor enable to DAPM, including 10 ms post PLL start up delay via POST_PMU event. - ADC is powered on using DAPM, so needless to do it in set_bias_level, thus remove it. - Let DAPM handle BCLK divisor enable. With this, we can remove set_bias_level as it doesn't do anything anymore. - Fix incorrect argument order to a snd_soc_component_update_bits call. - Although the chip itself can operate using BCLK as the clock source, either directly to the divider chain or via the PLL, this is not yet supported in the driver, so require the clocks property for the MCLK for now. - Move PLL rate structure out of .h file where it does not belong. /Ricard Ricard Wanderlof (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 +++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1072 +++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 379 ++++++ 6 files changed, 1625 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[PATCH v4 2/2] ASoC: codec: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 15 Nov '21

15 Nov '21

New codec driver for Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1071 ++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 379 +++++++++++ 4 files changed, 1459 insertions(+) create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 326f2d611ad4..b81c4721c0d5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -220,6 +220,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TDA7419 imply SND_SOC_TFA9879 imply SND_SOC_TFA989X + imply SND_SOC_TLV320ADC3XXX imply SND_SOC_TLV320ADCX140 imply SND_SOC_TLV320AIC23_I2C imply SND_SOC_TLV320AIC23_SPI @@ -1486,6 +1487,12 @@ config SND_SOC_TFA989X Note that the driver currently bypasses the built-in "CoolFlux DSP" and does not support (hardware) volume control. +config SND_SOC_TLV320ADC3XXX + tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" + help + Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 + ADCs. + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9acfbcbfc46d..5112c12e6d76 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -238,6 +238,7 @@ snd-soc-tda7419-objs := tda7419.o snd-soc-tas2770-objs := tas2770.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o +snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o @@ -572,6 +573,7 @@ obj-$(CONFIG_SND_SOC_TDA7419) += snd-soc-tda7419.o obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TFA989X) += snd-soc-tfa989x.o +obj-$(CONFIG_SND_SOC_TLV320ADC3XXX) += snd-soc-tlv320adc3xxx.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c new file mode 100644 index 000000000000..fd048b084c1f --- /dev/null +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -0,0 +1,1071 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov +// +// History: +// +// Author: "Shahina Shaik" < shahina.s(a)mistralsolutions.com > +// Copyright: (C) 2010 Mistral Solutions Pvt Ltd. +// +// Author: Dongge wu <dgwu(a)ambarella.com> +// 2015/10/28 - [Dongge wu] Created file +// Copyright (C) 2014-2018, Ambarella, Inc. +// +// Author: Ricard Wanderlof <ricardw(a)axis.com> +// 2020/11/05: Fixing driver for Linux 4.14: more clocking modes, etc. +// 2021/09/03: Porting driver to Linux 5.4 and later. Added GPIO pin control, +// DAPM controlled clock chain, general cleanup. +// Copyright (C) 2021 Axis Communications AB +// + +#include <dt-bindings/sound/tlv320adc3xxx.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio/driver.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/cdev.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "tlv320adc3xxx.h" + +enum adc3xxx_type { + ADC3001 = 0, + ADC3101 +}; + +struct adc3xxx { + struct device *dev; + enum adc3xxx_type type; + struct clk *mclk; + struct regmap *regmap; + unsigned int pll_mode; + unsigned int rst_pin; + unsigned int rst_active; + unsigned int sysclk; + unsigned int gpio_cfg[ADC3XXX_GPIOS_MAX]; /* value+1 (0 => not set) */ + unsigned int micbias_vg[ADC3XXX_MICBIAS_PINS]; + int master; + u8 page_no; + int use_pll; +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif +}; + +static const unsigned int adc3xxx_gpio_ctrl_reg[ADC3XXX_GPIOS_MAX] = { + ADC3XXX_GPIO1_CTRL, + ADC3XXX_GPIO2_CTRL +}; + +static const unsigned int adc3xxx_micbias_shift[ADC3XXX_MICBIAS_PINS] = { + ADC3XXX_MICBIAS1_SHIFT, + ADC3XXX_MICBIAS2_SHIFT +}; + +static const struct reg_default adc3xxx_defaults[] = { + /* Page 0 */ + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, { 3, 0x00 }, + { 4, 0x00 }, { 5, 0x11 }, { 6, 0x04 }, { 7, 0x00 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x00 }, { 11, 0x00 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x00 }, { 15, 0x00 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x01 }, { 19, 0x01 }, + { 20, 0x80 }, { 21, 0x80 }, { 22, 0x04 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x01 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x02 }, { 30, 0x01 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x10 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x00 }, { 38, 0x02 }, { 39, 0x00 }, + { 40, 0x00 }, { 41, 0x00 }, { 42, 0x00 }, { 43, 0x00 }, + { 44, 0x00 }, { 45, 0x00 }, { 46, 0x00 }, { 47, 0x00 }, + { 48, 0x00 }, { 49, 0x00 }, { 50, 0x00 }, { 51, 0x00 }, + { 52, 0x00 }, { 53, 0x12 }, { 54, 0x00 }, { 55, 0x00 }, + { 56, 0x00 }, { 57, 0x00 }, { 58, 0x00 }, { 59, 0x44 }, + { 60, 0x00 }, { 61, 0x01 }, { 62, 0x00 }, { 63, 0x00 }, + { 64, 0x00 }, { 65, 0x00 }, { 66, 0x00 }, { 67, 0x00 }, + { 68, 0x00 }, { 69, 0x00 }, { 70, 0x00 }, { 71, 0x00 }, + { 72, 0x00 }, { 73, 0x00 }, { 74, 0x00 }, { 75, 0x00 }, + { 76, 0x00 }, { 77, 0x00 }, { 78, 0x00 }, { 79, 0x00 }, + { 80, 0x00 }, { 81, 0x00 }, { 82, 0x88 }, { 83, 0x00 }, + { 84, 0x00 }, { 85, 0x00 }, { 86, 0x00 }, { 87, 0x00 }, + { 88, 0x7f }, { 89, 0x00 }, { 90, 0x00 }, { 91, 0x00 }, + { 92, 0x00 }, { 93, 0x00 }, { 94, 0x00 }, { 95, 0x00 }, + { 96, 0x7f }, { 97, 0x00 }, { 98, 0x00 }, { 99, 0x00 }, + { 100, 0x00 }, { 101, 0x00 }, { 102, 0x00 }, { 103, 0x00 }, + { 104, 0x00 }, { 105, 0x00 }, { 106, 0x00 }, { 107, 0x00 }, + { 108, 0x00 }, { 109, 0x00 }, { 110, 0x00 }, { 111, 0x00 }, + { 112, 0x00 }, { 113, 0x00 }, { 114, 0x00 }, { 115, 0x00 }, + { 116, 0x00 }, { 117, 0x00 }, { 118, 0x00 }, { 119, 0x00 }, + { 120, 0x00 }, { 121, 0x00 }, { 122, 0x00 }, { 123, 0x00 }, + { 124, 0x00 }, { 125, 0x00 }, { 126, 0x00 }, { 127, 0x00 }, + + /* Page 1 */ + { 128, 0x00 }, { 129, 0x00 }, { 130, 0x00 }, { 131, 0x00 }, + { 132, 0x00 }, { 133, 0x00 }, { 134, 0x00 }, { 135, 0x00 }, + { 136, 0x00 }, { 137, 0x00 }, { 138, 0x00 }, { 139, 0x00 }, + { 140, 0x00 }, { 141, 0x00 }, { 142, 0x00 }, { 143, 0x00 }, + { 144, 0x00 }, { 145, 0x00 }, { 146, 0x00 }, { 147, 0x00 }, + { 148, 0x00 }, { 149, 0x00 }, { 150, 0x00 }, { 151, 0x00 }, + { 152, 0x00 }, { 153, 0x00 }, { 154, 0x00 }, { 155, 0x00 }, + { 156, 0x00 }, { 157, 0x00 }, { 158, 0x00 }, { 159, 0x00 }, + { 160, 0x00 }, { 161, 0x00 }, { 162, 0x00 }, { 163, 0x00 }, + { 164, 0x00 }, { 165, 0x00 }, { 166, 0x00 }, { 167, 0x00 }, + { 168, 0x00 }, { 169, 0x00 }, { 170, 0x00 }, { 171, 0x00 }, + { 172, 0x00 }, { 173, 0x00 }, { 174, 0x00 }, { 175, 0x00 }, + { 176, 0x00 }, { 177, 0x00 }, { 178, 0x00 }, { 179, 0x00 }, + { 180, 0xff }, { 181, 0x00 }, { 182, 0x3f }, { 183, 0xff }, + { 184, 0x00 }, { 185, 0x3f }, { 186, 0x00 }, { 187, 0x80 }, + { 188, 0x80 }, { 189, 0x00 }, { 190, 0x00 }, { 191, 0x00 }, +}; + +static bool adc3xxx_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADC3XXX_RESET: + return true; + default: + return false; + } +} + +static const struct regmap_range_cfg adc3xxx_ranges[] = { + { + .range_min = 0, + .range_max = 2 * ADC3XXX_PAGE_SIZE, + .selector_reg = ADC3XXX_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = ADC3XXX_PAGE_SIZE, + } +}; + +static const struct regmap_config adc3xxx_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = adc3xxx_defaults, + .num_reg_defaults = ARRAY_SIZE(adc3xxx_defaults), + + .volatile_reg = adc3xxx_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + + .ranges = adc3xxx_ranges, + .num_ranges = ARRAY_SIZE(adc3xxx_ranges), + .max_register = 2 * ADC3XXX_PAGE_SIZE, +}; + +struct adc3xxx_rate_divs { + u32 mclk; + u32 rate; + u8 pll_p; + u8 pll_r; + u8 pll_j; + u16 pll_d; + u8 nadc; + u8 madc; + u8 aosr; +}; + +/* + * PLL and Clock settings. + * If p member is 0, PLL is not used. + * The order of the entries in this table have the PLL entries before + * the non-PLL entries, so that the PLL modes are preferred unless + * the PLL mode setting says otherwise. + */ +static const struct adc3xxx_rate_divs adc3xxx_divs[] = { + /* mclk, rate, p, r, j, d, nadc, madc, aosr */ + /* 8k rate */ + { 12000000, 8000, 1, 1, 7, 1680, 42, 2, 128 }, + { 12288000, 8000, 1, 1, 7, 0000, 42, 2, 128 }, + /* 11.025k rate */ + { 12000000, 11025, 1, 1, 6, 8208, 29, 2, 128 }, + /* 16k rate */ + { 12000000, 16000, 1, 1, 7, 1680, 21, 2, 128 }, + { 12288000, 16000, 1, 1, 7, 0000, 21, 2, 128 }, + /* 22.05k rate */ + { 12000000, 22050, 1, 1, 7, 560, 15, 2, 128 }, + /* 32k rate */ + { 12000000, 32000, 1, 1, 8, 1920, 12, 2, 128 }, + { 12288000, 32000, 1, 1, 8, 0000, 12, 2, 128 }, + /* 44.1k rate */ + { 12000000, 44100, 1, 1, 7, 5264, 8, 2, 128 }, + /* 48k rate */ + { 12000000, 48000, 1, 1, 7, 1680, 7, 2, 128 }, + { 12288000, 48000, 1, 1, 7, 0000, 7, 2, 128 }, + { 24576000, 48000, 1, 1, 3, 5000, 7, 2, 128 }, /* With PLL */ + { 24576000, 48000, 0, 0, 0, 0000, 2, 2, 128 }, /* Without PLL */ + /* 88.2k rate */ + { 12000000, 88200, 1, 1, 7, 5264, 4, 4, 64 }, + /* 96k rate */ + { 12000000, 96000, 1, 1, 8, 1920, 4, 4, 64 }, +}; + +static int adc3xxx_get_divs(struct device *dev, int mclk, int rate, int pll_mode) +{ + int i; + + dev_dbg(dev, "mclk = %d, rate = %d, clock mode %u\n", + mclk, rate, pll_mode); + for (i = 0; i < ARRAY_SIZE(adc3xxx_divs); i++) { + const struct adc3xxx_rate_divs *mode = &adc3xxx_divs[i]; + + /* Skip this entry if it doesn't fulfill the intended clock + * mode requirement. We consider anything besides the two + * modes below to be the same as ADC3XXX_PLL_AUTO. + */ + if ((pll_mode == ADC3XXX_PLL_BYPASS && mode->pll_p) || + (pll_mode == ADC3XXX_PLL_ENABLE && !mode->pll_p)) + continue; + + if (mode->rate == rate && mode->mclk == mclk) + return i; + } + + dev_info(dev, "Master clock rate %d and sample rate %d is not supported\n", + mclk, rate); + return -EINVAL; +} + +static int adc3xxx_pll_delay(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* 10msec delay needed after PLL power-up to allow + * PLL and dividers to stabilize (datasheet p13). + */ + usleep_range(10000, 20000); + + return 0; +} + +static const char * const adc_softstepping_text[] = { "1 step", "2 step", "off" }; +static SOC_ENUM_SINGLE_DECL(adc_softstepping_enum, ADC3XXX_ADC_DIGITAL, 0, + adc_softstepping_text); + +static const char * const multiplier_text[] = { "1", "2", "4", "8", "16", "32", "64", "128" }; +static SOC_ENUM_SINGLE_DECL(left_agc_attack_mult_enum, + ADC3XXX_LEFT_CHN_AGC_4, 0, multiplier_text); +static SOC_ENUM_SINGLE_DECL(right_agc_attack_mult_enum, + ADC3XXX_RIGHT_CHN_AGC_4, 0, multiplier_text); +static SOC_ENUM_SINGLE_DECL(left_agc_decay_mult_enum, + ADC3XXX_LEFT_CHN_AGC_5, 0, multiplier_text); +static SOC_ENUM_SINGLE_DECL(right_agc_decay_mult_enum, + ADC3XXX_RIGHT_CHN_AGC_5, 0, multiplier_text); + +static const char * const dither_dc_offset_text[] = { + "0mV", "15mV", "30mV", "45mV", "60mV", "75mV", "90mV", "105mV", + "-15mV", "-30mV", "-45mV", "-60mV", "-75mV", "-90mV", "-105mV" +}; +static const unsigned int dither_dc_offset_values[] = { + 0, 1, 2, 3, 4, 5, 6, 7, 9, 10, 11, 12, 13, 14, 15 +}; +static SOC_VALUE_ENUM_DOUBLE_DECL(dither_dc_offset_enum, + ADC3XXX_DITHER_CTRL, + 4, 0, 0xf, dither_dc_offset_text, + dither_dc_offset_values); + +static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fine_tlv, -40, 10, 0); +/* AGC target: 8 values: -5.5, -8, -10, -12, -14, -17, -20, -24 dB */ +/* It would be nice to declare these in the order above, but empirically + * TLV_DB_SCALE_ITEM doesn't take lightly to the increment (second) parameter + * being negative, despite there being examples to the contrary in other + * drivers. So declare these in the order from lowest to highest, and + * set the invert flag in the SOC_DOUBLE_R_TLV declaration instead. + */ +static const DECLARE_TLV_DB_RANGE(agc_target_tlv, + 0, 0, TLV_DB_SCALE_ITEM(-2400, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(-2000, 300, 0), + 4, 6, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 7, 7, TLV_DB_SCALE_ITEM(-550, 0, 0)); +/* Since the 'disabled' value (mute) is at the highest value in the dB + * range (i.e. just before -32 dB) rather than the lowest, we need to resort + * to using a TLV_DB_RANGE in order to get the mute value in the right place. + */ +static const DECLARE_TLV_DB_RANGE(agc_thresh_tlv, + 0, 30, TLV_DB_SCALE_ITEM(-9000, 200, 0), + 31, 31, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +/* AGC hysteresis: 4 values: 1, 2, 4 dB, disabled (= mute) */ +static const DECLARE_TLV_DB_RANGE(agc_hysteresis_tlv, + 0, 1, TLV_DB_SCALE_ITEM(100, 100, 0), + 2, 2, TLV_DB_SCALE_ITEM(400, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +static const DECLARE_TLV_DB_SCALE(agc_max_tlv, 0, 50, 0); +/* Input attenuation: -6 dB or 0 dB */ +static const DECLARE_TLV_DB_SCALE(input_attenuation_tlv, -600, 600, 0); + +static const struct snd_kcontrol_new adc3xxx_snd_controls[] = { + SOC_DOUBLE_R_TLV("PGA Capture Volume", ADC3XXX_LEFT_APGA_CTRL, + ADC3XXX_RIGHT_APGA_CTRL, 0, 80, 0, pga_tlv), + SOC_DOUBLE("PGA Capture Switch", ADC3XXX_ADC_FGA, 7, 3, 1, 1), + SOC_DOUBLE_R("AGC Capture Switch", ADC3XXX_LEFT_CHN_AGC_1, + ADC3XXX_RIGHT_CHN_AGC_1, 7, 1, 0), + SOC_DOUBLE_R_TLV("AGC Target Level Capture Volume", ADC3XXX_LEFT_CHN_AGC_1, + ADC3XXX_RIGHT_CHN_AGC_2, 4, 0x07, 1, agc_target_tlv), + SOC_DOUBLE_R_TLV("AGC Noise Threshold Capture Volume", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 1, 0x1f, 1, agc_thresh_tlv), + SOC_DOUBLE_R_TLV("AGC Hysteresis Capture Volume", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 6, 3, 0, agc_hysteresis_tlv), + SOC_DOUBLE_R("AGC Clip Stepping Capture Switch", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 0, 1, 0), + /* + * Oddly enough, the data sheet says the default value + * for the left/right AGC maximum gain register field + * (ADC3XXX_LEFT/RIGHT_CHN_AGC_3 bits 0..6) is 0x7f = 127 + * (verified empirically) even though this value (indeed, above + * 0x50) is specified as 'Reserved. Do not use.' in the accompanying + * table in the data sheet. + */ + SOC_DOUBLE_R_TLV("AGC Maximum Capture Volume", ADC3XXX_LEFT_CHN_AGC_3, + ADC3XXX_RIGHT_CHN_AGC_3, 0, 0x50, 0, agc_max_tlv), + SOC_DOUBLE_R("AGC Attack Time", ADC3XXX_LEFT_CHN_AGC_4, + ADC3XXX_RIGHT_CHN_AGC_4, 3, 0x1f, 0), + /* Would like to have the multipliers as LR pairs, but there is + * no SOC_ENUM_foo which accepts two values in separate registers. + */ + SOC_ENUM("AGC Left Attack Time Multiplier", left_agc_attack_mult_enum), + SOC_ENUM("AGC Right Attack Time Multiplier", right_agc_attack_mult_enum), + SOC_DOUBLE_R("AGC Decay Time", ADC3XXX_LEFT_CHN_AGC_5, + ADC3XXX_RIGHT_CHN_AGC_5, 3, 0x1f, 0), + SOC_ENUM("AGC Left Decay Time Multiplier", left_agc_decay_mult_enum), + SOC_ENUM("AGC Right Decay Time Multiplier", right_agc_decay_mult_enum), + SOC_DOUBLE_R("AGC Noise Debounce", ADC3XXX_LEFT_CHN_AGC_6, + ADC3XXX_RIGHT_CHN_AGC_6, 0, 0x1f, 0), + SOC_DOUBLE_R("AGC Signal Debounce", ADC3XXX_LEFT_CHN_AGC_7, + ADC3XXX_RIGHT_CHN_AGC_7, 0, 0x0f, 0), + /* Read only register */ + SOC_DOUBLE_R_S_TLV("AGC Applied Capture Volume", ADC3XXX_LEFT_AGC_GAIN, + ADC3XXX_RIGHT_AGC_GAIN, 0, -24, 40, 6, 0, adc_tlv), + /* ADC soft stepping */ + SOC_ENUM("ADC Soft Stepping", adc_softstepping_enum), + /* Left/Right Input attenuation */ + SOC_SINGLE_TLV("Left Input IN_1L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_2L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_3L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_1R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_2L_3L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 6, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_1L_1R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_2R_3R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_1R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_2R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_3R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_1L Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_2R_3R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 6, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_1L_1R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_2L_3L Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 2, 1, 1, input_attenuation_tlv), + SOC_DOUBLE_R_S_TLV("ADC Volume Control Capture Volume", ADC3XXX_LADC_VOL, + ADC3XXX_RADC_VOL, 0, -24, 40, 6, 0, adc_tlv), + /* Empirically, the following doesn't work the way it's supposed + * to. Values 0, -0.1, -0.2 and -0.3 dB result in the same level, and + * -0.4 dB drops about 0.12 dB on a specific chip. + */ + SOC_DOUBLE_TLV("ADC Fine Volume Control Capture Volume", ADC3XXX_ADC_FGA, + 4, 0, 4, 1, adc_fine_tlv), + SOC_SINGLE("Left ADC Unselected CM Bias Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 6, 1, 0), + SOC_SINGLE("Right ADC Unselected CM Bias Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 6, 1, 0), + SOC_ENUM("Dither Control DC Offset", dither_dc_offset_enum), +}; + +/* Left input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Right input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1L Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Left Digital Mic input for left ADC */ +static const struct snd_kcontrol_new left_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Left ADC Capture Switch", + ADC3XXX_ADC_DIGITAL, 3, 0x1, 0), +}; + +/* Right Digital Mic input for Right ADC */ +static const struct snd_kcontrol_new right_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Right ADC Capture Switch", + ADC3XXX_ADC_DIGITAL, 2, 0x1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget adc3xxx_dapm_widgets[] = { + + /* Left Input Selection */ + SND_SOC_DAPM_MIXER("Left Input", SND_SOC_NOPM, 0, 0, + &left_input_mixer_controls[0], + ARRAY_SIZE(left_input_mixer_controls)), + /* Right Input Selection */ + SND_SOC_DAPM_MIXER("Right Input", SND_SOC_NOPM, 0, 0, + &right_input_mixer_controls[0], + ARRAY_SIZE(right_input_mixer_controls)), + /* PGA selection */ + SND_SOC_DAPM_PGA("Left PGA", ADC3XXX_LEFT_APGA_CTRL, 7, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right PGA", ADC3XXX_RIGHT_APGA_CTRL, 7, 1, NULL, 0), + + /* Digital Microphone Input Control for Left/Right ADC */ + SND_SOC_DAPM_MIXER("Left DMic Input", SND_SOC_NOPM, 0, 0, + &left_input_dmic_controls[0], + ARRAY_SIZE(left_input_dmic_controls)), + SND_SOC_DAPM_MIXER("Right DMic Input", SND_SOC_NOPM, 0, 0, + &right_input_dmic_controls[0], + ARRAY_SIZE(right_input_dmic_controls)), + + /* Left/Right ADC */ + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ADC3XXX_ADC_DIGITAL, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ADC3XXX_ADC_DIGITAL, 6, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("IN_1L"), + SND_SOC_DAPM_INPUT("IN_1R"), + SND_SOC_DAPM_INPUT("IN_2L"), + SND_SOC_DAPM_INPUT("IN_2R"), + SND_SOC_DAPM_INPUT("IN_3L"), + SND_SOC_DAPM_INPUT("IN_3R"), + SND_SOC_DAPM_INPUT("DIFL_1L_1R"), + SND_SOC_DAPM_INPUT("DIFL_2L_3L"), + SND_SOC_DAPM_INPUT("DIFL_2R_3R"), + SND_SOC_DAPM_INPUT("DIFR_1L_1R"), + SND_SOC_DAPM_INPUT("DIFR_2L_3L"), + SND_SOC_DAPM_INPUT("DIFR_2R_3R"), + SND_SOC_DAPM_INPUT("DMic_L"), + SND_SOC_DAPM_INPUT("DMic_R"), + + /* Digital audio interface output */ + SND_SOC_DAPM_AIF_OUT("AIF_OUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Clocks */ + SND_SOC_DAPM_SUPPLY("PLL_CLK", ADC3XXX_PLL_PROG_PR, ADC3XXX_ENABLE_PLL_SHIFT, + 0, adc3xxx_pll_delay, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_SUPPLY("ADC_CLK", ADC3XXX_ADC_NADC, ADC3XXX_ENABLE_NADC_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_MOD_CLK", ADC3XXX_ADC_MADC, ADC3XXX_ENABLE_MADC_SHIFT, + 0, NULL, 0), + + /* This refers to the generated BCLK in master mode. */ + SND_SOC_DAPM_SUPPLY("BCLK", ADC3XXX_BCLK_N_DIV, ADC3XXX_ENABLE_BCLK_SHIFT, + 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route adc3xxx_intercon[] = { + /* Left input selection from switches */ + { "Left Input", "IN_1L Capture Switch", "IN_1L" }, + { "Left Input", "IN_2L Capture Switch", "IN_2L" }, + { "Left Input", "IN_3L Capture Switch", "IN_3L" }, + { "Left Input", "DIF_2L_3L Capture Switch", "DIFL_2L_3L" }, + { "Left Input", "DIF_1L_1R Capture Switch", "DIFL_1L_1R" }, + { "Left Input", "DIF_2R_3R Capture Switch", "DIFL_2R_3R" }, + { "Left Input", "IN_1R Capture Switch", "IN_1R" }, + + /* Left input selection to left PGA */ + { "Left PGA", NULL, "Left Input" }, + + /* Left PGA to left ADC */ + { "Left ADC", NULL, "Left PGA" }, + + /* Right input selection from switches */ + { "Right Input", "IN_1R Capture Switch", "IN_1R" }, + { "Right Input", "IN_2R Capture Switch", "IN_2R" }, + { "Right Input", "IN_3R Capture Switch", "IN_3R" }, + { "Right Input", "DIF_2R_3R Capture Switch", "DIFR_2R_3R" }, + { "Right Input", "DIF_1L_1R Capture Switch", "DIFR_1L_1R" }, + { "Right Input", "DIF_2L_3L Capture Switch", "DIFR_2L_3L" }, + { "Right Input", "IN_1L Capture Switch", "IN_1L" }, + + /* Right input selection to right PGA */ + { "Right PGA", NULL, "Right Input" }, + + /* Right PGA to right ADC */ + { "Right ADC", NULL, "Right PGA" }, + + /* Left DMic Input selection from switch */ + { "Left DMic Input", "Left ADC Capture Switch", "DMic_L" }, + + /* Left DMic to left ADC */ + { "Left ADC", NULL, "Left DMic Input" }, + + /* Right DMic Input selection from switch */ + { "Right DMic Input", "Right ADC Capture Switch", "DMic_R" }, + + /* Right DMic to right ADC */ + { "Right ADC", NULL, "Right DMic Input" }, + + /* ADC to AIF output */ + { "AIF_OUT", NULL, "Left ADC" }, + { "AIF_OUT", NULL, "Right ADC" }, + + /* Clocking */ + { "ADC_MOD_CLK", NULL, "ADC_CLK" }, + { "Left ADC", NULL, "ADC_MOD_CLK" }, + { "Right ADC", NULL, "ADC_MOD_CLK" }, + + { "BCLK", NULL, "ADC_CLK" }, +}; + +static const struct snd_soc_dapm_route adc3xxx_pll_intercon[] = { + { "ADC_CLK", NULL, "PLL_CLK" }, +}; + +static const struct snd_soc_dapm_route adc3xxx_bclk_out_intercon[] = { + { "AIF_OUT", NULL, "BCLK" } +}; + +#ifdef CONFIG_GPIOLIB +static int adc3xxx_gpio_request(struct gpio_chip *chip, unsigned int offset) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + + if (offset >= ADC3XXX_GPIOS_MAX) + return -EINVAL; + + /* GPIO1 is offset 0, GPIO2 is offset 1 */ + /* We check here that the GPIO pins are either not configured in the + * DT, or that they purposely are set as outputs. + * (Input mode not yet implemented). + */ + if (adc3xxx->gpio_cfg[offset] != 0 && + adc3xxx->gpio_cfg[offset] != ADC3XXX_GPIO_GPO + 1) + return -EINVAL; + + return 0; +} + +static int adc3xxx_gpio_direction_out(struct gpio_chip *chip, + unsigned int offset, int value) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + + /* Set GPIO output function. */ + return regmap_update_bits(adc3xxx->regmap, + adc3xxx_gpio_ctrl_reg[offset], + ADC3XXX_GPIO_CTRL_CFG_MASK | + ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK, + ADC3XXX_GPIO_GPO << ADC3XXX_GPIO_CTRL_CFG_SHIFT | + !!value << ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_SHIFT); +} + +/* With only GPIO outputs configured, we never get the .direction_out call, + * so we set the output mode and output value in the same call. Hence + * .set in practice does the same thing as .direction_out . + */ +static void adc3xxx_gpio_set(struct gpio_chip *chip, unsigned int offset, + int value) +{ + (void) adc3xxx_gpio_direction_out(chip, offset, value); +} + +/* Even though we only support GPIO output for now, some GPIO clients + * want to read the current pin state using the .get callback. + */ +static int adc3xxx_gpio_get(struct gpio_chip *chip, unsigned int offset) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + unsigned int regval; + int ret; + + /* We only allow output pins, so just read the value set in the output + * pin register field. + */ + ret = regmap_read(adc3xxx->regmap, adc3xxx_gpio_ctrl_reg[offset], &regval); + if (ret) + return ret; + return !!(regval & ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK); +} + +static const struct gpio_chip adc3xxx_gpio_chip = { + .label = "adc3xxx", + .owner = THIS_MODULE, + .request = adc3xxx_gpio_request, + .direction_output = adc3xxx_gpio_direction_out, + .set = adc3xxx_gpio_set, + .get = adc3xxx_gpio_get, + .can_sleep = 1, +}; + +static void adc3xxx_free_gpio(struct adc3xxx *adc3xxx) +{ + gpiochip_remove(&adc3xxx->gpio_chip); +} + +#endif + +static void adc3xxx_init_gpio(struct adc3xxx *adc3xxx) +{ + int gpio, micbias; +#ifdef CONFIG_GPIOLIB + int ret; + + adc3xxx->gpio_chip = adc3xxx_gpio_chip; + adc3xxx->gpio_chip.ngpio = ADC3XXX_GPIOS_MAX; + adc3xxx->gpio_chip.parent = adc3xxx->dev; + adc3xxx->gpio_chip.base = -1; + + ret = gpiochip_add_data(&adc3xxx->gpio_chip, adc3xxx); + if (ret) + dev_err(adc3xxx->dev, "Failed to add gpios: %d\n", ret); +#endif + + /* Set up potential GPIO configuration from the devicetree. + * This allows us to set up things which are not software + * controllable GPIOs, such as PDM microphone I/O, + */ + for (gpio = 0; gpio < ADC3XXX_GPIOS_MAX; gpio++) { + unsigned int cfg = adc3xxx->gpio_cfg[gpio]; + + if (cfg) { + cfg--; /* actual value to use is stored +1 */ + regmap_update_bits(adc3xxx->regmap, + adc3xxx_gpio_ctrl_reg[gpio], + ADC3XXX_GPIO_CTRL_CFG_MASK, + cfg << ADC3XXX_GPIO_CTRL_CFG_SHIFT); + } + } + + /* Set up micbias voltage */ + for (micbias = 0; micbias < ADC3XXX_MICBIAS_PINS; micbias++) { + unsigned int vg = adc3xxx->micbias_vg[micbias]; + + regmap_update_bits(adc3xxx->regmap, + ADC3XXX_MICBIAS_CTRL, + ADC3XXX_MICBIAS_MASK << adc3xxx_micbias_shift[micbias], + vg << adc3xxx_micbias_shift[micbias]); + } +} + +static int adc3xxx_parse_dt_gpio(struct adc3xxx *adc3xxx, + const char *propname, unsigned int *cfg) +{ + struct device *dev = adc3xxx->dev; + struct device_node *np = dev->of_node; + unsigned int val; + + if (!of_property_read_u32(np, propname, &val)) { + if (val & ~15 || val == 7 || val >= 11) { + dev_err(dev, "Invalid property value for '%s'\n", propname); + return -EINVAL; + } + if (val == ADC3XXX_GPIO_GPI) + dev_warn(dev, "GPIO Input read not yet implemented\n"); + *cfg = val + 1; /* 0 => not set up, all others shifted +1 */ + } + return 0; +} + +static int adc3xxx_parse_dt_micbias(struct adc3xxx *adc3xxx, + const char *propname, unsigned int *vg) +{ + struct device *dev = adc3xxx->dev; + struct device_node *np = dev->of_node; + unsigned int val; + + if (!of_property_read_u32(np, propname, &val)) { + if (val >= ADC3XXX_MICBIAS_AVDD) { + dev_err(dev, "Invalid property value for '%s'\n", propname); + return -EINVAL; + } + *vg = val; + } + return 0; +} + +static int adc3xxx_parse_pll_mode(uint32_t val, unsigned int *pll_mode) +{ + if (val != ADC3XXX_PLL_ENABLE && val != ADC3XXX_PLL_BYPASS && + val != ADC3XXX_PLL_AUTO) + return -EINVAL; + + *pll_mode = val; + + return 0; +} + +static void adc3xxx_setup_pll(struct snd_soc_component *component, + int div_entry) +{ + int i = div_entry; + + /* P & R values */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_PR, + (adc3xxx_divs[i].pll_p << ADC3XXX_PLLP_SHIFT) | + (adc3xxx_divs[i].pll_r << ADC3XXX_PLLR_SHIFT)); + /* J value */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_J, + adc3xxx_divs[i].pll_j & ADC3XXX_PLLJ_MASK); + /* D value */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_LSB, + adc3xxx_divs[i].pll_d & ADC3XXX_PLLD_LSB_MASK); + snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_MSB, + (adc3xxx_divs[i].pll_d >> 8) & ADC3XXX_PLLD_MSB_MASK); +} + +static int adc3xxx_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 snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(dai->component); + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + int i, width = 16; + u8 iface_len, bdiv; + + i = adc3xxx_get_divs(component->dev, adc3xxx->sysclk, + params_rate(params), adc3xxx->pll_mode); + + if (i < 0) + return i; + + /* select data word length */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface_len = ADC3XXX_IFACE_16BITS; + width = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_len = ADC3XXX_IFACE_20BITS; + width = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_len = ADC3XXX_IFACE_24BITS; + width = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_len = ADC3XXX_IFACE_32BITS; + width = 32; + break; + default: + dev_err(component->dev, "Unsupported serial data format\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, ADC3XXX_INTERFACE_CTRL_1, + ADC3XXX_WLENGTH_MASK, iface_len); + if (adc3xxx_divs[i].pll_p) { /* If PLL used for this mode */ + adc3xxx_setup_pll(component, i); + snd_soc_component_write(component, ADC3XXX_CLKGEN_MUX, ADC3XXX_USE_PLL); + if (!adc3xxx->use_pll) { + snd_soc_dapm_add_routes(dapm, adc3xxx_pll_intercon, + ARRAY_SIZE(adc3xxx_pll_intercon)); + adc3xxx->use_pll = 1; + } + } else { + snd_soc_component_write(component, ADC3XXX_CLKGEN_MUX, ADC3XXX_NO_PLL); + if (adc3xxx->use_pll) { + snd_soc_dapm_del_routes(dapm, adc3xxx_pll_intercon, + ARRAY_SIZE(adc3xxx_pll_intercon)); + adc3xxx->use_pll = 0; + } + } + + /* NADC */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_NADC, + ADC3XXX_NADC_MASK, adc3xxx_divs[i].nadc); + /* MADC */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_MADC, + ADC3XXX_MADC_MASK, adc3xxx_divs[i].madc); + /* AOSR */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_AOSR, + ADC3XXX_AOSR_MASK, adc3xxx_divs[i].aosr); + /* BDIV N Value */ + /* BCLK is (by default) set up to be derived from ADC_CLK */ + bdiv = (adc3xxx_divs[i].aosr * adc3xxx_divs[i].madc) / (2 * width); + snd_soc_component_update_bits(component, ADC3XXX_BCLK_N_DIV, + ADC3XXX_BDIV_MASK, bdiv); + + return 0; +} + +static const char *adc3xxx_pll_mode_text(int pll_mode) +{ + switch (pll_mode) { + case ADC3XXX_PLL_AUTO: + return "PLL auto"; + case ADC3XXX_PLL_ENABLE: + return "PLL enable"; + case ADC3XXX_PLL_BYPASS: + return "PLL bypass"; + default: + break; + } + + return "PLL unknown"; +} + +static int adc3xxx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + int ret; + + ret = adc3xxx_parse_pll_mode(clk_id, &adc3xxx->pll_mode); + if (ret < 0) + return ret; + + adc3xxx->sysclk = freq; + dev_dbg(component->dev, "Set sysclk to %u Hz, %s\n", + freq, adc3xxx_pll_mode_text(adc3xxx->pll_mode)); + return 0; +} + +static int adc3xxx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + u8 clkdir = 0, format = 0; + int master = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + master = 1; + clkdir = ADC3XXX_BCLK_MASTER | ADC3XXX_WCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBC_CFC: + master = 0; + break; + default: + dev_err(component->dev, "Invalid DAI clock setup\n"); + return -EINVAL; + } + + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF: + format = ADC3XXX_FORMAT_DSP; + break; + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF: + format = ADC3XXX_FORMAT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_RJF; + break; + case SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_LJF; + break; + default: + dev_err(component->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* Add/del route enabling BCLK output as applicable */ + if (master && !adc3xxx->master) + snd_soc_dapm_add_routes(dapm, adc3xxx_bclk_out_intercon, + ARRAY_SIZE(adc3xxx_bclk_out_intercon)); + else if (!master && adc3xxx->master) + snd_soc_dapm_del_routes(dapm, adc3xxx_bclk_out_intercon, + ARRAY_SIZE(adc3xxx_bclk_out_intercon)); + adc3xxx->master = master; + + /* set clock direction and format */ + return snd_soc_component_update_bits(component, + ADC3XXX_INTERFACE_CTRL_1, + ADC3XXX_CLKDIR_MASK | ADC3XXX_FORMAT_MASK, + clkdir | format); +} + +static struct snd_soc_dai_ops adc3xxx_dai_ops = { + .hw_params = adc3xxx_hw_params, + .set_sysclk = adc3xxx_set_dai_sysclk, + .set_fmt = adc3xxx_set_dai_fmt, +}; + +struct snd_soc_dai_driver adc3xxx_dai = { + .name = "tlv320adc3xxx-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ADC3XXX_RATES, + .formats = ADC3XXX_FORMATS, + }, + .ops = &adc3xxx_dai_ops, +}; + +static const struct snd_soc_component_driver soc_component_dev_adc3xxx = { + .controls = adc3xxx_snd_controls, + .num_controls = ARRAY_SIZE(adc3xxx_snd_controls), + .dapm_widgets = adc3xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adc3xxx_dapm_widgets), + .dapm_routes = adc3xxx_intercon, + .num_dapm_routes = ARRAY_SIZE(adc3xxx_intercon), +}; + +static int adc3xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; + struct adc3xxx *adc3xxx = NULL; + enum of_gpio_flags flags; + int ret; + + adc3xxx = devm_kzalloc(dev, sizeof(struct adc3xxx), GFP_KERNEL); + if (!adc3xxx) + return -ENOMEM; + adc3xxx->dev = dev; + + adc3xxx->mclk = devm_clk_get(dev, NULL); + if (IS_ERR(adc3xxx->mclk)) { + /* + * The chip itself supports running off the BCLK either + * directly or via the PLL, but the driver does not (yet), so + * having a specified mclk is required. Otherwise, we could + * use the lack of a clocks property to indicate when BCLK is + * intended as the clock source. + */ + return PTR_ERR(adc3xxx->mclk); + } else if (adc3xxx->mclk) { + ret = clk_prepare_enable(adc3xxx->mclk); + if (ret < 0) + return ret; + dev_dbg(dev, "Enabled MCLK, freq %lu Hz\n", clk_get_rate(adc3xxx->mclk)); + } + + adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmdin-gpio1", &adc3xxx->gpio_cfg[0]); + adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmclk-gpio2", &adc3xxx->gpio_cfg[1]); + adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias1-vg", &adc3xxx->micbias_vg[0]); + adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias2-vg", &adc3xxx->micbias_vg[1]); + + adc3xxx->regmap = devm_regmap_init_i2c(i2c, &adc3xxx_regmap); + if (IS_ERR(adc3xxx->regmap)) { + ret = PTR_ERR(adc3xxx->regmap); + return ret; + } + + i2c_set_clientdata(i2c, adc3xxx); + + adc3xxx->rst_pin = of_get_named_gpio_flags(np, "reset-gpios", 0, &flags); + if (adc3xxx->rst_pin < 0 || !gpio_is_valid(adc3xxx->rst_pin)) + return -ENXIO; + + adc3xxx->rst_active = !(flags & OF_GPIO_ACTIVE_LOW); + adc3xxx->type = id->driver_data; + + ret = devm_gpio_request(dev, adc3xxx->rst_pin, "adc3xxx reset"); + if (ret < 0) { + dev_err(dev, "Failed to request rst_pin: %d\n", ret); + return ret; + } + + /* Reset codec chip */ + gpio_direction_output(adc3xxx->rst_pin, adc3xxx->rst_active); + usleep_range(2000, 100000); /* Requirement: > 10 ns (datasheet p13) */ + gpio_direction_output(adc3xxx->rst_pin, !adc3xxx->rst_active); + + /* Potentially set up pins used as GPIOs */ + adc3xxx_init_gpio(adc3xxx); + + ret = snd_soc_register_component(dev, + &soc_component_dev_adc3xxx, &adc3xxx_dai, 1); + if (ret < 0) + dev_err(dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int __exit adc3xxx_i2c_remove(struct i2c_client *client) +{ + struct adc3xxx *adc3xxx = i2c_get_clientdata(client); + + if (adc3xxx->mclk) + clk_disable_unprepare(adc3xxx->mclk); +#ifdef CONFIG_GPIOLIB + adc3xxx_free_gpio(adc3xxx); +#endif + snd_soc_unregister_component(&client->dev); + return 0; +} + +static const struct of_device_id tlv320adc3xxx_of_match[] = { + { .compatible = "ti,tlv320adc3001", }, + { .compatible = "ti,tlv320adc3101", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320adc3xxx_of_match); + +static const struct i2c_device_id adc3xxx_i2c_id[] = { + { "tlv320adc3001", ADC3001 }, + { "tlv320adc3101", ADC3101 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, adc3xxx_i2c_id); + +static struct i2c_driver adc3xxx_i2c_driver = { + .driver = { + .name = "tlv320adc3xxx-codec", + .of_match_table = tlv320adc3xxx_of_match, + }, + .probe = adc3xxx_i2c_probe, + .remove = adc3xxx_i2c_remove, + .id_table = adc3xxx_i2c_id, +}; + +module_i2c_driver(adc3xxx_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320ADC3xxx codec driver"); +MODULE_AUTHOR(" shahina.s(a)mistralsolutions.com "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/tlv320adc3xxx.h b/sound/soc/codecs/tlv320adc3xxx.h new file mode 100644 index 000000000000..82f82c3e7bc5 --- /dev/null +++ b/sound/soc/codecs/tlv320adc3xxx.h @@ -0,0 +1,379 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov + * + * History: + * + * Author: "Shahina Shaik" < shahina.s(a)mistralsolutions.com > + * Copyright: (C) 2010 Mistral Solutions Pvt Ltd. + * + * Author: Dongge wu <dgwu(a)ambarella.com> + * 2015/10/28 - [Dongge wu] Created file + * Copyright (C) 2014-2018, Ambarella, Inc. + * + * Author: Ricard Wanderlof <ricardw(a)axis.com> + * 2020/11/05: Fixing driver for Linux 4.14: more clocking modes, etc. + * 2021/09/03: Porting to Linux 5.4, with enhancements. + * Copyright (C) 2021 Axis Communications AB + * + */ + +#ifndef _ADC3XXX_H +#define _ADC3XXX_H + +/* 8 bit mask value */ +#define ADC3XXX_8BITS_MASK 0xFF + +#define ADC3XXX_PAGE_SIZE 128 +#define ADC3XXX_REG(page, reg) ((page * ADC3XXX_PAGE_SIZE) + reg) + +#define ADC3XXX_MICBIAS_PINS 2 + +/* Number of GPIO pins exposed via the gpiolib interface */ +#define ADC3XXX_GPIOS_MAX 2 + +/* + * PLL modes, to be used for clk_id for set_sysclk callback. + * + * The default behavior (AUTO) is to take the first matching entry in the clock + * table, which is intended to be the PLL based one if there is more than one. + * + * Setting the clock source using simple-card (clocks or + * system-clock-frequency property) sets clk_id = 0 = ADC3XXX_PLL_AUTO. + */ +#define ADC3XXX_PLL_AUTO 0 /* Use first available mode */ +#define ADC3XXX_PLL_ENABLE 1 /* Use PLL for clock generation */ +#define ADC3XXX_PLL_BYPASS 2 /* Don't use PLL for clock generation */ + + +/****************************************************************************/ +/* Page 0 Registers */ +/****************************************************************************/ + +/* Page select register */ +#define ADC3XXX_PAGE_SELECT ADC3XXX_REG(0, 0) +/* Software reset register */ +#define ADC3XXX_RESET ADC3XXX_REG(0, 1) + +/* 2-3 Reserved */ + +/* PLL programming register B */ +#define ADC3XXX_CLKGEN_MUX ADC3XXX_REG(0, 4) +/* PLL P and R-Val */ +#define ADC3XXX_PLL_PROG_PR ADC3XXX_REG(0, 5) +/* PLL J-Val */ +#define ADC3XXX_PLL_PROG_J ADC3XXX_REG(0, 6) +/* PLL D-Val MSB */ +#define ADC3XXX_PLL_PROG_D_MSB ADC3XXX_REG(0, 7) +/* PLL D-Val LSB */ +#define ADC3XXX_PLL_PROG_D_LSB ADC3XXX_REG(0, 8) + +/* 9-17 Reserved */ + +/* ADC NADC */ +#define ADC3XXX_ADC_NADC ADC3XXX_REG(0, 18) +/* ADC MADC */ +#define ADC3XXX_ADC_MADC ADC3XXX_REG(0, 19) +/* ADC AOSR */ +#define ADC3XXX_ADC_AOSR ADC3XXX_REG(0, 20) +/* ADC IADC */ +#define ADC3XXX_ADC_IADC ADC3XXX_REG(0, 21) + +/* 23-24 Reserved */ + +/* CLKOUT MUX */ +#define ADC3XXX_CLKOUT_MUX ADC3XXX_REG(0, 25) +/* CLOCKOUT M divider value */ +#define ADC3XXX_CLKOUT_M_DIV ADC3XXX_REG(0, 26) +/*Audio Interface Setting Register 1*/ +#define ADC3XXX_INTERFACE_CTRL_1 ADC3XXX_REG(0, 27) +/* Data Slot Offset (Ch_Offset_1) */ +#define ADC3XXX_CH_OFFSET_1 ADC3XXX_REG(0, 28) +/* ADC interface control 2 */ +#define ADC3XXX_INTERFACE_CTRL_2 ADC3XXX_REG(0, 29) +/* BCLK N Divider */ +#define ADC3XXX_BCLK_N_DIV ADC3XXX_REG(0, 30) +/* Secondary audio interface control 1 */ +#define ADC3XXX_INTERFACE_CTRL_3 ADC3XXX_REG(0, 31) +/* Secondary audio interface control 2 */ +#define ADC3XXX_INTERFACE_CTRL_4 ADC3XXX_REG(0, 32) +/* Secondary audio interface control 3 */ +#define ADC3XXX_INTERFACE_CTRL_5 ADC3XXX_REG(0, 33) +/* I2S sync */ +#define ADC3XXX_I2S_SYNC ADC3XXX_REG(0, 34) + +/* 35 Reserved */ + +/* ADC flag register */ +#define ADC3XXX_ADC_FLAG ADC3XXX_REG(0, 36) +/* Data slot offset 2 (Ch_Offset_2) */ +#define ADC3XXX_CH_OFFSET_2 ADC3XXX_REG(0, 37) +/* I2S TDM control register */ +#define ADC3XXX_I2S_TDM_CTRL ADC3XXX_REG(0, 38) + +/* 39-41 Reserved */ + +/* Interrupt flags (overflow) */ +#define ADC3XXX_INTR_FLAG_1 ADC3XXX_REG(0, 42) +/* Interrupt flags (overflow) */ +#define ADC3XXX_INTR_FLAG_2 ADC3XXX_REG(0, 43) + +/* 44 Reserved */ + +/* Interrupt flags ADC */ +#define ADC3XXX_INTR_FLAG_ADC1 ADC3XXX_REG(0, 45) + +/* 46 Reserved */ + +/* Interrupt flags ADC */ +#define ADC3XXX_INTR_FLAG_ADC2 ADC3XXX_REG(0, 47) +/* INT1 interrupt control */ +#define ADC3XXX_INT1_CTRL ADC3XXX_REG(0, 48) +/* INT2 interrupt control */ +#define ADC3XXX_INT2_CTRL ADC3XXX_REG(0, 49) + +/* 50 Reserved */ + +/* DMCLK/GPIO2 control */ +#define ADC3XXX_GPIO2_CTRL ADC3XXX_REG(0, 51) +/* DMDIN/GPIO1 control */ +#define ADC3XXX_GPIO1_CTRL ADC3XXX_REG(0, 52) +/* DOUT Control */ +#define ADC3XXX_DOUT_CTRL ADC3XXX_REG(0, 53) + +/* 54-56 Reserved */ + +/* ADC sync control 1 */ +#define ADC3XXX_SYNC_CTRL_1 ADC3XXX_REG(0, 57) +/* ADC sync control 2 */ +#define ADC3XXX_SYNC_CTRL_2 ADC3XXX_REG(0, 58) +/* ADC CIC filter gain control */ +#define ADC3XXX_CIC_GAIN_CTRL ADC3XXX_REG(0, 59) + +/* 60 Reserved */ + +/* ADC processing block selection */ +#define ADC3XXX_PRB_SELECT ADC3XXX_REG(0, 61) +/* Programmable instruction mode control bits */ +#define ADC3XXX_INST_MODE_CTRL ADC3XXX_REG(0, 62) + +/* 63-79 Reserved */ + +/* Digital microphone polarity control */ +#define ADC3XXX_MIC_POLARITY_CTRL ADC3XXX_REG(0, 80) +/* ADC Digital */ +#define ADC3XXX_ADC_DIGITAL ADC3XXX_REG(0, 81) +/* ADC Fine Gain Adjust */ +#define ADC3XXX_ADC_FGA ADC3XXX_REG(0, 82) +/* Left ADC Channel Volume Control */ +#define ADC3XXX_LADC_VOL ADC3XXX_REG(0, 83) +/* Right ADC Channel Volume Control */ +#define ADC3XXX_RADC_VOL ADC3XXX_REG(0, 84) +/* ADC phase compensation */ +#define ADC3XXX_ADC_PHASE_COMP ADC3XXX_REG(0, 85) +/* Left Channel AGC Control Register 1 */ +#define ADC3XXX_LEFT_CHN_AGC_1 ADC3XXX_REG(0, 86) +/* Left Channel AGC Control Register 2 */ +#define ADC3XXX_LEFT_CHN_AGC_2 ADC3XXX_REG(0, 87) +/* Left Channel AGC Control Register 3 */ +#define ADC3XXX_LEFT_CHN_AGC_3 ADC3XXX_REG(0, 88) +/* Left Channel AGC Control Register 4 */ +#define ADC3XXX_LEFT_CHN_AGC_4 ADC3XXX_REG(0, 89) +/* Left Channel AGC Control Register 5 */ +#define ADC3XXX_LEFT_CHN_AGC_5 ADC3XXX_REG(0, 90) +/* Left Channel AGC Control Register 6 */ +#define ADC3XXX_LEFT_CHN_AGC_6 ADC3XXX_REG(0, 91) +/* Left Channel AGC Control Register 7 */ +#define ADC3XXX_LEFT_CHN_AGC_7 ADC3XXX_REG(0, 92) +/* Left AGC gain */ +#define ADC3XXX_LEFT_AGC_GAIN ADC3XXX_REG(0, 93) +/* Right Channel AGC Control Register 1 */ +#define ADC3XXX_RIGHT_CHN_AGC_1 ADC3XXX_REG(0, 94) +/* Right Channel AGC Control Register 2 */ +#define ADC3XXX_RIGHT_CHN_AGC_2 ADC3XXX_REG(0, 95) +/* Right Channel AGC Control Register 3 */ +#define ADC3XXX_RIGHT_CHN_AGC_3 ADC3XXX_REG(0, 96) +/* Right Channel AGC Control Register 4 */ +#define ADC3XXX_RIGHT_CHN_AGC_4 ADC3XXX_REG(0, 97) +/* Right Channel AGC Control Register 5 */ +#define ADC3XXX_RIGHT_CHN_AGC_5 ADC3XXX_REG(0, 98) +/* Right Channel AGC Control Register 6 */ +#define ADC3XXX_RIGHT_CHN_AGC_6 ADC3XXX_REG(0, 99) +/* Right Channel AGC Control Register 7 */ +#define ADC3XXX_RIGHT_CHN_AGC_7 ADC3XXX_REG(0, 100) +/* Right AGC gain */ +#define ADC3XXX_RIGHT_AGC_GAIN ADC3XXX_REG(0, 101) + +/* 102-127 Reserved */ + +/****************************************************************************/ +/* Page 1 Registers */ +/****************************************************************************/ +/* 1-25 Reserved */ + +/* Dither control */ +#define ADC3XXX_DITHER_CTRL ADC3XXX_REG(1, 26) + +/* 27-50 Reserved */ + +/* MICBIAS Configuration Register */ +#define ADC3XXX_MICBIAS_CTRL ADC3XXX_REG(1, 51) +/* Left ADC input selection for Left PGA */ +#define ADC3XXX_LEFT_PGA_SEL_1 ADC3XXX_REG(1, 52) + +/* 53 Reserved */ + +/* Right ADC input selection for Left PGA */ +#define ADC3XXX_LEFT_PGA_SEL_2 ADC3XXX_REG(1, 54) +/* Right ADC input selection for right PGA */ +#define ADC3XXX_RIGHT_PGA_SEL_1 ADC3XXX_REG(1, 55) + +/* 56 Reserved */ + +/* Right ADC input selection for right PGA */ +#define ADC3XXX_RIGHT_PGA_SEL_2 ADC3XXX_REG(1, 57) + +/* 58 Reserved */ + +/* Left analog PGA settings */ +#define ADC3XXX_LEFT_APGA_CTRL ADC3XXX_REG(1, 59) +/* Right analog PGA settings */ +#define ADC3XXX_RIGHT_APGA_CTRL ADC3XXX_REG(1, 60) +/* ADC Low current Modes */ +#define ADC3XXX_LOW_CURRENT_MODES ADC3XXX_REG(1, 61) +/* ADC analog PGA flags */ +#define ADC3XXX_ANALOG_PGA_FLAGS ADC3XXX_REG(1, 62) + +/* 63-127 Reserved */ + +/****************************************************************************/ +/* Macros and definitions */ +/****************************************************************************/ + +#define ADC3XXX_RATES SNDRV_PCM_RATE_8000_96000 +#define ADC3XXX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/****************************************************************************/ +/* ADC3XXX Register bits */ +/****************************************************************************/ +/* PLL Enable bits */ +#define ADC3XXX_ENABLE_PLL_SHIFT 7 +#define ADC3XXX_ENABLE_PLL (1 << ADC3XXX_ENABLE_PLL_SHIFT) +#define ADC3XXX_ENABLE_NADC_SHIFT 7 +#define ADC3XXX_ENABLE_NADC (1 << ADC3XXX_ENABLE_NADC_SHIFT) +#define ADC3XXX_ENABLE_MADC_SHIFT 7 +#define ADC3XXX_ENABLE_MADC (1 << ADC3XXX_ENABLE_MADC_SHIFT) +#define ADC3XXX_ENABLE_BCLK_SHIFT 7 +#define ADC3XXX_ENABLE_BCLK (1 << ADC3XXX_ENABLE_BCLK_SHIFT) + +/* Power bits */ +#define ADC3XXX_LADC_PWR_ON 0x80 +#define ADC3XXX_RADC_PWR_ON 0x40 + +#define ADC3XXX_SOFT_RESET 0x01 +#define ADC3XXX_BCLK_MASTER 0x08 +#define ADC3XXX_WCLK_MASTER 0x04 + +/* Interface register masks */ +#define ADC3XXX_FORMAT_MASK 0xc0 +#define ADC3XXX_FORMAT_SHIFT 6 +#define ADC3XXX_WLENGTH_MASK 0x30 +#define ADC3XXX_WLENGTH_SHIFT 4 +#define ADC3XXX_CLKDIR_MASK 0x0c +#define ADC3XXX_CLKDIR_SHIFT 2 + +/* Interface register bit patterns */ +#define ADC3XXX_FORMAT_I2S (0 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_DSP (1 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_RJF (2 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_LJF (3 << ADC3XXX_FORMAT_SHIFT) + +#define ADC3XXX_IFACE_16BITS (0 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_20BITS (1 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_24BITS (2 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_32BITS (3 << ADC3XXX_WLENGTH_SHIFT) + +/* PLL P/R bit offsets */ +#define ADC3XXX_PLLP_SHIFT 4 +#define ADC3XXX_PLLR_SHIFT 0 +#define ADC3XXX_PLL_PR_MASK 0x7f +#define ADC3XXX_PLLJ_MASK 0x3f +#define ADC3XXX_PLLD_MSB_MASK 0x3f +#define ADC3XXX_PLLD_LSB_MASK 0xff +#define ADC3XXX_NADC_MASK 0x7f +#define ADC3XXX_MADC_MASK 0x7f +#define ADC3XXX_AOSR_MASK 0xff +#define ADC3XXX_IADC_MASK 0xff +#define ADC3XXX_BDIV_MASK 0x7f + +/* PLL_CLKIN bits */ +#define ADC3XXX_PLL_CLKIN_SHIFT 2 +#define ADC3XXX_PLL_CLKIN_MCLK 0x0 +#define ADC3XXX_PLL_CLKIN_BCLK 0x1 +#define ADC3XXX_PLL_CLKIN_ZERO 0x3 + +/* CODEC_CLKIN bits */ +#define ADC3XXX_CODEC_CLKIN_SHIFT 0 +#define ADC3XXX_CODEC_CLKIN_MCLK 0x0 +#define ADC3XXX_CODEC_CLKIN_BCLK 0x1 +#define ADC3XXX_CODEC_CLKIN_PLL_CLK 0x3 + +#define ADC3XXX_USE_PLL ((ADC3XXX_PLL_CLKIN_MCLK << ADC3XXX_PLL_CLKIN_SHIFT) | \ + (ADC3XXX_CODEC_CLKIN_PLL_CLK << ADC3XXX_CODEC_CLKIN_SHIFT)) +#define ADC3XXX_NO_PLL ((ADC3XXX_PLL_CLKIN_ZERO << ADC3XXX_PLL_CLKIN_SHIFT) | \ + (ADC3XXX_CODEC_CLKIN_MCLK << ADC3XXX_CODEC_CLKIN_SHIFT)) + +/* Analog PGA control bits */ +#define ADC3XXX_LPGA_MUTE 0x80 +#define ADC3XXX_RPGA_MUTE 0x80 + +#define ADC3XXX_LPGA_GAIN_MASK 0x7f +#define ADC3XXX_RPGA_GAIN_MASK 0x7f + +/* ADC current modes */ +#define ADC3XXX_ADC_LOW_CURR_MODE 0x01 + +/* Left ADC Input selection bits */ +#define ADC3XXX_LCH_SEL1_SHIFT 0 +#define ADC3XXX_LCH_SEL2_SHIFT 2 +#define ADC3XXX_LCH_SEL3_SHIFT 4 +#define ADC3XXX_LCH_SEL4_SHIFT 6 + +#define ADC3XXX_LCH_SEL1X_SHIFT 0 +#define ADC3XXX_LCH_SEL2X_SHIFT 2 +#define ADC3XXX_LCH_SEL3X_SHIFT 4 +#define ADC3XXX_LCH_COMMON_MODE 0x40 +#define ADC3XXX_BYPASS_LPGA 0x80 + +/* Right ADC Input selection bits */ +#define ADC3XXX_RCH_SEL1_SHIFT 0 +#define ADC3XXX_RCH_SEL2_SHIFT 2 +#define ADC3XXX_RCH_SEL3_SHIFT 4 +#define ADC3XXX_RCH_SEL4_SHIFT 6 + +#define ADC3XXX_RCH_SEL1X_SHIFT 0 +#define ADC3XXX_RCH_SEL2X_SHIFT 2 +#define ADC3XXX_RCH_SEL3X_SHIFT 4 +#define ADC3XXX_RCH_COMMON_MODE 0x40 +#define ADC3XXX_BYPASS_RPGA 0x80 + +/* MICBIAS control bits */ +#define ADC3XXX_MICBIAS_MASK 0x2 +#define ADC3XXX_MICBIAS1_SHIFT 5 +#define ADC3XXX_MICBIAS2_SHIFT 3 + +#define ADC3XXX_ADC_MAX_VOLUME 64 +#define ADC3XXX_ADC_POS_VOL 24 + +/* GPIO control bits (GPIO1_CTRL and GPIO2_CTRL) */ +#define ADC3XXX_GPIO_CTRL_CFG_MASK 0x3c +#define ADC3XXX_GPIO_CTRL_CFG_SHIFT 2 +#define ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK 0x01 +#define ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_SHIFT 0 +#define ADC3XXX_GPIO_CTRL_INPUT_VALUE_MASK 0x02 +#define ADC3XXX_GPIO_CTRL_INPUT_VALUE_SHIFT 1 + +#endif /* _ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[RESEND PATCH 00/10] ASoC: Intel: sof_sdw: Use fixed DAI link id
by Bard Liao 15 Nov '21

15 Nov '21

This series provides a way to use constant dailink numbers for different devices. So that we don't need to renumber them in topologies. Some patches with different purpose are sent together in this series since they are dependent. Bard Liao (8): ASoC: intel: sof_sdw: return the original error number ASoC: intel: sof_sdw: rename be_index/link_id to link_index ASoC: intel: sof_sdw: Use a fixed DAI link id for AMP ASoC: intel: sof_sdw: move DMIC link id overwrite to create_sdw_dailink ASoC: intel: sof_sdw: remove SOF_RT715_DAI_ID_FIX quirk ASoC: intel: sof_sdw: remove sof_sdw_mic_codec_mockup_init ASoC: intel: sof_sdw: remove get_next_be_id ASoC: intel: sof_sdw: add link adr order check Pierre-Louis Bossart (2): ASoC: Intel: sof_sdw: fix jack detection on HP Spectre x360 convertible ASoC: Intel: sof_sdw: add SKU for Dell Latitude 9520 sound/soc/intel/boards/sof_sdw.c | 152 ++++++++++---------- sound/soc/intel/boards/sof_sdw_common.h | 7 +- sound/soc/intel/boards/sof_sdw_rt715.c | 7 - sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 7 - 4 files changed, 85 insertions(+), 88 deletions(-) -- 2.17.1

2 11

hw: add option to specify dynamic DAPM routing to apply when the plugin is opened
by GitHub pull_request - opened 15 Nov '21

15 Nov '21

alsa-project/alsa-lib pull request #192 was opened from bku-sue: The kernel's sound soc subsystem has a DAPM functionality, which makes complex audio routing easy, and most of it is done inside the kernel. However, in the case of dynamic routing (when there is a multiplexer in the signal path), the multiplexer is exposed as an enum MIXER control which must be set from userspace. As the selection of these controls directly influences what audio path is captured/played, the sensible thing to do is to create multiple HW plugins in the ALSA conf, each referring to the same HW device number, but with different routing parameters. Request URL : https://github.com/alsa-project/alsa-lib/pull/192 Patch URL : https://github.com/alsa-project/alsa-lib/pull/192.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

hw: add "min_rate" and "max_rate" as alternatives to single "rate" parameter
by GitHub pull_request - opened 15 Nov '21

15 Nov '21

alsa-project/alsa-lib pull request #191 was opened from bku-sue: Our typical use case is to limit the rate to between 44100 and 48000, as with these boundaries the resampling becomes easier, than if only allowing 48000. Request URL : https://github.com/alsa-project/alsa-lib/pull/191 Patch URL : https://github.com/alsa-project/alsa-lib/pull/191.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Token to create an account on the wiki
by Lucas 15 Nov '21

15 Nov '21

I was hoping to add some Edirol/Roland devices to the wiki. Signing up at https://alsa-project.org/main/index.php?title=Special:UserLogin&type=signup… says ask the mailing list for a token. Thanks, Lucas

2 3

[PATCH] ALSA: hda: intel: More comprehensive PM runtime setup for controller driver
by Takashi Iwai 15 Nov '21

15 Nov '21

Currently we haven't explicitly enable and allow/forbid the runtime PM at the probe and the remove phases of HD-audio controller driver, and this was the reason of a GPF mentioned in the commit e81478bbe7a1 ("ALSA: hda: fix general protection fault in azx_runtime_idle"); namely, even after the resources are released, the runtime PM might be still invoked by the bound graphics driver during the remove of the controller driver. Although we've fixed it by clearing the drvdata reference, it'd be also better to cover the runtime PM issue more properly. This patch adds a few more pm_runtime_*() calls at the probe and the remove time for setting and cleaning up the runtime PM. Particularly, now more explicitly pm_runtime_enable() and _disable() get called as well as pm_runtime_forbid() call at the remove callback, so that a use-after-free should be avoided. Reported-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Tested-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211110210307.1172004-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fe51163f2d82..45e85180048c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1347,8 +1347,14 @@ static void azx_free(struct azx *chip) if (hda->freed) return; - if (azx_has_pm_runtime(chip) && chip->running) + if (azx_has_pm_runtime(chip) && chip->running) { pm_runtime_get_noresume(&pci->dev); + pm_runtime_disable(&pci->dev); + pm_runtime_set_suspended(&pci->dev); + pm_runtime_forbid(&pci->dev); + pm_runtime_dont_use_autosuspend(&pci->dev); + } + chip->running = 0; azx_del_card_list(chip); @@ -2322,6 +2328,8 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); pm_runtime_allow(&pci->dev); + pm_runtime_set_active(&pci->dev); + pm_runtime_enable(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.31.1

1 0

[PATCH] ALSA: hda: fix general protection fault in azx_runtime_idle
by Kai Vehmanen 15 Nov '21

15 Nov '21

Fix a corner case between PCI device driver remove callback and runtime PM idle callback. Following sequence of events can happen: - at azx_create, context is allocated with devm_kzalloc() and stored as pci_set_drvdata() - user-space requests to unbind audio driver - dd.c:__device_release_driver() calls PCI remove - pci-driver.c:pci_device_remove() calls the audio driver azx_remove() callback and this is completed - pci-driver.c:pm_runtime_put_sync() leads to a call to rpm_idle() which again calls azx_runtime_idle() - the azx context object, as returned by dev_get_drvdata(), is no longer valid -> access fault in azx_runtime_idle when executing struct snd_card *card = dev_get_drvdata(dev); chip = card->private_data; if (chip->disabled || hda->init_failed) This was discovered by i915_module_load test with 5.15.0 based linux-next tree. Example log caught by i915_module_load test with linux-next https://intel-gfx-ci.01.org/tree/linux-next/ <4> [264.038232] general protection fault, probably for non-canonical address 0x6b6b6b6b6b6b73f0: 0000 [#1] PREEMPT SMP NOPTI <4> [264.038248] CPU: 0 PID: 5374 Comm: i915_module_loa Not tainted 5.15.0-next-20211109-gc8109c2ba35e-next-20211109 #1 [...] <4> [264.038267] RIP: 0010:azx_runtime_idle+0x12/0x60 [snd_hda_intel] [...] <4> [264.038355] Call Trace: <4> [264.038359] <TASK> <4> [264.038362] __rpm_callback+0x3d/0x110 <4> [264.038371] rpm_idle+0x27f/0x380 <4> [264.038376] __pm_runtime_idle+0x3b/0x100 <4> [264.038382] pci_device_remove+0x6d/0xa0 <4> [264.038388] device_release_driver_internal+0xef/0x1e0 <4> [264.038395] unbind_store+0xeb/0x120 <4> [264.038400] kernfs_fop_write_iter+0x11a/0x1c0 Fix the issue by setting drvdata to NULL at end of azx_remove(). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) Some non-persistent direct links showing the bug trigger on different platforms with linux-next 20211109: - https://intel-gfx-ci.01.org/tree/linux-next/next-20211109/fi-tgl-1115g4/igt… - https://intel-gfx-ci.01.org/tree/linux-next/next-20211109/fi-jsl-1/igt@i915… Notably with 20211110 linux-next, the bug does not trigger: - https://intel-gfx-ci.01.org/tree/linux-next/next-20211110/fi-tgl-1115g4/igt… diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7762718cf429..b90c817e3f6f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2364,6 +2364,7 @@ static void azx_remove(struct pci_dev *pci) cancel_delayed_work_sync(&hda->probe_work); device_lock(&pci->dev); + pci_set_drvdata(pci, NULL); snd_card_free(card); } } base-commit: 6322ec8d0de924cf9672b23c1b5052afafc2f03b -- 2.33.0

2 7

ucm2: Add UCM support for Dell Desktop
by GitHub pull_request - opened 15 Nov '21

15 Nov '21

alsa-project/alsa-ucm-conf pull request #120 was opened from shumingfan: The Dell Desktop uses two USB audio codecs as front and rear panel. The front UAC has two outputs of speaker/headphone and one input of headset mic. The rear UAC has one output of line-out. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/120 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/120.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

HP Probook Mute volume + mic LED indicators not working (Audio codec = ALC236)
by GitHub issues - opened 14 Nov '21

14 Nov '21

alsa-project/alsa-lib issue #190 was opened from antintin0: Tested on kernel 5.13 and 5.15.2 Not sure if this is the right place to ask, but no one else seems to be able to help. I have an HP Probook x360 435 g7 with the Realtek ALC236 codec and my mute volume + mute mic LED indicators aren't working (they do on windows). I found there were some patches mentioning mute LEDs on ALC236 on the below links: https://patchwork.kernel.org/project/alsa-devel/patch/20210701091417.9696-1… https://mailman.alsa-project.org/pipermail/alsa-devel/2021-November/192225.… But obviously they're not working for me, so I tried following this: https://wiki.archlinux.org/title/Laptop#Audio_mute_LED Theres nothing specific for ALC236 but I tried pretty much all the relevant parameters for the snd-hda-intel module option "model" under "ALC22x/23x/25x/269/27x/28x/29x (and vendor-specific ALC3xxx models)" and none worked. Any help would be appreciated Issue URL : https://github.com/alsa-project/alsa-lib/issues/190 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ALSA: hda/realtek: Add quirk for ASRock NUC Box 1100
by Werner Sembach 14 Nov '21

14 Nov '21

This applies a SND_PCI_QUIRK(...) to the ASRock NUC Box 1100 series. This fixes the issue of the headphone jack not being detected unless warm rebooted from a certain other OS. When booting a certain other OS some coeff settings are changed that enable the audio jack. These settings are preserved on a warm reboot and can be easily dumped. The relevant indexes and values where gathered by naively diff-ing and reading a working and a non-working coeff dump. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- sound/pci/hda/patch_realtek.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f1727faec69..701c80ed83dc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6521,6 +6521,27 @@ static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *co alc_write_coef_idx(codec, 0x45, 0x5089); } +static const struct coef_fw alc233_fixup_no_audio_jack_coefs[] = { + WRITE_COEF(0x1a, 0x9003), WRITE_COEF(0x1b, 0x0e2b), WRITE_COEF(0x37, 0xfe06), + WRITE_COEF(0x38, 0x4981), WRITE_COEF(0x45, 0xd489), WRITE_COEF(0x46, 0x0074), + WRITE_COEF(0x49, 0x0149), + {} +}; + +static void alc233_fixup_no_audio_jack(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* + * The audio jack input and output is not detected on the ASRock NUC Box + * 1100 series when cold booting without this fix. Warm rebooting from a + * certain other OS makes the audio functional, as COEF settings are + * preserved in this case. This fix sets these altered COEF values as + * the default. + */ + alc_process_coef_fw(codec, alc233_fixup_no_audio_jack_coefs); +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6740,6 +6761,7 @@ enum { ALC287_FIXUP_13S_GEN2_SPEAKERS, ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, + ALC233_FIXUP_NO_AUDIO_JACK, }; static const struct hda_fixup alc269_fixups[] = { @@ -8460,6 +8482,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, }, + [ALC233_FIXUP_NO_AUDIO_JACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc233_fixup_no_audio_jack, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8894,6 +8920,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), -- 2.25.1

2 1

Possible issue with Intel 200 PCH HD on X299 AORUS Gaming 7
by Ramon Fried 13 Nov '21

13 Nov '21

Hi all. Apparently on Linux Headphones and Line are switched. Dual booting Windows on the same machine proves that it's indeed connected correctly. I initially thought it was related to Pulseaudio, but together with Pulseaudio guys help we figured it's probably in ALSA. http://alsa-project.org/db/?f=535021978814678ea328b0d3a053ba3cbd39b709 Where should I open a bug ? Is it kernel related, if user-space, to which ALSA component should I open a bug to ? Thanks, Ramon

2 2

[PATCH v3 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 12 Nov '21

12 Nov '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..c4fed6335230 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..59ed510e6cf1 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[PATCH] ASoC: mediatek: mt8173: Fix debugfs registration for components
by AngeloGioacchino Del Regno 12 Nov '21

12 Nov '21

When registering the mt8173-afe-pcm driver, we are also adding two components: one is for the PCM DAIs and one is for the HDMI DAIs, but when debugfs is enabled, we're getting the following issue: [ 17.279176] debugfs: Directory '11220000.audio-controller' with parent 'mtk-rt5650' already present! [ 17.288345] debugfs: Directory '11220000.audio-controller' with parent 'mtk-rt5650' already present! To overcome to that without any potentially big rewrite of this driver, similarly to what was done in mt8195-afe-pcm, add a debugfs_prefix to the components before actually adding them. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> --- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 51 ++++++++++++++++++---- 1 file changed, 43 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index 6350390414d4..31494930433f 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1054,6 +1054,7 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) int irq_id; struct mtk_base_afe *afe; struct mt8173_afe_private *afe_priv; + struct snd_soc_component *comp_pcm, *comp_hdmi; ret = dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(33)); if (ret) @@ -1142,23 +1143,55 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) if (ret) goto err_pm_disable; - ret = devm_snd_soc_register_component(&pdev->dev, - &mt8173_afe_pcm_dai_component, - mt8173_afe_pcm_dais, - ARRAY_SIZE(mt8173_afe_pcm_dais)); + comp_pcm = devm_kzalloc(&pdev->dev, sizeof(*comp_pcm), GFP_KERNEL); + if (!comp_pcm) { + ret = -ENOMEM; + goto err_pm_disable; + } + + ret = snd_soc_component_initialize(comp_pcm, + &mt8173_afe_pcm_dai_component, + &pdev->dev); if (ret) goto err_pm_disable; - ret = devm_snd_soc_register_component(&pdev->dev, - &mt8173_afe_hdmi_dai_component, - mt8173_afe_hdmi_dais, - ARRAY_SIZE(mt8173_afe_hdmi_dais)); +#ifdef CONFIG_DEBUG_FS + comp_pcm->debugfs_prefix = "pcm"; +#endif + + ret = snd_soc_add_component(comp_pcm, + mt8173_afe_pcm_dais, + ARRAY_SIZE(mt8173_afe_pcm_dais)); + if (ret) + goto err_pm_disable; + + comp_hdmi = devm_kzalloc(&pdev->dev, sizeof(*comp_hdmi), GFP_KERNEL); + if (!comp_hdmi) { + ret = -ENOMEM; + goto err_pm_disable; + } + + ret = snd_soc_component_initialize(comp_hdmi, + &mt8173_afe_hdmi_dai_component, + &pdev->dev); if (ret) goto err_pm_disable; +#ifdef CONFIG_DEBUG_FS + comp_hdmi->debugfs_prefix = "hdmi"; +#endif + + ret = snd_soc_add_component(comp_hdmi, + mt8173_afe_hdmi_dais, + ARRAY_SIZE(mt8173_afe_hdmi_dais)); + if (ret) + goto err_cleanup_components; + dev_info(&pdev->dev, "MT8173 AFE driver initialized.\n"); return 0; +err_cleanup_components: + snd_soc_unregister_component(&pdev->dev); err_pm_disable: pm_runtime_disable(&pdev->dev); return ret; @@ -1166,6 +1199,8 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) static int mt8173_afe_pcm_dev_remove(struct platform_device *pdev) { + snd_soc_unregister_component(&pdev->dev); + pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) mt8173_afe_runtime_suspend(&pdev->dev); -- 2.33.1

2 1

[PATCH] ASoC: SOF: build compression interface into snd_sof.ko
by Arnd Bergmann 12 Nov '21

12 Nov '21

From: Arnd Bergmann <arnd(a)arndb.de> With CONFIG_SND_SOC_SOF_COMPRESS=m, the compression code is not built into a the main SOF driver when that is built-in: x86_64-linux-ld: sound/soc/sof/ipc.o: in function `ipc_stream_message': ipc.c:(.text+0x5a2): undefined reference to `snd_sof_compr_fragment_elapsed' x86_64-linux-ld: sound/soc/sof/topology.o: in function `sof_dai_load': topology.c:(.text+0x32d1): undefined reference to `snd_sof_compr_init_elapsed_work' x86_64-linux-ld: topology.c:(.text+0x32e1): undefined reference to `snd_sof_compr_init_elapsed_work' Make this a 'bool' symbol so it just decides whether the code gets built at all. Fixes: 858f7a5c45ca ("ASoC: SOF: Introduce fragment elapsed notification API") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sof/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 6bb4db87af03..041c54639c4d 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -47,7 +47,7 @@ config SND_SOC_SOF_OF Say Y if you need this option. If unsure select "N". config SND_SOC_SOF_COMPRESS - tristate + bool select SND_SOC_COMPRESS config SND_SOC_SOF_DEBUG_PROBES -- 2.29.2

4 7

[PATCH 1/2] ASoC: rt5682: Avoid the unexpected IRQ event during going to suspend
by derek.fang＠realtek.com 12 Nov '21

12 Nov '21

From: Derek Fang <derek.fang(a)realtek.com> When the system suspends, the codec driver will set SAR to power saving mode if a headset is plugged in. There is a chance to generate an unexpected IRQ, and leads to issues after resuming such as noise from OMTP type headsets. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 78b4cb5fb6c8..a486ac268c33 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -48,6 +48,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_SAR_IL_CMD_6, 0x0110}, {RT5682_CHARGE_PUMP_1, 0x0210}, {RT5682_HP_LOGIC_CTRL_2, 0x0007}, + {RT5682_SAR_IL_CMD_2, 0xac00}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) @@ -2943,9 +2944,6 @@ static int rt5682_suspend(struct snd_soc_component *component) cancel_delayed_work_sync(&rt5682->jack_detect_work); cancel_delayed_work_sync(&rt5682->jd_check_work); if (rt5682->hs_jack && rt5682->jack_type == SND_JACK_HEADSET) { - snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, - RT5682_MB1_PATH_MASK | RT5682_MB2_PATH_MASK, - RT5682_CTRL_MB1_REG | RT5682_CTRL_MB2_REG); val = snd_soc_component_read(component, RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK; @@ -2967,10 +2965,15 @@ static int rt5682_suspend(struct snd_soc_component *component) /* enter SAR ADC power saving mode */ snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | - RT5682_SAR_BUTDET_RST_MASK | RT5682_SAR_SEL_MB1_MB2_MASK, 0); + RT5682_SAR_SEL_MB1_MB2_MASK, 0); + usleep_range(5000, 6000); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_MB1_PATH_MASK | RT5682_MB2_PATH_MASK, + RT5682_CTRL_MB1_REG | RT5682_CTRL_MB2_REG); + usleep_range(10000, 12000); snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, - RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_BUTDET_RST_MASK, - RT5682_SAR_BUTT_DET_EN | RT5682_SAR_BUTDET_POW_SAV | RT5682_SAR_BUTDET_RST_NORMAL); + RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK, + RT5682_SAR_BUTT_DET_EN | RT5682_SAR_BUTDET_POW_SAV); } regcache_cache_only(rt5682->regmap, true); -- 2.17.1

2 2

[PATCH v2 0/3] ASoC: rt9120: Fix settings and make compatible with rt9120s
by cy_huang 12 Nov '21

12 Nov '21

From: ChiYuan Huang <cy_huang(a)richtek.com> This patch series add the below changes - Fix the wrong ocp level setting. - Fix clock auto sync issue. - Make the driver compatible with rt9120s ChiYuan Huang (3): ASoC: rt9120: Update internal ocp level to the correct value ASoC: rt9120: Fix clock auto sync issue when fs is the multiple of 48 ASoC: rt9120: Add the compatibility with rt9120s sound/soc/codecs/rt9120.c | 58 +++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 48 insertions(+), 10 deletions(-) -- 2.7.4

2 4

[PATCH] ASoC: rsnd: fixup DMAEngine API
by Kuninori Morimoto 12 Nov '21

12 Nov '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit d5bb69dc54ec1 ("ASoC: sh: rcar: dma: : use proper DMAENGINE API for termination") updated DMAEngine API _all() to _sync(), but it should be _async(). _all() and _async() are almost same, the difference is only return error code. _sync() will call dmaengine_synchronize() and will be kernel panic. This patch is needed for v5.15 or later. [ 27.293264] BUG: scheduling while atomic: irq/130-ec70000/131/0x00000003 [ 27.300084] 2 locks held by irq/130-ec70000/131: [ 27.304743] #0: ffff0004c274d908 (&group->lock){....}-{2:2}, at: _snd_pcm_stream_lock_irqsave+0x48/0x54 [ 27.314344] #1: ffff0004c1788c60 (&priv->lock#2){....}-{2:2}, at: rsnd_soc_dai_trigger+0x70/0x7bc [ 27.323409] irq event stamp: 206 [ 27.326664] hardirqs last enabled at (205): [<ffff80001082de50>] _raw_spin_unlock_irq+0x50/0xa0 [ 27.335529] hardirqs last disabled at (206): [<ffff80001082d9e4>] _raw_spin_lock_irqsave+0xc4/0xd0 [ 27.344564] softirqs last enabled at (0): [<ffff800010037324>] copy_process+0x644/0x1b10 [ 27.352819] softirqs last disabled at (0): [<0000000000000000>] 0x0 [ 27.359142] CPU: 0 PID: 131 Comm: irq/130-ec70000 Not tainted 5.14.0-rc1+ #918 [ 27.366429] Hardware name: Renesas H3ULCB Kingfisher board based on r8a77950 (DT) [ 27.373975] Call trace: [ 27.376442] dump_backtrace+0x0/0x1b4 [ 27.380141] show_stack+0x24/0x30 [ 27.383488] dump_stack_lvl+0x8c/0xb8 [ 27.387184] dump_stack+0x18/0x34 [ 27.390528] __schedule_bug+0x8c/0x9c [ 27.394224] __schedule+0x790/0x8dc [ 27.397746] schedule+0x7c/0x110 [ 27.401003] synchronize_irq+0x94/0xd0 [ 27.404786] rcar_dmac_device_synchronize+0x20/0x2c [ 27.409710] rsnd_dmaen_stop+0x50/0x64 [ 27.413495] rsnd_soc_dai_trigger+0x554/0x7bc [ 27.417890] snd_soc_pcm_dai_trigger+0xe8/0x264 Cc: <stable(a)kernel.org> Fixes: commit d5bb69dc54ec1 ("ASoC: sh: rcar: dma: : use proper DMAENGINE API for termination") Link: https://lore.kernel.org/r/TY2PR01MB3692889E1A7476C4322CC296D8AE9@TY2PR01MB3… Reported-by: Yoshihiro Shimoda <yoshihiro.shimoda.uh(a)renesas.com> Acked-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 16c6e0265749..03e0d4eca781 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -102,7 +102,7 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); if (dmaen->chan) - dmaengine_terminate_sync(dmaen->chan); + dmaengine_terminate_async(dmaen->chan); return 0; } -- 2.25.1

2 1

[PATCH v5] dt-bindings: dsp: mediatek: Add mt8195 DSP binding support
by YC Hung 12 Nov '21

12 Nov '21

From: "yc.hung" <yc.hung(a)mediatek.com> This describes the mt8195 DSP device tree node. Signed-off-by: yc.hung <yc.hung(a)mediatek.com> --- Changes since v4: - remove patch 1 with clocks as it will be reviewed and merged to SOF tree. https://github.com/thesofproject/linux/pull/3237 - refine DT file binding files with Rob's comments. .../bindings/dsp/mtk,mt8195-dsp.yaml | 142 ++++++++++++++++++ 1 file changed, 142 insertions(+) create mode 100644 Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml diff --git a/Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml b/Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml new file mode 100644 index 000000000000..aeeb7af69625 --- /dev/null +++ b/Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml @@ -0,0 +1,142 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/dsp/mtk,mt8195-dsp.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek mt8195 DSP core + +maintainers: + - YC Hung <yc.hung(a)mediatek.com> + +description: | + Some boards from mt8195 contain a DSP core used for + advanced pre- and post- audio processing. +properties: + compatible: + const: mediatek,mt8195-dsp + + reg: + items: + - description: Address and size of the DSP Cfg registers + - description: Address and size of the DSP SRAM + + reg-names: + items: + - const: cfg + - const: sram + + interrupts: + items: + - description: watchdog interrupt + + interrupt-names: + items: + - const: wdt + + clocks: + items: + - description: mux for audio dsp clock + - description: 26M clock + - description: mux for audio dsp local bus + - description: default audio dsp local bus clock source + - description: clock gate for audio dsp clock + - description: mux for audio dsp access external bus + + clock-names: + items: + - const: adsp_sel + - const: clk26m_ck + - const: audio_local_bus + - const: mainpll_d7_d2 + - const: scp_adsp_audiodsp + - const: audio_h + + power-domains: + maxItems: 1 + + mboxes: + items: + - description: a mailbox is ised for ipc reply between host and audio DSP. + - description: a mailbox is ised for ipc reuqest between host and audio DSP. + + mbox-names: + items: + - const: mbox0 + - const: mbox1 + + memory-region: + items: + - description: A phandle to a reserved memory region is used for dma buffer between host and DSP. + - description: A phandle to a reserved memory region is used for DSP system memory. + + sound: + description: + Sound subnode includes ASoC platform, DPTx codec node, and + HDMI codec node. + + type: object + + properties: + mediatek,platform: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of MT8195 ASoC platform. + + mediatek,dptx-codec: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of MT8195 Display Port Tx codec node. + + mediatek,hdmi-codec: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of MT8195 HDMI codec node. + +required: + - compatible + - reg + - reg-names + - interrupts + - interrupt-names + - clocks + - clock-names + - memory-region + - power-domains + - mbox-names + - mboxes + - sound + +additionalProperties: true + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + adsp: dsp@10803000 { + compatible = "mediatek,mt8195-dsp"; + reg = <0x10803000 0x1000>, + <0x10840000 0x40000>; + reg-names = "cfg", "sram"; + interrupts = <GIC_SPI 694 IRQ_TYPE_LEVEL_HIGH 0>; + interrupt-names = "wdt"; + clocks = <&topckgen 10>, //CLK_TOP_ADSP + <&clk26m>, + <&topckgen 107>, //CLK_TOP_AUDIO_LOCAL_BUS + <&topckgen 136>, //CLK_TOP_MAINPLL_D7_D2 + <&scp_adsp 0>, //CLK_SCP_ADSP_AUDIODSP + <&topckgen 34>; //CLK_TOP_AUDIO_H + clock-names = "adsp_sel", + "clk26m_ck", + "audio_local_bus", + "mainpll_d7_d2", + "scp_adsp_audiodsp", + "audio_h"; + memory-region = <&adsp_dma_mem_reserved>, + <&adsp_mem_reserved>; + power-domains = <&spm 6>; //MT8195_POWER_DOMAIN_ADSP + mbox-names = "mbox0", "mbox1"; + mboxes = <&adsp_mailbox 0>, <&adsp_mailbox 1>; + sound { + mediatek,dptx-codec = <&dp_tx>; + mediatek,hdmi-codec = <&hdmi0>; + mediatek,platform = <&afe>; + }; + }; -- 2.18.0

2 1

Unable to connect KEF X300A USB speakers
by GitHub issues - edited 12 Nov '21

12 Nov '21

alsa-project/alsa-lib issue #189 was edited from simbatl: I'm trying to use a pair of KEF X300A speakers. They work with aplay on Ubuntu 20.04 server, but fail on Ubuntu 18.04, 20.04 and 21.10 desktop. Here is output from pulseaudio -vvv: [pulse.log](https://github.com/alsa-project/alsa-lib/files/7527770/pulse.log) And here is output from alsa-info.sh: [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/7527773/alsa-… Anyone who know how to solve this issue? :) Issue URL : https://github.com/alsa-project/alsa-lib/issues/189 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Unable to connect KEF X300A USB speakers
by GitHub issues - opened 12 Nov '21

12 Nov '21

alsa-project/alsa-lib issue #189 was opened from simbatl: I'm trying to USE a pair of KEF X300A speakers. They work with aplay on Ubuntu 20.04 server, but fail on Ubuntu 18.04, 20.04 and 21.10 desktop. Here is output from pulseaudio -vvv: [pulse.log](https://github.com/alsa-project/alsa-lib/files/7527770/pulse.log) And here is output from alsa-info.sh: [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/7527773/alsa-… Anyone who know how to solve this issue? :) Issue URL : https://github.com/alsa-project/alsa-lib/issues/189 Repository URL: https://github.com/alsa-project/alsa-lib

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Issues again with Comet Lake PCH cAVS and pipewire
by GitHub issues - opened 12 Nov '21

12 Nov '21

alsa-project/alsa-lib issue #188 was opened from leiserfg: At a certain point after v1.2.5-65-gf0d540f8 pipewire stopped showing any device for the cards Comet Lake PCH cAVS. Issue URL : https://github.com/alsa-project/alsa-lib/issues/188 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ALSA: hda/realtek: Add quirk for ASRock NUC Box 1100
by Werner Sembach 12 Nov '21

12 Nov '21

This applies a SND_PCI_QUIRK(...) to the ASRock NUC Box 1100 series. This fixes the issue of the headphone jack not being detected unless warm rebooted from a certain other OS. When booting a certain other OS some coeff settings are changed that enable the audio jack. These settings are preserved on a warm reboot and can be easily dumped. The relevant indexes and values where gathered by naively diff-ing and reading a working and a non-working coeff dump. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_realtek.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f1727faec69..2aef9866f170 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6521,6 +6521,25 @@ static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *co alc_write_coef_idx(codec, 0x45, 0x5089); } +static void alc233_fixup_asrock_nuc_box_1100_no_audio_jack(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* + * The audio jack input and output is not detected on the ASRock NUC Box 1100 series when + * cold booting without this fix. Warm rebooting from a certain other OS makes the audio + * functional, as COEF settings are preserved in this case. This fix sets these altered + * COEF values as the default. + */ + alc_write_coef_idx(codec, 0x1a, 0x9003); + alc_write_coef_idx(codec, 0x1b, 0x0e2b); + alc_write_coef_idx(codec, 0x37, 0xfe06); + alc_write_coef_idx(codec, 0x38, 0x4981); + alc_write_coef_idx(codec, 0x45, 0xd489); + alc_write_coef_idx(codec, 0x46, 0x0074); + alc_write_coef_idx(codec, 0x49, 0x0149); +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6740,6 +6759,7 @@ enum { ALC287_FIXUP_13S_GEN2_SPEAKERS, ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, + ALC233_FIXUP_ASROCK_NUC_BOX_1100_NO_AUDIO_JACK, }; static const struct hda_fixup alc269_fixups[] = { @@ -8460,6 +8480,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, }, + [ALC233_FIXUP_ASROCK_NUC_BOX_1100_NO_AUDIO_JACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc233_fixup_asrock_nuc_box_1100_no_audio_jack, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8894,6 +8918,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), + SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_ASROCK_NUC_BOX_1100_NO_AUDIO_JACK), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), -- 2.25.1

2 2

[PATCH] ALSA: cmipci: Drop stale variable assignment
by Takashi Iwai 12 Nov '21

12 Nov '21

Since the recent code refactoring using devres, the variable cm in snd_cmipci_probe() is no longer referred. Fixes: 87e082ad84a7 ("ALSA: cmipci: Allocate resources with device-managed APIs") Reported-by: kernel test robot <lkp(a)intel.com> Link: https://lore.kernel.org/r/cc6383a2-cafb-ffe7-0b4f-27a310a1005c@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/cmipci.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index ea20236f35db..9a678b5cf285 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3218,7 +3218,6 @@ static int snd_cmipci_probe(struct pci_dev *pci, { static int dev; struct snd_card *card; - struct cmipci *cm; int err; if (dev >= SNDRV_CARDS) @@ -3229,10 +3228,9 @@ static int snd_cmipci_probe(struct pci_dev *pci, } err = snd_devm_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, - sizeof(*cm), &card); + sizeof(struct cmipci), &card); if (err < 0) return err; - cm = card->private_data; switch (pci->device) { case PCI_DEVICE_ID_CMEDIA_CM8738: -- 2.31.1

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Re: [PATCH v3 2/5] dt-bindings: mfd: timers: Update maintainers for st,stm32-timers
by Lee Jones 12 Nov '21

12 Nov '21

On Wed, 10 Nov 2021, Rob Herring wrote: > On Wed, 10 Nov 2021 16:01:41 +0100, patrice.chotard(a)foss.st.com wrote: > > From: Patrice Chotard <patrice.chotard(a)foss.st.com> > > > > Benjamin has left the company, remove his name from maintainers. > > > > Signed-off-by: Patrice Chotard <patrice.chotard(a)foss.st.com> > > --- > > Documentation/devicetree/bindings/mfd/st,stm32-timers.yaml | 1 - > > 1 file changed, 1 deletion(-) > > > > Lee indicated he was going to pick this one up, so: > > Acked-by: Rob Herring <robh(a)kernel.org> Since you already merged the treewide patch, you may as well take this too. We'll work through any conflicts that may occur as a result. Acked-by: Lee Jones <lee.jones(a)linaro.org> -- Lee Jones [李琼斯] Senior Technical Lead - Developer Services Linaro.org │ Open source software for Arm SoCs Follow Linaro: Facebook | Twitter | Blog

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UCM can't match the USB device ID in Ubuntu 20.04
by GitHub issues - opened 12 Nov '21

12 Nov '21

alsa-project/alsa-ucm-conf issue #119 was opened from shumingfan: We created the UCM2 conf files and used the 'components' string to match the right UCM config. Unfortunately, the system can't apply the right UCM config. The test files and alsa-lib version shows below. USB-Audio.conf https://github.com/perexg/alsa-ucm-conf/commit/e3517b1cbb4ac0c69496329eeb15… Dell-Desktop UCM2 files https://github.com/shumingfan/alsa-ucm-conf/commit/16dce692843b05d9cd720dda… $ alsaucm --version alsaucm: version 1.2.2 Is it possible to resolve this problem when using alsa-lib-1.2.2? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/119 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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speaker-test fails with message "Unknown field hint" on Scarlett 8i6
by GitHub issues - opened 11 Nov '21

11 Nov '21

alsa-project/alsa-plugins issue #40 was opened from montemedio: I am using a Scarlett 8i6 Gen1. It used to work fine, but with the newest version of `speaker-test`, I get this error message: ``` $ speaker-test -Dusbstream:CARD=USB speaker-test 1.2.5.1 Playback device is usbstream:CARD=USB Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm_usb_stream.c:503:(_snd_pcm_usb_stream_open) Unknown field hint Playback open error: -22,Invalid argument ``` ``` $ cat /proc/asound/cards 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xd3a10000 irq 35 1 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xd3000000 irq 17 2 [USB ]: USB-Audio - Scarlett 8i6 USB Focusrite Scarlett 8i6 USB at usb-0000:00:1a.0-1.2, high speed ``` Issue URL : https://github.com/alsa-project/alsa-plugins/issues/40 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: fireworks: add support for Loud Onyx 1200f quirk
by Takashi Sakamoto 11 Nov '21

11 Nov '21

Loud Technologies shipped Onyx 1200f 2008 in its Mackie brand and already discontinued. The model uses component of Fireworks board module as its communication and DSP function. The latest firmware (v4.6.0) has a quirk that tx packet includes wrong value (0x1f) in its dbs field at middle and higher sampling transfer frequency. It brings ALSA fireworks driver discontinuity of data block counter. This commit fixes it by assuming it as a quirk of firmware version 4.6.0. $ cd linux-firewire-tools/src $ python crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 0404b9ef bus_info_length 4, crc_length 4, crc 47599 404 31333934 bus_name "1394" 408 e064a212 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 100, max_rec 10 (2048), max_rom 2, gen 1, spd 2 (S400) 40c 000ff209 company_id 000ff2 | 410 62550ce0 device_id 0962550ce0 | EUI-64 000ff20962550ce0 root directory ----------------------------------------------------------------- 414 0008088e directory_length 8, crc 2190 418 03000ff2 vendor 41c 8100000f --> descriptor leaf at 458 420 1701200f model 424 81000018 --> descriptor leaf at 484 428 0c008380 node capabilities 42c 8d000003 --> eui-64 leaf at 438 430 d1000005 --> unit directory at 444 434 08000ff2 (immediate value) eui-64 leaf at 438 ----------------------------------------------------------------- 438 000281ae leaf_length 2, crc 33198 43c 000ff209 company_id 000ff2 | 440 62550ce0 device_id 0962550ce0 | EUI-64 000ff20962550ce0 unit directory at 444 ----------------------------------------------------------------- 444 00045d94 directory_length 4, crc 23956 448 1200a02d specifier id: 1394 TA 44c 13010000 version 450 1701200f model 454 8100000c --> descriptor leaf at 484 descriptor leaf at 458 ----------------------------------------------------------------- 458 000a199d leaf_length 10, crc 6557 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 4d61636b "Mack" 468 69650000 "ie" 46c 00000000 470 00000000 474 00000000 478 00000000 47c 00000000 480 00000000 descriptor leaf at 484 ----------------------------------------------------------------- 484 000a0964 leaf_length 10, crc 2404 488 00000000 textual descriptor 48c 00000000 minimal ASCII 490 4f6e7978 "Onyx" 494 20313230 " 120" 498 30460000 "0F" 49c 00000000 4a0 00000000 4a4 00000000 4a8 00000000 4ac 00000000 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireworks/fireworks_stream.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index ac66f08acd6b..53dbd4d4b0d0 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -50,8 +50,9 @@ static int init_stream(struct snd_efw *efw, struct amdtp_stream *stream) efw->firmware_version == 0x5070300 || efw->firmware_version == 0x5080000)) efw->tx_stream.flags |= CIP_UNALIGHED_DBC; - // AudioFire9 always reports wrong dbs. - if (efw->is_af9) + // AudioFire9 always reports wrong dbs. Onyx 1200F with the latest firmware (v4.6.0) + // also report wrong dbs at 88.2 kHz or greater. + if (efw->is_af9 || efw->firmware_version == 0x4060000) efw->tx_stream.flags |= CIP_WRONG_DBS; // Firmware version 5.5 reports fixed interval for dbc. if (efw->firmware_version == 0x5050000) -- 2.32.0

2 1

[PATCH] ALSA: usb-audio: add the profile name of Dell desktop
by shumingf＠realtek.com 11 Nov '21

11 Nov '21

From: Shuming Fan <shumingf(a)realtek.com> Add the profile name to let userspace pick correct UCM profile Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/usb/card.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 1764b9302d46..9c67ecbe710e 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -398,6 +398,10 @@ static const struct usb_audio_device_name usb_audio_names[] = { PROFILE_NAME(0x0bda, 0x4014, "Dell", "WD15 Dock", "Dell-WD15-Dock"), /* Dell WD19 Dock */ PROFILE_NAME(0x0bda, 0x402e, "Dell", "WD19 Dock", "Dell-WD15-Dock"), + /* Dell Desktop - Front */ + PROFILE_NAME(0x0bda, 0x4c54, "Dell", "Desktop", "Dell-Desktop-Front"), + /* Dell Desktop - Rear */ + PROFILE_NAME(0x0bda, 0x4c55, "Dell", "Desktop", "Dell-Desktop-Rear"), DEVICE_NAME(0x0ccd, 0x0028, "TerraTec", "Aureon5.1MkII"), -- 2.33.0

5 8

[PATCH] ALSA: hda: Free card instance properly at probe errors
by Takashi Iwai 10 Nov '21

10 Nov '21

The recent change in hda-intel driver to allow repeated probes surfaced a problem that has been hidden until; the probe process in the work calls azx_free() at the error path, and this skips the card free process that eventually releases codec instances. As a result, we get a kernel WARNING like: snd_hda_intel 0000:00:1f.3: Cannot probe codecs, giving up ------------[ cut here ]------------ WARNING: CPU: 14 PID: 186 at sound/hda/hdac_bus.c:73 .... For fixing this, we need to call snd_card_free() instead of azx_free(). Additionally, the device drvdata has to be cleared, as the driver binding itself is still active. Then the PM and other driver callbacks will ignore the procedure. Fixes: c0f1886de7e1 ("ALSA: hda: intel: Allow repeatedly probing on codec configuration errors") Reported-and-tested-by: Scott Branden <scott.branden(a)broadcom.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/063e2397-7edb-5f48-7b0d-618b938d9dd8@broadcom.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7762718cf429..90e9263ac0bd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2327,7 +2327,8 @@ static int azx_probe_continue(struct azx *chip) out_free: if (err < 0) { - azx_free(chip); + pci_set_drvdata(pci, NULL); + snd_card_free(chip->card); return err; } -- 2.26.2

1 0

5.15 Linux Regression in sound hda
by Scott Branden 10 Nov '21

10 Nov '21

I'm reporting a new WARN_ON in sound/hda/hdac_bus.c that was not present in 5.15. Sorry, I don't have time to bisect this right now but report is below. dmesg in 5.14 kernel was: snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops 0xffffffffaa2ac520) GACT probability on Mirror/redirect action on Simple TC action Loaded u32 classifier Performance counters on input device check on Actions configured hdaudio hdaudioC0D0: Unable to bind the codec hdaudio hdaudioC0D2: Unable to bind the codec dmesg in 5.15 kernel produces WARN_ON in sound/hda/hdac_bus.c snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops 0xffffffffad8b02a0) GACT probability on Mirror/redirect action on Simple TC action Loaded u32 classifier Performance counters on input device check on Actions configured snd_hda_intel 0000:00:1f.3: Cannot probe codecs, giving up Initializing XFRM netlink socket ------------[ cut here ]------------ NET: Registered PF_INET6 protocol family WARNING: CPU: 14 PID: 186 at sound/hda/hdac_bus.c:73 snd_hdac_bus_exit+0x40/0x50 Segment Routing with IPv6 Modules linked in: CPU: 14 PID: 186 Comm: kworker/14:2 Tainted: G I 5.15.0 #1 Hardware name: ASUSTeK COMPUTER INC. System Product Name/WS C246 PRO, BIOS 1401 08/13/2020 In-situ OAM (IOAM) with IPv6 Workqueue: events azx_probe_work sit: IPv6, IPv4 and MPLS over IPv4 tunneling driver RIP: 0010:snd_hdac_bus_exit+0x40/0x50 NET: Registered PF_PACKET protocol family Code: 75 19 48 8b 57 58 48 8d 47 58 48 39 c2 75 1b 48 81 c7 f8 02 00 00 e9 ff 97 57 ff 0f 0b 48 8b 57 58 48 8d 47 58 48 39 c2 74 e5 <0f> 0b 48 81 c7 f8 02 00 00 e9 e2 97 57 ff 66 90 0f 1f 44 00 00 41 RPC: Registered rdma transport module. RSP: 0000:ffffa545007b7e88 EFLAGS: 00010202 RPC: Registered rdma backchannel transport module. Key type dns_resolver registered RAX: ffff89f483078880 RBX: ffff89f483078828 RCX: 0000000080200010 RDX: ffff89f4830792f0 RSI: 0000000000000001 RDI: ffff89f483078828 RBP: ffff89f483078e70 R08: 0000000000000000 R09: ffffffffacf094a4 R10: 0000000000000004 R11: 0000000000000032 R12: ffff89f7dbda50c0 R13: ffff89f7dbda9100 R14: 0000000000000000 R15: ffff89f7dbda9105 FS: 0000000000000000(0000) GS:ffff89f7dbd80000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000000 CR3: 0000000194a0c001 CR4: 00000000003706e0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: azx_free+0xba/0x170 process_one_work+0x1db/0x380 worker_thread+0x4d/0x3e0 ? rescuer_thread+0x370/0x370 kthread+0x124/0x150 ? set_kthread_struct+0x40/0x40 ret_from_fork+0x1f/0x30 ---[ end trace 9779b9e2ef53104c ]---

2 3

[PATCH] ALSA: hda/realtek: Add quirk for HP EliteBook 840 G7 mute LED
by Kai-Heng Feng 10 Nov '21

10 Nov '21

The mute and micmute LEDs don't work on HP EliteBook 840 G7. The same quirk for other HP laptops can let LEDs work, so apply it. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a6098c942aeb5..2f1727faec698 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8636,6 +8636,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8720, "HP EliteBook x360 1040 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8728, "HP EliteBook 840 G7", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.32.0

2 1

ucm2: Add support for SC7180 Trogdor Lazor Chromebooks
by GitHub pull_request - edited 10 Nov '21

10 Nov '21

alsa-project/alsa-ucm-conf pull request #118 was edited from kholk: Almost all Trogdor Chromebooks are using RT5682 for headphones/headset on 3.5mm jack, internal microphone and headset microphone, and MAX98357A for the internal speakers (Lazor and Limozeen). This adds support for the sc7180-rt5682-max98357a-1mic sound card found on the aforementioned Chromebooks. `Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com>` Tested on sc7180-trogdor-lazor-limozeen-nots. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/118 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/118.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Add support for SC7180 Trogdor Lazor Chromebooks
by GitHub pull_request - opened 10 Nov '21

10 Nov '21

alsa-project/alsa-ucm-conf pull request #118 was opened from kholk: Almost all Trogdor Chromebooks are using RT5682 for headphones/headset on 3.5mm jack, internal microphone and headset microphone, and MAX98357A for the internal speakers (Lazor and Limozeen). This adds support for the sc7180-rt5682-max98357a-1mic sound card found on the aforementioned Chromebooks. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Tested on sc7180-trogdor-lazor-limozeen-nots. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/118 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/118.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: memalloc: Remove a stale comment
by Takashi Iwai 10 Nov '21

10 Nov '21

The comment about the reused vmalloc helpers is no longer valid after the recent change for the noncontig allocator. Drop the stale comment. Fixes: ad4f93ca4138 ("ALSA: memalloc: Use proper SG helpers for noncontig allocations") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 8c362bb5b81a..9fc971a704a9 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -615,7 +615,6 @@ static const struct snd_malloc_ops snd_dma_noncontig_ops = { .free = snd_dma_noncontig_free, .mmap = snd_dma_noncontig_mmap, .sync = snd_dma_noncontig_sync, - /* re-use vmalloc helpers for get_* ops */ .get_addr = snd_dma_noncontig_get_addr, .get_page = snd_dma_noncontig_get_page, .get_chunk_size = snd_dma_noncontig_get_chunk_size, -- 2.26.2

1 0

[PATCH AUTOSEL 4.4 11/12] ALSA: gus: fix null pointer dereference on pointer block
by Sasha Levin 10 Nov '21

10 Nov '21

From: Chengfeng Ye <cyeaa(a)connect.ust.hk> [ Upstream commit a0d21bb3279476c777434c40d969ea88ca64f9aa ] The pointer block return from snd_gf1_dma_next_block could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> Link: https://lore.kernel.org/r/20211024104611.9919-1-cyeaa@connect.ust.hk Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/gus/gus_dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index 36c27c8323601..2e27cd3427c87 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -141,6 +141,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) } block = snd_gf1_dma_next_block(gus); spin_unlock(&gus->dma_lock); + if (!block) + return; snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 -- 2.33.0

1 0

[PATCH AUTOSEL 4.9 12/13] ALSA: gus: fix null pointer dereference on pointer block
by Sasha Levin 10 Nov '21

10 Nov '21

From: Chengfeng Ye <cyeaa(a)connect.ust.hk> [ Upstream commit a0d21bb3279476c777434c40d969ea88ca64f9aa ] The pointer block return from snd_gf1_dma_next_block could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> Link: https://lore.kernel.org/r/20211024104611.9919-1-cyeaa@connect.ust.hk Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/gus/gus_dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index 36c27c8323601..2e27cd3427c87 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -141,6 +141,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) } block = snd_gf1_dma_next_block(gus); spin_unlock(&gus->dma_lock); + if (!block) + return; snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 -- 2.33.0

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[PATCH AUTOSEL 4.14 13/14] ALSA: gus: fix null pointer dereference on pointer block
by Sasha Levin 10 Nov '21

10 Nov '21

From: Chengfeng Ye <cyeaa(a)connect.ust.hk> [ Upstream commit a0d21bb3279476c777434c40d969ea88ca64f9aa ] The pointer block return from snd_gf1_dma_next_block could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> Link: https://lore.kernel.org/r/20211024104611.9919-1-cyeaa@connect.ust.hk Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/gus/gus_dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index 36c27c8323601..2e27cd3427c87 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -141,6 +141,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) } block = snd_gf1_dma_next_block(gus); spin_unlock(&gus->dma_lock); + if (!block) + return; snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 -- 2.33.0

1 0

[PATCH AUTOSEL 4.19 20/21] ALSA: gus: fix null pointer dereference on pointer block
by Sasha Levin 10 Nov '21

10 Nov '21

From: Chengfeng Ye <cyeaa(a)connect.ust.hk> [ Upstream commit a0d21bb3279476c777434c40d969ea88ca64f9aa ] The pointer block return from snd_gf1_dma_next_block could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> Link: https://lore.kernel.org/r/20211024104611.9919-1-cyeaa@connect.ust.hk Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/gus/gus_dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index 7f95f452f1064..48e76b8fede41 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -141,6 +141,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) } block = snd_gf1_dma_next_block(gus); spin_unlock(&gus->dma_lock); + if (!block) + return; snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 -- 2.33.0

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[PATCH AUTOSEL 4.19 14/21] ALSA: ISA: not for M68K
by Sasha Levin 10 Nov '21

10 Nov '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 ] On m68k, compiling drivers under SND_ISA causes build errors: ../sound/core/isadma.c: In function 'snd_dma_program': ../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration] 33 | flags = claim_dma_lock(); | ^~~~~~~~~~~~~~ ../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration] 41 | release_dma_lock(flags); | ^~~~~~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare': ../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function) 253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); | ^~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare': ../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function) 322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); | ^~~~~~~~~~~~ and more... Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-m68k(a)lists.linux-m68k.org Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Link: https://lore.kernel.org/r/20211016062602.3588-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/Makefile | 2 ++ sound/isa/Kconfig | 2 +- sound/pci/Kconfig | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/Makefile b/sound/core/Makefile index ee4a4a6b99ba7..d123587c0fd8f 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index d7db1eeebc844..f8f3433925bb4 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -21,7 +21,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 4105d9f653d90..bbaf46dc3f804 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -278,6 +278,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. -- 2.33.0

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[PATCH AUTOSEL 4.19 09/21] ASoC: nau8824: Add DMI quirk mechanism for active-high jack-detect
by Sasha Levin 10 Nov '21

10 Nov '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 92d3360108f1839ca40451bad20ff67dd24a1964 ] Add a quirk mechanism to allow specifying that active-high jack-detection should be used on platforms where this info is not available in devicetree. And add an entry for the Cyberbook T116 tablet to the DMI table, so that jack-detection will work properly on this tablet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20211002211459.110124-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 40 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 663a208c2f784..4af87340b1655 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -11,6 +11,7 @@ #include <linux/module.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/init.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -30,6 +31,12 @@ #include "nau8824.h" +#define NAU8824_JD_ACTIVE_HIGH BIT(0) + +static int nau8824_quirk; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static int nau8824_config_sysclk(struct nau8824 *nau8824, int clk_id, unsigned int freq); @@ -1878,6 +1885,34 @@ static int nau8824_read_device_properties(struct device *dev, return 0; } +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id nau8824_quirk_table[] = { + { + /* Cyberbook T116 rugged tablet */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Default string"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "20170531"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + {} +}; + +static void nau8824_check_quirks(void) +{ + const struct dmi_system_id *dmi_id; + + if (quirk_override != -1) { + nau8824_quirk = quirk_override; + return; + } + + dmi_id = dmi_first_match(nau8824_quirk_table); + if (dmi_id) + nau8824_quirk = (unsigned long)dmi_id->driver_data; +} + static int nau8824_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1902,6 +1937,11 @@ static int nau8824_i2c_probe(struct i2c_client *i2c, nau8824->irq = i2c->irq; sema_init(&nau8824->jd_sem, 1); + nau8824_check_quirks(); + + if (nau8824_quirk & NAU8824_JD_ACTIVE_HIGH) + nau8824->jkdet_polarity = 0; + nau8824_print_device_properties(nau8824); ret = regmap_read(nau8824->regmap, NAU8824_REG_I2C_DEVICE_ID, &value); -- 2.33.0

1 0

[PATCH AUTOSEL 5.4 29/30] ALSA: gus: fix null pointer dereference on pointer block
by Sasha Levin 10 Nov '21

10 Nov '21

From: Chengfeng Ye <cyeaa(a)connect.ust.hk> [ Upstream commit a0d21bb3279476c777434c40d969ea88ca64f9aa ] The pointer block return from snd_gf1_dma_next_block could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> Link: https://lore.kernel.org/r/20211024104611.9919-1-cyeaa@connect.ust.hk Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/gus/gus_dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index a1c770d826dda..6d664dd8dde0b 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -126,6 +126,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) } block = snd_gf1_dma_next_block(gus); spin_unlock(&gus->dma_lock); + if (!block) + return; snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 -- 2.33.0

1 0

[PATCH AUTOSEL 5.4 22/30] ALSA: ISA: not for M68K
by Sasha Levin 10 Nov '21

10 Nov '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 ] On m68k, compiling drivers under SND_ISA causes build errors: ../sound/core/isadma.c: In function 'snd_dma_program': ../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration] 33 | flags = claim_dma_lock(); | ^~~~~~~~~~~~~~ ../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration] 41 | release_dma_lock(flags); | ^~~~~~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare': ../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function) 253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); | ^~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare': ../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function) 322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); | ^~~~~~~~~~~~ and more... Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-m68k(a)lists.linux-m68k.org Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Link: https://lore.kernel.org/r/20211016062602.3588-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/Makefile | 2 ++ sound/isa/Kconfig | 2 +- sound/pci/Kconfig | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/Makefile b/sound/core/Makefile index ee4a4a6b99ba7..d123587c0fd8f 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b690ed937cbe8..df2e45c8814e9 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -22,7 +22,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7630f808d087c..6edde2f145025 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -279,6 +279,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. -- 2.33.0

1 0

[PATCH AUTOSEL 5.4 15/30] ASoC: nau8824: Add DMI quirk mechanism for active-high jack-detect
by Sasha Levin 10 Nov '21

10 Nov '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 92d3360108f1839ca40451bad20ff67dd24a1964 ] Add a quirk mechanism to allow specifying that active-high jack-detection should be used on platforms where this info is not available in devicetree. And add an entry for the Cyberbook T116 tablet to the DMI table, so that jack-detection will work properly on this tablet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20211002211459.110124-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 40 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 15bd8335f6678..c8ccfa2fff848 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/init.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -27,6 +28,12 @@ #include "nau8824.h" +#define NAU8824_JD_ACTIVE_HIGH BIT(0) + +static int nau8824_quirk; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static int nau8824_config_sysclk(struct nau8824 *nau8824, int clk_id, unsigned int freq); @@ -1875,6 +1882,34 @@ static int nau8824_read_device_properties(struct device *dev, return 0; } +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id nau8824_quirk_table[] = { + { + /* Cyberbook T116 rugged tablet */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Default string"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "20170531"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + {} +}; + +static void nau8824_check_quirks(void) +{ + const struct dmi_system_id *dmi_id; + + if (quirk_override != -1) { + nau8824_quirk = quirk_override; + return; + } + + dmi_id = dmi_first_match(nau8824_quirk_table); + if (dmi_id) + nau8824_quirk = (unsigned long)dmi_id->driver_data; +} + static int nau8824_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1899,6 +1934,11 @@ static int nau8824_i2c_probe(struct i2c_client *i2c, nau8824->irq = i2c->irq; sema_init(&nau8824->jd_sem, 1); + nau8824_check_quirks(); + + if (nau8824_quirk & NAU8824_JD_ACTIVE_HIGH) + nau8824->jkdet_polarity = 0; + nau8824_print_device_properties(nau8824); ret = regmap_read(nau8824->regmap, NAU8824_REG_I2C_DEVICE_ID, &value); -- 2.33.0

1 0

[PATCH AUTOSEL 5.4 13/30] ASoC: SOF: Intel: hda-dai: fix potential locking issue
by Sasha Levin 10 Nov '21

10 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit a20f3b10de61add5e14b6ce4df982f4df2a4cbbc ] The initial hdac_stream code was adapted a third time with the same locking issues. Move the spin_lock outside the loops and make sure the fields are protected on read/write. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Acked-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210924192417.169243-5-pierre-louis.bossart@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-dai.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 3f645200d3a5c..b3cdd10c83ae1 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -67,6 +67,7 @@ static struct hdac_ext_stream * return NULL; } + spin_lock_irq(&bus->reg_lock); list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = stream_to_hdac_ext_stream(stream); @@ -106,12 +107,12 @@ static struct hdac_ext_stream * * is updated in snd_hdac_ext_stream_decouple(). */ if (!res->decoupled) - snd_hdac_ext_stream_decouple(bus, res, true); - spin_lock_irq(&bus->reg_lock); + snd_hdac_ext_stream_decouple_locked(bus, res, true); + res->link_locked = 1; res->link_substream = substream; - spin_unlock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); return res; } -- 2.33.0

1 0

[PATCH AUTOSEL 5.10 48/50] ALSA: gus: fix null pointer dereference on pointer block
by Sasha Levin 10 Nov '21

10 Nov '21

From: Chengfeng Ye <cyeaa(a)connect.ust.hk> [ Upstream commit a0d21bb3279476c777434c40d969ea88ca64f9aa ] The pointer block return from snd_gf1_dma_next_block could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> Link: https://lore.kernel.org/r/20211024104611.9919-1-cyeaa@connect.ust.hk Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/gus/gus_dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index a1c770d826dda..6d664dd8dde0b 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -126,6 +126,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) } block = snd_gf1_dma_next_block(gus); spin_unlock(&gus->dma_lock); + if (!block) + return; snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 -- 2.33.0

1 0

[PATCH AUTOSEL 5.10 37/50] ALSA: ISA: not for M68K
by Sasha Levin 10 Nov '21

10 Nov '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 ] On m68k, compiling drivers under SND_ISA causes build errors: ../sound/core/isadma.c: In function 'snd_dma_program': ../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration] 33 | flags = claim_dma_lock(); | ^~~~~~~~~~~~~~ ../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration] 41 | release_dma_lock(flags); | ^~~~~~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare': ../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function) 253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); | ^~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare': ../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function) 322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); | ^~~~~~~~~~~~ and more... Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-m68k(a)lists.linux-m68k.org Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Link: https://lore.kernel.org/r/20211016062602.3588-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/Makefile | 2 ++ sound/isa/Kconfig | 2 +- sound/pci/Kconfig | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/Makefile b/sound/core/Makefile index ee4a4a6b99ba7..d123587c0fd8f 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 6ffa48dd59830..570b88e0b2018 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -22,7 +22,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 93bc9bef7641f..41ce125971777 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -279,6 +279,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. -- 2.33.0

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[PATCH AUTOSEL 5.10 30/50] ASoC: Intel: sof_sdw: add missing quirk for Dell SKU 0A45
by Sasha Levin 10 Nov '21

10 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 64ba6d2ce72ffde70dc5a1794917bf1573203716 ] This device is based on SDCA codecs but with a single amplifier instead of two. BugLink: https://github.com/thesofproject/linux/issues/3161 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 <bard.liao(a)intel.com> Link: https://lore.kernel.org/r/20211004213512.220836-6-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_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 25548555d8d79..d9b864856be19 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -187,6 +187,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_RT715_DAI_ID_FIX | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A45") + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_RT715_DAI_ID_FIX), + }, /* AlderLake devices */ { .callback = sof_sdw_quirk_cb, -- 2.33.0

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[PATCH AUTOSEL 5.10 29/50] ALSA: intel-dsp-config: add quirk for APL/GLK/TGL devices based on ES8336 codec
by Sasha Levin 10 Nov '21

10 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 9d36ceab94151f07cf3fcb067213ac87937adf12 ] These devices are based on an I2C/I2S device, we need to force the use of the SOF driver otherwise the legacy HDaudio driver will be loaded - only HDMI will be supported. Co-developed-by: Huajun Li <huajun.li(a)intel.com> Signed-off-by: Huajun Li <huajun.li(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211004213512.220836-3-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/intel-dsp-config.c | 22 ++++++++++++++++++++-- 1 file changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 61e1de6d7be0a..6cdb3db7507b1 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -30,6 +30,7 @@ struct config_entry { u32 flags; u16 device; const struct dmi_system_id *dmi_table; + u8 codec_hid[ACPI_ID_LEN]; }; /* @@ -55,7 +56,7 @@ static const struct config_entry config_table[] = { /* * Apollolake (Broxton-P) * the legacy HDAudio driver is used except on Up Squared (SOF) and - * Chromebooks (SST) + * Chromebooks (SST), as well as devices based on the ES8336 codec */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) { @@ -72,6 +73,11 @@ static const struct config_entry config_table[] = { {} } }, + { + .flags = FLAG_SOF, + .device = 0x5a98, + .codec_hid = "ESSX8336", + }, #endif #if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) { @@ -136,7 +142,7 @@ static const struct config_entry config_table[] = { /* * Geminilake uses legacy HDAudio driver except for Google - * Chromebooks + * Chromebooks and devices based on the ES8336 codec */ /* Geminilake */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) @@ -153,6 +159,11 @@ static const struct config_entry config_table[] = { {} } }, + { + .flags = FLAG_SOF, + .device = 0x3198, + .codec_hid = "ESSX8336", + }, #endif /* @@ -310,6 +321,11 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x43c8, }, + { + .flags = FLAG_SOF, + .device = 0xa0c8, + .codec_hid = "ESSX8336", + }, #endif /* Elkhart Lake */ @@ -337,6 +353,8 @@ static const struct config_entry *snd_intel_dsp_find_config continue; if (table->dmi_table && !dmi_check_system(table->dmi_table)) continue; + if (table->codec_hid[0] && !acpi_dev_present(table->codec_hid, NULL, -1)) + continue; return table; } return NULL; -- 2.33.0

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[PATCH AUTOSEL 5.10 27/50] ASoC: nau8824: Add DMI quirk mechanism for active-high jack-detect
by Sasha Levin 10 Nov '21

10 Nov '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 92d3360108f1839ca40451bad20ff67dd24a1964 ] Add a quirk mechanism to allow specifying that active-high jack-detection should be used on platforms where this info is not available in devicetree. And add an entry for the Cyberbook T116 tablet to the DMI table, so that jack-detection will work properly on this tablet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20211002211459.110124-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 40 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 15bd8335f6678..c8ccfa2fff848 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/init.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -27,6 +28,12 @@ #include "nau8824.h" +#define NAU8824_JD_ACTIVE_HIGH BIT(0) + +static int nau8824_quirk; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static int nau8824_config_sysclk(struct nau8824 *nau8824, int clk_id, unsigned int freq); @@ -1875,6 +1882,34 @@ static int nau8824_read_device_properties(struct device *dev, return 0; } +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id nau8824_quirk_table[] = { + { + /* Cyberbook T116 rugged tablet */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Default string"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "20170531"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + {} +}; + +static void nau8824_check_quirks(void) +{ + const struct dmi_system_id *dmi_id; + + if (quirk_override != -1) { + nau8824_quirk = quirk_override; + return; + } + + dmi_id = dmi_first_match(nau8824_quirk_table); + if (dmi_id) + nau8824_quirk = (unsigned long)dmi_id->driver_data; +} + static int nau8824_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1899,6 +1934,11 @@ static int nau8824_i2c_probe(struct i2c_client *i2c, nau8824->irq = i2c->irq; sema_init(&nau8824->jd_sem, 1); + nau8824_check_quirks(); + + if (nau8824_quirk & NAU8824_JD_ACTIVE_HIGH) + nau8824->jkdet_polarity = 0; + nau8824_print_device_properties(nau8824); ret = regmap_read(nau8824->regmap, NAU8824_REG_I2C_DEVICE_ID, &value); -- 2.33.0

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[PATCH AUTOSEL 5.10 25/50] ASoC: SOF: Intel: hda-dai: fix potential locking issue
by Sasha Levin 10 Nov '21

10 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit a20f3b10de61add5e14b6ce4df982f4df2a4cbbc ] The initial hdac_stream code was adapted a third time with the same locking issues. Move the spin_lock outside the loops and make sure the fields are protected on read/write. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Acked-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210924192417.169243-5-pierre-louis.bossart@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-dai.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index c6cb8c212eca5..ef316311e959a 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -68,6 +68,7 @@ static struct hdac_ext_stream * return NULL; } + spin_lock_irq(&bus->reg_lock); list_for_each_entry(stream, &bus->stream_list, list) { struct hdac_ext_stream *hstream = stream_to_hdac_ext_stream(stream); @@ -107,12 +108,12 @@ static struct hdac_ext_stream * * is updated in snd_hdac_ext_stream_decouple(). */ if (!res->decoupled) - snd_hdac_ext_stream_decouple(bus, res, true); - spin_lock_irq(&bus->reg_lock); + snd_hdac_ext_stream_decouple_locked(bus, res, true); + res->link_locked = 1; res->link_substream = substream; - spin_unlock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); return res; } -- 2.33.0

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[PATCH AUTOSEL 5.14 71/75] ALSA: usb-audio: fix null pointer dereference on pointer cs_desc
by Sasha Levin 10 Nov '21

10 Nov '21

From: Chengfeng Ye <cyeaa(a)connect.ust.hk> [ Upstream commit b97053df0f04747c3c1e021ecbe99db675342954 ] The pointer cs_desc return from snd_usb_find_clock_source could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> Link: https://lore.kernel.org/r/20211024111736.11342-1-cyeaa@connect.ust.hk Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/clock.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 14456f61539e1..6c3f3312113f6 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -499,6 +499,10 @@ int snd_usb_set_sample_rate_v2v3(struct snd_usb_audio *chip, union uac23_clock_source_desc *cs_desc; cs_desc = snd_usb_find_clock_source(chip, clock, fmt->protocol); + + if (!cs_desc) + return 0; + if (fmt->protocol == UAC_VERSION_3) bmControls = le32_to_cpu(cs_desc->v3.bmControls); else -- 2.33.0

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[PATCH AUTOSEL 5.14 70/75] ALSA: gus: fix null pointer dereference on pointer block
by Sasha Levin 10 Nov '21

10 Nov '21

From: Chengfeng Ye <cyeaa(a)connect.ust.hk> [ Upstream commit a0d21bb3279476c777434c40d969ea88ca64f9aa ] The pointer block return from snd_gf1_dma_next_block could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> Link: https://lore.kernel.org/r/20211024104611.9919-1-cyeaa@connect.ust.hk Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/gus/gus_dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index a1c770d826dda..6d664dd8dde0b 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -126,6 +126,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) } block = snd_gf1_dma_next_block(gus); spin_unlock(&gus->dma_lock); + if (!block) + return; snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 -- 2.33.0

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[PATCH AUTOSEL 5.14 54/75] ALSA: ISA: not for M68K
by Sasha Levin 10 Nov '21

10 Nov '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 ] On m68k, compiling drivers under SND_ISA causes build errors: ../sound/core/isadma.c: In function 'snd_dma_program': ../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration] 33 | flags = claim_dma_lock(); | ^~~~~~~~~~~~~~ ../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration] 41 | release_dma_lock(flags); | ^~~~~~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare': ../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function) 253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); | ^~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare': ../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function) 322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); | ^~~~~~~~~~~~ and more... Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-m68k(a)lists.linux-m68k.org Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Link: https://lore.kernel.org/r/20211016062602.3588-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/Makefile | 2 ++ sound/isa/Kconfig | 2 +- sound/pci/Kconfig | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/Makefile b/sound/core/Makefile index d774792850f31..79e1407cd0de7 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 6ffa48dd59830..570b88e0b2018 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -22,7 +22,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 93bc9bef7641f..41ce125971777 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -279,6 +279,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. -- 2.33.0

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[PATCH AUTOSEL 5.14 49/75] ALSA: usb-audio: Add support for the Pioneer DJM 750MK2 Mixer/Soundcard
by Sasha Levin 10 Nov '21

10 Nov '21

From: William Overton <willovertonuk(a)gmail.com> [ Upstream commit 6d27788160362a7ee6c0d317636fe4b1ddbe59a7 ] The kernel already has support for very similar Pioneer djm products and this work is based on that. Added device to quirks-table.h and added control info to mixer_quirks.c. Tested on my hardware and all working. Signed-off-by: William Overton <willovertonuk(a)gmail.com> Link: https://lore.kernel.org/r/20211010145841.11907-1-willovertonuk@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_quirks.c | 34 +++++++++++++++++++++++ sound/usb/quirks-table.h | 58 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 92 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 4a4d3361ac047..ef92ac37cf1a9 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2654,6 +2654,7 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_750_IDX 0x1 #define SND_DJM_850_IDX 0x2 #define SND_DJM_900NXS2_IDX 0x3 +#define SND_DJM_750MK2_IDX 0x4 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \ @@ -2843,10 +2844,40 @@ static const struct snd_djm_ctl snd_djm_ctls_900nxs2[] = { SND_DJM_CTL("Ch5 Input", 900nxs2_cap5, 3, SND_DJM_WINDEX_CAP) }; +// DJM-750MK2 +static const u16 snd_djm_opts_750mk2_cap1[] = { + 0x0100, 0x0102, 0x0103, 0x0106, 0x0107, 0x0108, 0x0109, 0x010a }; +static const u16 snd_djm_opts_750mk2_cap2[] = { + 0x0200, 0x0202, 0x0203, 0x0206, 0x0207, 0x0208, 0x0209, 0x020a }; +static const u16 snd_djm_opts_750mk2_cap3[] = { + 0x0300, 0x0302, 0x0303, 0x0306, 0x0307, 0x0308, 0x0309, 0x030a }; +static const u16 snd_djm_opts_750mk2_cap4[] = { + 0x0400, 0x0402, 0x0403, 0x0406, 0x0407, 0x0408, 0x0409, 0x040a }; +static const u16 snd_djm_opts_750mk2_cap5[] = { + 0x0507, 0x0508, 0x0509, 0x050a, 0x0511, 0x0512, 0x0513, 0x0514 }; + +static const u16 snd_djm_opts_750mk2_pb1[] = { 0x0100, 0x0101, 0x0104 }; +static const u16 snd_djm_opts_750mk2_pb2[] = { 0x0200, 0x0201, 0x0204 }; +static const u16 snd_djm_opts_750mk2_pb3[] = { 0x0300, 0x0301, 0x0304 }; + + +static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { + SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Ch1 Input", 750mk2_cap1, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch2 Input", 750mk2_cap2, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch3 Input", 750mk2_cap3, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch4 Input", 750mk2_cap4, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch5 Input", 750mk2_cap5, 3, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch1 Output", 750mk2_pb1, 0, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch2 Output", 750mk2_pb2, 1, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch3 Output", 750mk2_pb3, 2, SND_DJM_WINDEX_PB) +}; + static const struct snd_djm_device snd_djm_devices[] = { SND_DJM_DEVICE(250mk2), SND_DJM_DEVICE(750), + SND_DJM_DEVICE(750mk2), SND_DJM_DEVICE(850), SND_DJM_DEVICE(900nxs2) }; @@ -3094,6 +3125,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x08e4, 0x017f): /* Pioneer DJ DJM-750 */ err = snd_djm_controls_create(mixer, SND_DJM_750_IDX); break; + case USB_ID(0x2b73, 0x001b): /* Pioneer DJ DJM-750MK2 */ + err = snd_djm_controls_create(mixer, SND_DJM_750MK2_IDX); + break; case USB_ID(0x08e4, 0x0163): /* Pioneer DJ DJM-850 */ err = snd_djm_controls_create(mixer, SND_DJM_850_IDX); break; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 91d40b4c851c1..ac55c7dad950f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3960,6 +3960,64 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-750MK2 + * 10 channels playback & 12 channels capture @ 48kHz S24LE + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x001b), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 10, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .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 = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, { /* * Pioneer DJ DJM-850 -- 2.33.0

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[PATCH AUTOSEL 5.14 45/75] ASoC: Intel: sof_sdw: add missing quirk for Dell SKU 0A45
by Sasha Levin 10 Nov '21

10 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 64ba6d2ce72ffde70dc5a1794917bf1573203716 ] This device is based on SDCA codecs but with a single amplifier instead of two. BugLink: https://github.com/thesofproject/linux/issues/3161 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 <bard.liao(a)intel.com> Link: https://lore.kernel.org/r/20211004213512.220836-6-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_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index cb3afc4519cf6..041717c71c930 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -189,6 +189,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_RT715_DAI_ID_FIX | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A45") + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_RT715_DAI_ID_FIX), + }, /* AlderLake devices */ { .callback = sof_sdw_quirk_cb, -- 2.33.0

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[PATCH AUTOSEL 5.14 44/75] ASoC: Intel: soc-acpi: add missing quirk for TGL SDCA single amp
by Sasha Levin 10 Nov '21

10 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit f2470679b070a77ea22f8b791fae7084c2340c7d ] We don't have a configuration for a single amp on link1. BugLink: https://github.com/thesofproject/linux/issues/3161 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 <bard.liao(a)intel.com> Link: https://lore.kernel.org/r/20211004213512.220836-5-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- .../intel/common/soc-acpi-intel-tgl-match.c | 41 +++++++++++++++++++ 1 file changed, 41 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 66595e3ab13f6..def7d390b08d4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -155,6 +155,15 @@ static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1316_1_single_adr[] = { + { + .adr = 0x000131025D131601ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt1316-1" + } +}; + static const struct snd_soc_acpi_adr_device rt1316_1_group1_adr[] = { { .adr = 0x000131025D131601ull, /* unique ID is set for some reason */ @@ -310,6 +319,25 @@ static const struct snd_soc_acpi_link_adr tgl_3_in_1_sdca[] = { {} }; +static const struct snd_soc_acpi_link_adr tgl_3_in_1_sdca_mono[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1316_1_single_adr), + .adr_d = rt1316_1_single_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt714_3_adr), + .adr_d = rt714_3_adr, + }, + {} +}; + static const struct snd_soc_acpi_codecs tgl_max98373_amp = { .num_codecs = 1, .codecs = {"MX98373"} @@ -380,6 +408,19 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-tgl-rt711-rt1316-rt714.tplg", }, + { + /* + * link_mask should be 0xB, but all links are enabled by BIOS. + * This entry will be selected if there is no rt1316 amplifier exposed + * on link2 since it will fail to match the above entry. + */ + + .link_mask = 0xF, /* 4 active links required */ + .links = tgl_3_in_1_sdca_mono, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-tgl-rt711-l0-rt1316-l1-mono-rt714-l3.tplg", + }, + { .link_mask = 0x3, /* rt711 on link 0 and 1 rt1308 on link 1 */ .links = tgl_hp, -- 2.33.0

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[PATCH AUTOSEL 5.14 43/75] ALSA: intel-dsp-config: add quirk for APL/GLK/TGL devices based on ES8336 codec
by Sasha Levin 10 Nov '21

10 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 9d36ceab94151f07cf3fcb067213ac87937adf12 ] These devices are based on an I2C/I2S device, we need to force the use of the SOF driver otherwise the legacy HDaudio driver will be loaded - only HDMI will be supported. Co-developed-by: Huajun Li <huajun.li(a)intel.com> Signed-off-by: Huajun Li <huajun.li(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211004213512.220836-3-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/intel-dsp-config.c | 22 ++++++++++++++++++++-- 1 file changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index c9d0ba353463b..b9ac9e9e45a48 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -31,6 +31,7 @@ struct config_entry { u16 device; u8 acpi_hid[ACPI_ID_LEN]; const struct dmi_system_id *dmi_table; + u8 codec_hid[ACPI_ID_LEN]; }; /* @@ -56,7 +57,7 @@ static const struct config_entry config_table[] = { /* * Apollolake (Broxton-P) * the legacy HDAudio driver is used except on Up Squared (SOF) and - * Chromebooks (SST) + * Chromebooks (SST), as well as devices based on the ES8336 codec */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) { @@ -73,6 +74,11 @@ static const struct config_entry config_table[] = { {} } }, + { + .flags = FLAG_SOF, + .device = 0x5a98, + .codec_hid = "ESSX8336", + }, #endif #if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) { @@ -137,7 +143,7 @@ static const struct config_entry config_table[] = { /* * Geminilake uses legacy HDAudio driver except for Google - * Chromebooks + * Chromebooks and devices based on the ES8336 codec */ /* Geminilake */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) @@ -154,6 +160,11 @@ static const struct config_entry config_table[] = { {} } }, + { + .flags = FLAG_SOF, + .device = 0x3198, + .codec_hid = "ESSX8336", + }, #endif /* @@ -311,6 +322,11 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x43c8, }, + { + .flags = FLAG_SOF, + .device = 0xa0c8, + .codec_hid = "ESSX8336", + }, #endif /* Elkhart Lake */ @@ -354,6 +370,8 @@ static const struct config_entry *snd_intel_dsp_find_config continue; if (table->dmi_table && !dmi_check_system(table->dmi_table)) continue; + if (table->codec_hid[0] && !acpi_dev_present(table->codec_hid, NULL, -1)) + continue; return table; } return NULL; -- 2.33.0

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[PATCH AUTOSEL 5.14 39/75] ASoC: nau8824: Add DMI quirk mechanism for active-high jack-detect
by Sasha Levin 10 Nov '21

10 Nov '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 92d3360108f1839ca40451bad20ff67dd24a1964 ] Add a quirk mechanism to allow specifying that active-high jack-detection should be used on platforms where this info is not available in devicetree. And add an entry for the Cyberbook T116 tablet to the DMI table, so that jack-detection will work properly on this tablet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20211002211459.110124-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 40 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index f946ef65a4c19..f7018f2dd21fd 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/init.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -27,6 +28,12 @@ #include "nau8824.h" +#define NAU8824_JD_ACTIVE_HIGH BIT(0) + +static int nau8824_quirk; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static int nau8824_config_sysclk(struct nau8824 *nau8824, int clk_id, unsigned int freq); @@ -1845,6 +1852,34 @@ static int nau8824_read_device_properties(struct device *dev, return 0; } +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id nau8824_quirk_table[] = { + { + /* Cyberbook T116 rugged tablet */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Default string"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "20170531"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + {} +}; + +static void nau8824_check_quirks(void) +{ + const struct dmi_system_id *dmi_id; + + if (quirk_override != -1) { + nau8824_quirk = quirk_override; + return; + } + + dmi_id = dmi_first_match(nau8824_quirk_table); + if (dmi_id) + nau8824_quirk = (unsigned long)dmi_id->driver_data; +} + static int nau8824_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1869,6 +1904,11 @@ static int nau8824_i2c_probe(struct i2c_client *i2c, nau8824->irq = i2c->irq; sema_init(&nau8824->jd_sem, 1); + nau8824_check_quirks(); + + if (nau8824_quirk & NAU8824_JD_ACTIVE_HIGH) + nau8824->jkdet_polarity = 0; + nau8824_print_device_properties(nau8824); ret = regmap_read(nau8824->regmap, NAU8824_REG_I2C_DEVICE_ID, &value); -- 2.33.0

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[PATCH AUTOSEL 5.14 38/75] ASoC: rt5651: Use IRQF_NO_AUTOEN when requesting the IRQ
by Sasha Levin 10 Nov '21

10 Nov '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 6e037b72cf4ea6c28a131ea021d63ee4e7e6fa64 ] Use the new IRQF_NO_AUTOEN flag when requesting the IRQ, rather then disabling it immediately after requesting it. This fixes a possible race where the IRQ might trigger between requesting and disabling it; and this also leads to a small code cleanup. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20211003132255.31743-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index fc0c83b73f099..93820561b9f5d 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -2261,11 +2261,8 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, ret = devm_request_irq(&i2c->dev, rt5651->irq, rt5651_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING - | IRQF_ONESHOT, "rt5651", rt5651); - if (ret == 0) { - /* Gets re-enabled by rt5651_set_jack() */ - disable_irq(rt5651->irq); - } else { + | IRQF_ONESHOT | IRQF_NO_AUTOEN, "rt5651", rt5651); + if (ret) { dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", rt5651->irq, ret); rt5651->irq = -ENXIO; -- 2.33.0

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[PATCH AUTOSEL 5.14 37/75] ASoC: es8316: Use IRQF_NO_AUTOEN when requesting the IRQ
by Sasha Levin 10 Nov '21

10 Nov '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 1cf2aa665901054b140eb71748661ceae99b6b5a ] Use the new IRQF_NO_AUTOEN flag when requesting the IRQ, rather then disabling it immediately after requesting it. This fixes a possible race where the IRQ might trigger between requesting and disabling it; and this also leads to a small code cleanup. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20211003132255.31743-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 067757d1d70a3..5fb02635c1406 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -811,12 +811,9 @@ static int es8316_i2c_probe(struct i2c_client *i2c_client, mutex_init(&es8316->lock); ret = devm_request_threaded_irq(dev, es8316->irq, NULL, es8316_irq, - IRQF_TRIGGER_HIGH | IRQF_ONESHOT, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT | IRQF_NO_AUTOEN, "es8316", es8316); - if (ret == 0) { - /* Gets re-enabled by es8316_set_jack() */ - disable_irq(es8316->irq); - } else { + if (ret) { dev_warn(dev, "Failed to get IRQ %d: %d\n", es8316->irq, ret); es8316->irq = -ENXIO; } -- 2.33.0

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[PATCH AUTOSEL 5.14 35/75] ALSA: usb-audio: disable implicit feedback sync for Behringer UFX1204 and UFX1604
by Sasha Levin 10 Nov '21

10 Nov '21

From: Geraldo Nascimento <geraldogabriel(a)gmail.com> [ Upstream commit 28c369e60827f706cef4604a3e2848198f25bd26 ] Behringer UFX1204 and UFX1604 have Synchronous endpoints to which current ALSA code applies implicit feedback sync as if they were Asynchronous endpoints. This breaks UAC compliance and is unneeded. The commit 5e35dc0338d85ccebacf3f77eca1e5dea73155e8 and subsequent 1a15718b41df026cffd0e42cfdc38a1384ce19f9 were meant to clear up noise. Unfortunately, noise persisted for those using higher sample rates and this was only solved by commit d2e8f641257d0d3af6e45d6ac2d6f9d56b8ea964 Since there are no more reports of noise, let's get rid of the implicit-fb quirks breaking UAC compliance. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> Link: https://lore.kernel.org/r/YVYSnoQ7nxLXT0Dq@geday Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/implicit.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 590a0dbba7a21..34461947260b2 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -54,8 +54,6 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* Fixed EP */ /* FIXME: check the availability of generic matching */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0001, 0x81, 1), /* Behringer UFX1604 */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0002, 0x81, 1), /* Behringer UFX1204 */ IMPLICIT_FB_FIXED_DEV(0x2466, 0x8010, 0x81, 2), /* Fractal Audio Axe-Fx III */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0001, 0x81, 2), /* Solid State Logic SSL2 */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0002, 0x81, 2), /* Solid State Logic SSL2+ */ -- 2.33.0

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[PATCH AUTOSEL 5.14 33/75] ASoC: SOF: Intel: hda-dai: fix potential locking issue
by Sasha Levin 10 Nov '21

10 Nov '21

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

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[PATCH AUTOSEL 5.15 77/82] ALSA: usb-audio: fix null pointer dereference on pointer cs_desc
by Sasha Levin 10 Nov '21

10 Nov '21

From: Chengfeng Ye <cyeaa(a)connect.ust.hk> [ Upstream commit b97053df0f04747c3c1e021ecbe99db675342954 ] The pointer cs_desc return from snd_usb_find_clock_source could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> Link: https://lore.kernel.org/r/20211024111736.11342-1-cyeaa@connect.ust.hk Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/clock.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 81d5ce07d548b..98345a695dccb 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -496,6 +496,10 @@ int snd_usb_set_sample_rate_v2v3(struct snd_usb_audio *chip, union uac23_clock_source_desc *cs_desc; cs_desc = snd_usb_find_clock_source(chip, clock, fmt->protocol); + + if (!cs_desc) + return 0; + if (fmt->protocol == UAC_VERSION_3) bmControls = le32_to_cpu(cs_desc->v3.bmControls); else -- 2.33.0

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[PATCH AUTOSEL 5.15 76/82] ALSA: gus: fix null pointer dereference on pointer block
by Sasha Levin 10 Nov '21

10 Nov '21

From: Chengfeng Ye <cyeaa(a)connect.ust.hk> [ Upstream commit a0d21bb3279476c777434c40d969ea88ca64f9aa ] The pointer block return from snd_gf1_dma_next_block could be null, so there is a potential null pointer dereference issue. Fix this by adding a null check before dereference. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> Link: https://lore.kernel.org/r/20211024104611.9919-1-cyeaa@connect.ust.hk Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/gus/gus_dma.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index a1c770d826dda..6d664dd8dde0b 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -126,6 +126,8 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) } block = snd_gf1_dma_next_block(gus); spin_unlock(&gus->dma_lock); + if (!block) + return; snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 -- 2.33.0

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[PATCH AUTOSEL 5.15 58/82] ALSA: ISA: not for M68K
by Sasha Levin 10 Nov '21

10 Nov '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 ] On m68k, compiling drivers under SND_ISA causes build errors: ../sound/core/isadma.c: In function 'snd_dma_program': ../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration] 33 | flags = claim_dma_lock(); | ^~~~~~~~~~~~~~ ../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration] 41 | release_dma_lock(flags); | ^~~~~~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare': ../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function) 253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); | ^~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare': ../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function) 322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); | ^~~~~~~~~~~~ and more... Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-m68k(a)lists.linux-m68k.org Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Link: https://lore.kernel.org/r/20211016062602.3588-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/Makefile | 2 ++ sound/isa/Kconfig | 2 +- sound/pci/Kconfig | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/Makefile b/sound/core/Makefile index d774792850f31..79e1407cd0de7 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 6ffa48dd59830..570b88e0b2018 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -22,7 +22,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 93bc9bef7641f..41ce125971777 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -279,6 +279,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. -- 2.33.0

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[PATCH AUTOSEL 5.15 55/82] ASoC: rt5682: fix a little pop while playback
by Sasha Levin 10 Nov '21

10 Nov '21

From: Derek Fang <derek.fang(a)realtek.com> [ Upstream commit 4b19e4a77cc6baa0f840e8bae62ab974667f6207 ] A little pop can be heard obviously from HP while playing a silent. This patch fixes it by using two functions: 1. Enable HP 1bit output mode. 2. Change the charge pump switch size during playback on and off. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> Link: https://lore.kernel.org/r/20211014094054.811-1-derek.fang@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 56 +++++++++++++++++++++++++++++++++------ sound/soc/codecs/rt5682.h | 20 ++++++++++++++ 2 files changed, 68 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 4a64cab99c55b..d550c0705c28b 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -46,6 +46,8 @@ static const struct reg_sequence patch_list[] = { {RT5682_SAR_IL_CMD_1, 0x22b7}, {RT5682_SAR_IL_CMD_3, 0x0365}, {RT5682_SAR_IL_CMD_6, 0x0110}, + {RT5682_CHARGE_PUMP_1, 0x0210}, + {RT5682_HP_LOGIC_CTRL_2, 0x0007}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) @@ -1515,21 +1517,29 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - snd_soc_component_write(component, - RT5682_HP_LOGIC_CTRL_2, 0x0012); - snd_soc_component_write(component, - RT5682_HP_CTRL_2, 0x6000); + snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, + RT5682_HP_C2_DAC_AMP_MUTE, 0); + snd_soc_component_update_bits(component, RT5682_HP_LOGIC_CTRL_2, + RT5682_HP_LC2_SIG_SOUR2_MASK, RT5682_HP_LC2_SIG_SOUR2_REG); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x60); snd_soc_component_update_bits(component, RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080); + snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, + RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN, + RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN); + usleep_range(5000, 10000); + snd_soc_component_update_bits(component, RT5682_CHARGE_PUMP_1, + RT5682_CP_SW_SIZE_MASK, RT5682_CP_SW_SIZE_L); break; case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, + RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN, 0); + snd_soc_component_update_bits(component, RT5682_CHARGE_PUMP_1, + RT5682_CP_SW_SIZE_MASK, RT5682_CP_SW_SIZE_M); snd_soc_component_update_bits(component, RT5682_DEPOP_1, 0x60, 0x0); - snd_soc_component_write(component, - RT5682_HP_CTRL_2, 0x0000); snd_soc_component_update_bits(component, RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0000); break; @@ -1637,6 +1647,23 @@ static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_adcdat_pin_enum, static const struct snd_kcontrol_new rt5682_adcdat_pin_ctrl = SOC_DAPM_ENUM("ADCDAT", rt5682_adcdat_pin_enum); +static const unsigned int rt5682_hpo_sig_out_values[] = { + 2, + 7, +}; + +static const char * const rt5682_hpo_sig_out_mode[] = { + "Legacy", + "OneBit", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5682_hpo_sig_out_enum, + RT5682_HP_LOGIC_CTRL_2, 0, RT5682_HP_LC2_SIG_SOUR1_MASK, + rt5682_hpo_sig_out_mode, rt5682_hpo_sig_out_values); + +static const struct snd_kcontrol_new rt5682_hpo_sig_demux = + SOC_DAPM_ENUM("HPO Signal Demux", rt5682_hpo_sig_out_enum); + static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5682_PWR_ANLG_3, RT5682_PWR_LDO2_BIT, 0, NULL, 0), @@ -1820,6 +1847,10 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, &hpor_switch), + SND_SOC_DAPM_OUT_DRV("HPO Legacy", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HPO OneBit", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_DEMUX("HPO Signal Demux", SND_SOC_NOPM, 0, 0, &rt5682_hpo_sig_demux), + /* CLK DET */ SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682_CLK_DET, RT5682_SYS_CLK_DET_SFT, 0, NULL, 0), @@ -1987,10 +2018,19 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, {"HP Amp", NULL, "Vref1"}, - {"HPOL Playback", "Switch", "HP Amp"}, - {"HPOR Playback", "Switch", "HP Amp"}, + + {"HPO Signal Demux", NULL, "HP Amp"}, + + {"HPO Legacy", "Legacy", "HPO Signal Demux"}, + {"HPO OneBit", "OneBit", "HPO Signal Demux"}, + + {"HPOL Playback", "Switch", "HPO Legacy"}, + {"HPOR Playback", "Switch", "HPO Legacy"}, + {"HPOL", NULL, "HPOL Playback"}, {"HPOR", NULL, "HPOR Playback"}, + {"HPOL", NULL, "HPO OneBit"}, + {"HPOR", NULL, "HPO OneBit"}, }; static int rt5682_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index b59221048ebf9..8e3244a62c160 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -375,6 +375,14 @@ #define RT5682_R_VOL_MASK (0x3f) #define RT5682_R_VOL_SFT 0 +/* Headphone Amp Control 2 (0x0003) */ +#define RT5682_HP_C2_DAC_AMP_MUTE_SFT 15 +#define RT5682_HP_C2_DAC_AMP_MUTE (0x1 << 15) +#define RT5682_HP_C2_DAC_L_EN_SFT 14 +#define RT5682_HP_C2_DAC_L_EN (0x1 << 14) +#define RT5682_HP_C2_DAC_R_EN_SFT 13 +#define RT5682_HP_C2_DAC_R_EN (0x1 << 13) + /*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ #define RT5682_G_HP (0xf << 8) #define RT5682_G_HP_SFT 8 @@ -1265,6 +1273,10 @@ #define RT5682_HPA_CP_BIAS_6UA (0x3 << 2) /* Charge Pump Internal Register1 (0x0125) */ +#define RT5682_CP_SW_SIZE_MASK (0x7 << 8) +#define RT5682_CP_SW_SIZE_L (0x4 << 8) +#define RT5682_CP_SW_SIZE_M (0x2 << 8) +#define RT5682_CP_SW_SIZE_S (0x1 << 8) #define RT5682_CP_CLK_HP_MASK (0x3 << 4) #define RT5682_CP_CLK_HP_100KHZ (0x0 << 4) #define RT5682_CP_CLK_HP_200KHZ (0x1 << 4) @@ -1315,6 +1327,14 @@ #define RT5682_DEB_STO_DAC_MASK (0x7 << 4) #define RT5682_DEB_80_MS (0x0 << 4) +/* HP Behavior Logic Control 2 (0x01db) */ +#define RT5682_HP_LC2_SIG_SOUR2_MASK (0x1 << 4) +#define RT5682_HP_LC2_SIG_SOUR2_REG (0x1 << 4) +#define RT5682_HP_LC2_SIG_SOUR2_DC_CAL (0x0 << 4) +#define RT5682_HP_LC2_SIG_SOUR1_MASK (0x7) +#define RT5682_HP_LC2_SIG_SOUR1_1BIT (0x7) +#define RT5682_HP_LC2_SIG_SOUR1_LEGA (0x2) + /* SAR ADC Inline Command Control 1 (0x0210) */ #define RT5682_SAR_BUTT_DET_MASK (0x1 << 15) #define RT5682_SAR_BUTT_DET_EN (0x1 << 15) -- 2.33.0

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[PATCH AUTOSEL 5.15 52/82] ALSA: usb-audio: Add support for the Pioneer DJM 750MK2 Mixer/Soundcard
by Sasha Levin 10 Nov '21

10 Nov '21

From: William Overton <willovertonuk(a)gmail.com> [ Upstream commit 6d27788160362a7ee6c0d317636fe4b1ddbe59a7 ] The kernel already has support for very similar Pioneer djm products and this work is based on that. Added device to quirks-table.h and added control info to mixer_quirks.c. Tested on my hardware and all working. Signed-off-by: William Overton <willovertonuk(a)gmail.com> Link: https://lore.kernel.org/r/20211010145841.11907-1-willovertonuk@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_quirks.c | 34 +++++++++++++++++++++++ sound/usb/quirks-table.h | 58 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 92 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 46082dc57be09..d489c1de3baec 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2795,6 +2795,7 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_750_IDX 0x1 #define SND_DJM_850_IDX 0x2 #define SND_DJM_900NXS2_IDX 0x3 +#define SND_DJM_750MK2_IDX 0x4 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \ @@ -2984,10 +2985,40 @@ static const struct snd_djm_ctl snd_djm_ctls_900nxs2[] = { SND_DJM_CTL("Ch5 Input", 900nxs2_cap5, 3, SND_DJM_WINDEX_CAP) }; +// DJM-750MK2 +static const u16 snd_djm_opts_750mk2_cap1[] = { + 0x0100, 0x0102, 0x0103, 0x0106, 0x0107, 0x0108, 0x0109, 0x010a }; +static const u16 snd_djm_opts_750mk2_cap2[] = { + 0x0200, 0x0202, 0x0203, 0x0206, 0x0207, 0x0208, 0x0209, 0x020a }; +static const u16 snd_djm_opts_750mk2_cap3[] = { + 0x0300, 0x0302, 0x0303, 0x0306, 0x0307, 0x0308, 0x0309, 0x030a }; +static const u16 snd_djm_opts_750mk2_cap4[] = { + 0x0400, 0x0402, 0x0403, 0x0406, 0x0407, 0x0408, 0x0409, 0x040a }; +static const u16 snd_djm_opts_750mk2_cap5[] = { + 0x0507, 0x0508, 0x0509, 0x050a, 0x0511, 0x0512, 0x0513, 0x0514 }; + +static const u16 snd_djm_opts_750mk2_pb1[] = { 0x0100, 0x0101, 0x0104 }; +static const u16 snd_djm_opts_750mk2_pb2[] = { 0x0200, 0x0201, 0x0204 }; +static const u16 snd_djm_opts_750mk2_pb3[] = { 0x0300, 0x0301, 0x0304 }; + + +static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { + SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Ch1 Input", 750mk2_cap1, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch2 Input", 750mk2_cap2, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch3 Input", 750mk2_cap3, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch4 Input", 750mk2_cap4, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch5 Input", 750mk2_cap5, 3, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch1 Output", 750mk2_pb1, 0, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch2 Output", 750mk2_pb2, 1, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch3 Output", 750mk2_pb3, 2, SND_DJM_WINDEX_PB) +}; + static const struct snd_djm_device snd_djm_devices[] = { SND_DJM_DEVICE(250mk2), SND_DJM_DEVICE(750), + SND_DJM_DEVICE(750mk2), SND_DJM_DEVICE(850), SND_DJM_DEVICE(900nxs2) }; @@ -3235,6 +3266,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x08e4, 0x017f): /* Pioneer DJ DJM-750 */ err = snd_djm_controls_create(mixer, SND_DJM_750_IDX); break; + case USB_ID(0x2b73, 0x001b): /* Pioneer DJ DJM-750MK2 */ + err = snd_djm_controls_create(mixer, SND_DJM_750MK2_IDX); + break; case USB_ID(0x08e4, 0x0163): /* Pioneer DJ DJM-850 */ err = snd_djm_controls_create(mixer, SND_DJM_850_IDX); break; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 2af8c68fac275..b1522e43173e1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3892,6 +3892,64 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Pioneer DJ DJM-750MK2 + * 10 channels playback & 12 channels capture @ 48kHz S24LE + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x001b), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 10, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .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 = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, { /* * Pioneer DJ DJM-850 -- 2.33.0

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[PATCH AUTOSEL 5.15 48/82] ASoC: Intel: sof_sdw: add missing quirk for Dell SKU 0A45
by Sasha Levin 10 Nov '21

10 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 64ba6d2ce72ffde70dc5a1794917bf1573203716 ] This device is based on SDCA codecs but with a single amplifier instead of two. BugLink: https://github.com/thesofproject/linux/issues/3161 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 <bard.liao(a)intel.com> Link: https://lore.kernel.org/r/20211004213512.220836-6-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_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6b06248a9327a..f10496206ceed 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -213,6 +213,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_RT715_DAI_ID_FIX | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A45") + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_RT715_DAI_ID_FIX), + }, /* AlderLake devices */ { .callback = sof_sdw_quirk_cb, -- 2.33.0

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[PATCH AUTOSEL 5.15 47/82] ASoC: Intel: soc-acpi: add missing quirk for TGL SDCA single amp
by Sasha Levin 10 Nov '21

10 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit f2470679b070a77ea22f8b791fae7084c2340c7d ] We don't have a configuration for a single amp on link1. BugLink: https://github.com/thesofproject/linux/issues/3161 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 <bard.liao(a)intel.com> Link: https://lore.kernel.org/r/20211004213512.220836-5-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- .../intel/common/soc-acpi-intel-tgl-match.c | 41 +++++++++++++++++++ 1 file changed, 41 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 785d5f5f8a9c9..11801b905ecc2 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -156,6 +156,15 @@ static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1316_1_single_adr[] = { + { + .adr = 0x000131025D131601ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt1316-1" + } +}; + static const struct snd_soc_acpi_adr_device rt1316_1_group1_adr[] = { { .adr = 0x000131025D131601ull, /* unique ID is set for some reason */ @@ -320,6 +329,25 @@ static const struct snd_soc_acpi_link_adr tgl_3_in_1_sdca[] = { {} }; +static const struct snd_soc_acpi_link_adr tgl_3_in_1_sdca_mono[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1316_1_single_adr), + .adr_d = rt1316_1_single_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt714_3_adr), + .adr_d = rt714_3_adr, + }, + {} +}; + static const struct snd_soc_acpi_codecs tgl_max98373_amp = { .num_codecs = 1, .codecs = {"MX98373"} @@ -412,6 +440,19 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-tgl-rt711-rt1316-rt714.tplg", }, + { + /* + * link_mask should be 0xB, but all links are enabled by BIOS. + * This entry will be selected if there is no rt1316 amplifier exposed + * on link2 since it will fail to match the above entry. + */ + + .link_mask = 0xF, /* 4 active links required */ + .links = tgl_3_in_1_sdca_mono, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-tgl-rt711-l0-rt1316-l1-mono-rt714-l3.tplg", + }, + { .link_mask = 0x3, /* rt711 on link 0 and 1 rt1308 on link 1 */ .links = tgl_hp, -- 2.33.0

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[PATCH AUTOSEL 5.15 46/82] ALSA: intel-dsp-config: add quirk for APL/GLK/TGL devices based on ES8336 codec
by Sasha Levin 10 Nov '21

10 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 9d36ceab94151f07cf3fcb067213ac87937adf12 ] These devices are based on an I2C/I2S device, we need to force the use of the SOF driver otherwise the legacy HDaudio driver will be loaded - only HDMI will be supported. Co-developed-by: Huajun Li <huajun.li(a)intel.com> Signed-off-by: Huajun Li <huajun.li(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20211004213512.220836-3-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/intel-dsp-config.c | 22 ++++++++++++++++++++-- 1 file changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index c9d0ba353463b..b9ac9e9e45a48 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -31,6 +31,7 @@ struct config_entry { u16 device; u8 acpi_hid[ACPI_ID_LEN]; const struct dmi_system_id *dmi_table; + u8 codec_hid[ACPI_ID_LEN]; }; /* @@ -56,7 +57,7 @@ static const struct config_entry config_table[] = { /* * Apollolake (Broxton-P) * the legacy HDAudio driver is used except on Up Squared (SOF) and - * Chromebooks (SST) + * Chromebooks (SST), as well as devices based on the ES8336 codec */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) { @@ -73,6 +74,11 @@ static const struct config_entry config_table[] = { {} } }, + { + .flags = FLAG_SOF, + .device = 0x5a98, + .codec_hid = "ESSX8336", + }, #endif #if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) { @@ -137,7 +143,7 @@ static const struct config_entry config_table[] = { /* * Geminilake uses legacy HDAudio driver except for Google - * Chromebooks + * Chromebooks and devices based on the ES8336 codec */ /* Geminilake */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) @@ -154,6 +160,11 @@ static const struct config_entry config_table[] = { {} } }, + { + .flags = FLAG_SOF, + .device = 0x3198, + .codec_hid = "ESSX8336", + }, #endif /* @@ -311,6 +322,11 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x43c8, }, + { + .flags = FLAG_SOF, + .device = 0xa0c8, + .codec_hid = "ESSX8336", + }, #endif /* Elkhart Lake */ @@ -354,6 +370,8 @@ static const struct config_entry *snd_intel_dsp_find_config continue; if (table->dmi_table && !dmi_check_system(table->dmi_table)) continue; + if (table->codec_hid[0] && !acpi_dev_present(table->codec_hid, NULL, -1)) + continue; return table; } return NULL; -- 2.33.0

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[PATCH AUTOSEL 5.15 42/82] ASoC: nau8824: Add DMI quirk mechanism for active-high jack-detect
by Sasha Levin 10 Nov '21

10 Nov '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 92d3360108f1839ca40451bad20ff67dd24a1964 ] Add a quirk mechanism to allow specifying that active-high jack-detection should be used on platforms where this info is not available in devicetree. And add an entry for the Cyberbook T116 tablet to the DMI table, so that jack-detection will work properly on this tablet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20211002211459.110124-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8824.c | 40 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index f946ef65a4c19..f7018f2dd21fd 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/init.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -27,6 +28,12 @@ #include "nau8824.h" +#define NAU8824_JD_ACTIVE_HIGH BIT(0) + +static int nau8824_quirk; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static int nau8824_config_sysclk(struct nau8824 *nau8824, int clk_id, unsigned int freq); @@ -1845,6 +1852,34 @@ static int nau8824_read_device_properties(struct device *dev, return 0; } +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id nau8824_quirk_table[] = { + { + /* Cyberbook T116 rugged tablet */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Default string"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "Cherry Trail CR"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "20170531"), + }, + .driver_data = (void *)(NAU8824_JD_ACTIVE_HIGH), + }, + {} +}; + +static void nau8824_check_quirks(void) +{ + const struct dmi_system_id *dmi_id; + + if (quirk_override != -1) { + nau8824_quirk = quirk_override; + return; + } + + dmi_id = dmi_first_match(nau8824_quirk_table); + if (dmi_id) + nau8824_quirk = (unsigned long)dmi_id->driver_data; +} + static int nau8824_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1869,6 +1904,11 @@ static int nau8824_i2c_probe(struct i2c_client *i2c, nau8824->irq = i2c->irq; sema_init(&nau8824->jd_sem, 1); + nau8824_check_quirks(); + + if (nau8824_quirk & NAU8824_JD_ACTIVE_HIGH) + nau8824->jkdet_polarity = 0; + nau8824_print_device_properties(nau8824); ret = regmap_read(nau8824->regmap, NAU8824_REG_I2C_DEVICE_ID, &value); -- 2.33.0

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[PATCH AUTOSEL 5.15 41/82] ASoC: rt5651: Use IRQF_NO_AUTOEN when requesting the IRQ
by Sasha Levin 10 Nov '21

10 Nov '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 6e037b72cf4ea6c28a131ea021d63ee4e7e6fa64 ] Use the new IRQF_NO_AUTOEN flag when requesting the IRQ, rather then disabling it immediately after requesting it. This fixes a possible race where the IRQ might trigger between requesting and disabling it; and this also leads to a small code cleanup. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20211003132255.31743-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5651.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index fc0c83b73f099..93820561b9f5d 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -2261,11 +2261,8 @@ static int rt5651_i2c_probe(struct i2c_client *i2c, ret = devm_request_irq(&i2c->dev, rt5651->irq, rt5651_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING - | IRQF_ONESHOT, "rt5651", rt5651); - if (ret == 0) { - /* Gets re-enabled by rt5651_set_jack() */ - disable_irq(rt5651->irq); - } else { + | IRQF_ONESHOT | IRQF_NO_AUTOEN, "rt5651", rt5651); + if (ret) { dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", rt5651->irq, ret); rt5651->irq = -ENXIO; -- 2.33.0

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[PATCH AUTOSEL 5.15 40/82] ASoC: es8316: Use IRQF_NO_AUTOEN when requesting the IRQ
by Sasha Levin 10 Nov '21

10 Nov '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 1cf2aa665901054b140eb71748661ceae99b6b5a ] Use the new IRQF_NO_AUTOEN flag when requesting the IRQ, rather then disabling it immediately after requesting it. This fixes a possible race where the IRQ might trigger between requesting and disabling it; and this also leads to a small code cleanup. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/20211003132255.31743-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/es8316.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 067757d1d70a3..5fb02635c1406 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -811,12 +811,9 @@ static int es8316_i2c_probe(struct i2c_client *i2c_client, mutex_init(&es8316->lock); ret = devm_request_threaded_irq(dev, es8316->irq, NULL, es8316_irq, - IRQF_TRIGGER_HIGH | IRQF_ONESHOT, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT | IRQF_NO_AUTOEN, "es8316", es8316); - if (ret == 0) { - /* Gets re-enabled by es8316_set_jack() */ - disable_irq(es8316->irq); - } else { + if (ret) { dev_warn(dev, "Failed to get IRQ %d: %d\n", es8316->irq, ret); es8316->irq = -ENXIO; } -- 2.33.0

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[PATCH AUTOSEL 5.15 38/82] ALSA: usb-audio: disable implicit feedback sync for Behringer UFX1204 and UFX1604
by Sasha Levin 10 Nov '21

10 Nov '21

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

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[PATCH AUTOSEL 5.15 36/82] ASoC: SOF: Intel: hda-dai: fix potential locking issue
by Sasha Levin 10 Nov '21

10 Nov '21

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

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[PATCH AUTOSEL 5.15 14/82] ASoC: mediatek: mt8195: Add missing of_node_put()
by Sasha Levin 10 Nov '21

10 Nov '21

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

1 0

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

09 Nov '21

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

1 0

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

09 Nov '21

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

1 0

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

09 Nov '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..c4fed6335230 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..59ed510e6cf1 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[PATCH] ALSA: ctl: Fix copy of updated id with element read/write
by Alan Young 09 Nov '21

09 Nov '21

1 0

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

09 Nov '21

This is a codec driver for the TLV320ADC3001 and TLV320ADC3101 2 channel audio ADC chips from Texas Instruments. Based on an old driver for these chips that I was referred to by Texas Instruments, but which apparently was never upstreamed, I've taken patches from various incarnations, ported them and augmented them with various new features in order to make a working driver. There is currently no support for the on-chip "miniDSP". Although the TLV320ADC3001 and TLV320ADC3101 have similarities with other TI codec chips for which there are existing drivers, upon closer inspection I decided the differences were too great to warrant attempting to modify an existing driver, especially without access to all supported chips and resources for testing them all. There are certainly things that could be improved, among others in the area of PLL register value calculations, but it has been tested on the Axis ARTPEC-8 platform and has been found to work satisfactorily at least with a TLV320ADC3101 operating as an I2S slave at 48 kHz with 32 bit bit depth, 1 or 2 channels, both with the on-chip PLL enabled and disabled. I have not tested this on a TLV320ADC3001 chip; in the original driver the only differences were related to the now-removed initialization code, so I've just retained the compatible strings for both chips. Limitations: - GPIO pins on chip can only be set up as outputs. - Using BCLK as an input to clock chain is supported by the hw but not by driver. Changes v3->v4: More fixes for issues reported by kernel test robot (lkp(a)intel.com). Don't know why these weren't reported the first time. Changes v2->v3: Fix a couple of issues reported by the kernel test robot (lkp(a)intel.com). Changes v1->v2: Large portions of the driver have been reworked from v1 based on remarks from Mark. More specifically: - Remove an unneeded #ifdef and EXPORT_SYMBOL_GPL. - Add ADC3XXX_ prefix to all macro names to avoid namespace contention. - Clean up certain macro names with mixed lower case - upper case names. - Do away with the D0..D7 macros representing bit values. - Fix general formatting: spaces before and after {, }, replace multi spaces with tabs, maximum 2 blank lines anywhere, and other minor issues. - Page register does not need to be volatile. - Rather than use 0 / -6dB enum for input attenuator, use standard TLV. - The dither offset control contains reserved values so use VALUE_ENUM instead of straight ENUM. - Replace vector of SOC_ENUMs with individual variables. - Fix ALSA control names: - Use Volume at the end of all gain controls, and remove "Gain" where applicable. - Use "Switch" at the end of all switch controls, either adding it or changing "switch" into "Switch" as applicable. - Switches and Volumes related to capture should be set up as such, so add "Capture" to widget names where appropriate. This is mostly important for the PGA Capture and Switch widgets, not just for consistency, but also because amixer simple Switch controls expect different names (cap/nocap instead of on/off) for control values for capture. - Make widget names have capital first letters for all consitutent words. - When resuming, the registers will not have changed, so no need to call snd_soc_component_cache_sync(). - Move reset gpio init to I2C probe - Remove needless _priv at end of private struct name. - Make hex constants lower case - _CBM_CFM/_CBS_CFS -> _CBP_CFP/_CBC_CFC - Instead of using device specific ALSA controls to control the two GPIO pins, use GPIOLIB. GPIO pin functions still can be set up in the devicetree as the pins can be used for other functions which are not GPIO related. - Limitation: The GPIO pins can only be used as outputs, even though the hardware alternatively supports using them as inputs. - Handle GPIO configs in a vector rather than separately. - Use more appropriate module_i2c_driver instead of generic module_init/module_exit. - Remove useless initialization data. - As the ADC mute function had no control, add it. - Clean up GPIO initialization which was done both in the I2C probe and in the component probe. - Remove probe, remove, suspend and resume callbacks, which either do not do anything at all anymore, or uselessly just do set_bias_level. - The PLL should not be configured in the devicetree, so move the corresponding definitions to tlv320adc3xxx.h where they can be picked up by a machine driver if needed. - Default to PLL AUTO mode. - Use "reset-gpios" instead of too-generic "gpios" for dt property for configuring codec reset pin. - Use function for printing PLL mode rather than macro. - dev_info is too verbose for most of the printouts, change them to dev_dbg. - Add support for configuring micbias voltage in DT. - Remove ALSA control for mic bias. - Move PLL and NADC, MADC divisor enable to DAPM, including 10 ms post PLL start up delay via POST_PMU event. - ADC is powered on using DAPM, so needless to do it in set_bias_level, thus remove it. - Let DAPM handle BCLK divisor enable. With this, we can remove set_bias_level as it doesn't do anything anymore. - Fix incorrect argument order to a snd_soc_component_update_bits call. - Although the chip itself can operate using BCLK as the clock source, either directly to the divider chain or via the PLL, this is not yet supported in the driver, so require the clocks property for the MCLK for now. - Move PLL rate structure out of .h file where it does not belong. /Ricard Ricard Wanderlof (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 +++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1071 +++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 379 ++++++ 6 files changed, 1624 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[PATCH 0/2] ASoC: rt9120: Fix settings and add rt9120s compatibility
by cy_huang 09 Nov '21

09 Nov '21

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

3 4

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

09 Nov '21

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

2 2

[PATCH] ALSA: missing check for possible NULL after the call to kstrdup
by Austin Kim 09 Nov '21

09 Nov '21

From: Austin Kim <austin.kim(a)lge.com> If kcalloc() return NULL due to memory starvation, it is possible for kstrdup() to return NULL in similar case. So add null check after the call to kstrdup() is made. Signed-off-by: Austin Kim <austin.kim(a)lge.com> --- sound/synth/emux/emux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index 49d1976a132c..6dab3c5bac00 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -88,7 +88,7 @@ int snd_emux_register(struct snd_emux *emu, struct snd_card *card, int index, ch emu->name = kstrdup(name, GFP_KERNEL); emu->voices = kcalloc(emu->max_voices, sizeof(struct snd_emux_voice), GFP_KERNEL); - if (emu->voices == NULL) + if ((emu->name == NULL) || (emu->voices == NULL)) return -ENOMEM; /* create soundfont list */ -- 2.20.1

2 2

[PATCH] ALSA: memalloc: Unify x86 SG-buffer handling (take#2)
by Takashi Iwai 09 Nov '21

09 Nov '21

This is a second attempt to unify the x86-specific SG-buffer handling code with the new standard non-contiguous page handler. The first try (in commit 2d9ea39917a4) failed due to the wrong page and address calculations, hence reverted. Now it's corrected with the previous fix for noncontig pages, and the proper sg page iteration by this patch. After the migration, SNDRV_DMA_TYPE_DMA_SG becomes identical with SNDRV_DMA_TYPE_NONCONTIG on x86, while others still fall back to SNDRV_DMA_TYPE_DEV. Tested-by: Alex Xu (Hello71) <alex_y_xu(a)yahoo.ca> Link: https://lore.kernel.org/r/20211017074859.24112-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 14 +-- sound/core/Makefile | 1 - sound/core/memalloc.c | 43 ++++++++- sound/core/sgbuf.c | 201 --------------------------------------- 4 files changed, 46 insertions(+), 213 deletions(-) delete mode 100644 sound/core/sgbuf.c diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 1051b84e8579..653dfffb3ac8 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -36,13 +36,6 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */ #define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */ #define SNDRV_DMA_TYPE_DEV_WC 5 /* continuous write-combined */ -#ifdef CONFIG_SND_DMA_SGBUF -#define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */ -#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ -#else -#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ -#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC -#endif #ifdef CONFIG_GENERIC_ALLOCATOR #define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ #else @@ -51,6 +44,13 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_VMALLOC 7 /* vmalloc'ed buffer */ #define SNDRV_DMA_TYPE_NONCONTIG 8 /* non-coherent SG buffer */ #define SNDRV_DMA_TYPE_NONCOHERENT 9 /* non-coherent buffer */ +#ifdef CONFIG_SND_DMA_SGBUF +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_NONCONTIG +#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ +#else +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ +#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC +#endif /* * info for buffer allocation diff --git a/sound/core/Makefile b/sound/core/Makefile index 79e1407cd0de..350d704ced98 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -19,7 +19,6 @@ snd-$(CONFIG_SND_JACK) += ctljack.o jack.o snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \ pcm_memory.o memalloc.o snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o -snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 8c362bb5b81a..29b5c613ca23 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -621,6 +621,42 @@ static const struct snd_malloc_ops snd_dma_noncontig_ops = { .get_chunk_size = snd_dma_noncontig_get_chunk_size, }; +/* x86-specific SG-buffer with WC pages */ +#ifdef CONFIG_SND_DMA_SGBUF +#define sg_wc_address(it) ((unsigned long)page_address(sg_page_iter_page(it))) + +static void *snd_dma_sg_wc_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + void *p = snd_dma_noncontig_alloc(dmab, size); + struct sg_table *sgt = dmab->private_data; + struct sg_page_iter iter; + + if (!p) + return NULL; + for_each_sgtable_page(sgt, &iter, 0) + set_memory_wc(sg_wc_address(&iter), 1); + return p; +} + +static void snd_dma_sg_wc_free(struct snd_dma_buffer *dmab) +{ + struct sg_table *sgt = dmab->private_data; + struct sg_page_iter iter; + + for_each_sgtable_page(sgt, &iter, 0) + set_memory_wb(sg_wc_address(&iter), 1); + snd_dma_noncontig_free(dmab); +} + +static int snd_dma_sg_wc_mmap(struct snd_dma_buffer *dmab, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return dma_mmap_noncontiguous(dmab->dev.dev, area, + dmab->bytes, dmab->private_data); +} +#endif /* CONFIG_SND_DMA_SGBUF */ + /* * Non-coherent pages allocator */ @@ -680,14 +716,13 @@ static const struct snd_malloc_ops *dma_ops[] = { [SNDRV_DMA_TYPE_DEV_WC] = &snd_dma_wc_ops, [SNDRV_DMA_TYPE_NONCONTIG] = &snd_dma_noncontig_ops, [SNDRV_DMA_TYPE_NONCOHERENT] = &snd_dma_noncoherent_ops, +#ifdef CONFIG_SND_DMA_SGBUF + [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_wc_ops, +#endif #ifdef CONFIG_GENERIC_ALLOCATOR [SNDRV_DMA_TYPE_DEV_IRAM] = &snd_dma_iram_ops, #endif /* CONFIG_GENERIC_ALLOCATOR */ #endif /* CONFIG_HAS_DMA */ -#ifdef CONFIG_SND_DMA_SGBUF - [SNDRV_DMA_TYPE_DEV_SG] = &snd_dma_sg_ops, - [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_ops, -#endif }; static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab) diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c deleted file mode 100644 index 8352a5cdb19f..000000000000 --- a/sound/core/sgbuf.c +++ /dev/null @@ -1,201 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* - * Scatter-Gather buffer - * - * Copyright (c) by Takashi Iwai <tiwai(a)suse.de> - */ - -#include <linux/slab.h> -#include <linux/mm.h> -#include <linux/vmalloc.h> -#include <linux/export.h> -#include <sound/memalloc.h> -#include "memalloc_local.h" - -struct snd_sg_page { - void *buf; - dma_addr_t addr; -}; - -struct snd_sg_buf { - int size; /* allocated byte size */ - int pages; /* allocated pages */ - int tblsize; /* allocated table size */ - struct snd_sg_page *table; /* address table */ - struct page **page_table; /* page table (for vmap/vunmap) */ - struct device *dev; -}; - -/* table entries are align to 32 */ -#define SGBUF_TBL_ALIGN 32 -#define sgbuf_align_table(tbl) ALIGN((tbl), SGBUF_TBL_ALIGN) - -static void snd_dma_sg_free(struct snd_dma_buffer *dmab) -{ - struct snd_sg_buf *sgbuf = dmab->private_data; - struct snd_dma_buffer tmpb; - int i; - - if (!sgbuf) - return; - - vunmap(dmab->area); - dmab->area = NULL; - - tmpb.dev.type = SNDRV_DMA_TYPE_DEV; - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) - tmpb.dev.type = SNDRV_DMA_TYPE_DEV_WC; - tmpb.dev.dev = sgbuf->dev; - for (i = 0; i < sgbuf->pages; i++) { - if (!(sgbuf->table[i].addr & ~PAGE_MASK)) - continue; /* continuous pages */ - tmpb.area = sgbuf->table[i].buf; - tmpb.addr = sgbuf->table[i].addr & PAGE_MASK; - tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT; - snd_dma_free_pages(&tmpb); - } - - kfree(sgbuf->table); - kfree(sgbuf->page_table); - kfree(sgbuf); - dmab->private_data = NULL; -} - -#define MAX_ALLOC_PAGES 32 - -static void *snd_dma_sg_alloc(struct snd_dma_buffer *dmab, size_t size) -{ - struct snd_sg_buf *sgbuf; - unsigned int i, pages, chunk, maxpages; - struct snd_dma_buffer tmpb; - struct snd_sg_page *table; - struct page **pgtable; - int type = SNDRV_DMA_TYPE_DEV; - pgprot_t prot = PAGE_KERNEL; - void *area; - - dmab->private_data = sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); - if (!sgbuf) - return NULL; - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) { - type = SNDRV_DMA_TYPE_DEV_WC; -#ifdef pgprot_noncached - prot = pgprot_noncached(PAGE_KERNEL); -#endif - } - sgbuf->dev = dmab->dev.dev; - pages = snd_sgbuf_aligned_pages(size); - sgbuf->tblsize = sgbuf_align_table(pages); - table = kcalloc(sgbuf->tblsize, sizeof(*table), GFP_KERNEL); - if (!table) - goto _failed; - sgbuf->table = table; - pgtable = kcalloc(sgbuf->tblsize, sizeof(*pgtable), GFP_KERNEL); - if (!pgtable) - goto _failed; - sgbuf->page_table = pgtable; - - /* allocate pages */ - maxpages = MAX_ALLOC_PAGES; - while (pages > 0) { - chunk = pages; - /* don't be too eager to take a huge chunk */ - if (chunk > maxpages) - chunk = maxpages; - chunk <<= PAGE_SHIFT; - if (snd_dma_alloc_pages_fallback(type, dmab->dev.dev, - chunk, &tmpb) < 0) { - if (!sgbuf->pages) - goto _failed; - size = sgbuf->pages * PAGE_SIZE; - break; - } - chunk = tmpb.bytes >> PAGE_SHIFT; - for (i = 0; i < chunk; i++) { - table->buf = tmpb.area; - table->addr = tmpb.addr; - if (!i) - table->addr |= chunk; /* mark head */ - table++; - *pgtable++ = virt_to_page(tmpb.area); - tmpb.area += PAGE_SIZE; - tmpb.addr += PAGE_SIZE; - } - sgbuf->pages += chunk; - pages -= chunk; - if (chunk < maxpages) - maxpages = chunk; - } - - sgbuf->size = size; - area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, prot); - if (!area) - goto _failed; - return area; - - _failed: - snd_dma_sg_free(dmab); /* free the table */ - return NULL; -} - -static dma_addr_t snd_dma_sg_get_addr(struct snd_dma_buffer *dmab, - size_t offset) -{ - struct snd_sg_buf *sgbuf = dmab->private_data; - dma_addr_t addr; - - addr = sgbuf->table[offset >> PAGE_SHIFT].addr; - addr &= ~((dma_addr_t)PAGE_SIZE - 1); - return addr + offset % PAGE_SIZE; -} - -static struct page *snd_dma_sg_get_page(struct snd_dma_buffer *dmab, - size_t offset) -{ - struct snd_sg_buf *sgbuf = dmab->private_data; - unsigned int idx = offset >> PAGE_SHIFT; - - if (idx >= (unsigned int)sgbuf->pages) - return NULL; - return sgbuf->page_table[idx]; -} - -static unsigned int snd_dma_sg_get_chunk_size(struct snd_dma_buffer *dmab, - unsigned int ofs, - unsigned int size) -{ - struct snd_sg_buf *sg = dmab->private_data; - unsigned int start, end, pg; - - start = ofs >> PAGE_SHIFT; - end = (ofs + size - 1) >> PAGE_SHIFT; - /* check page continuity */ - pg = sg->table[start].addr >> PAGE_SHIFT; - for (;;) { - start++; - if (start > end) - break; - pg++; - if ((sg->table[start].addr >> PAGE_SHIFT) != pg) - return (start << PAGE_SHIFT) - ofs; - } - /* ok, all on continuous pages */ - return size; -} - -static int snd_dma_sg_mmap(struct snd_dma_buffer *dmab, - struct vm_area_struct *area) -{ - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); - return -ENOENT; /* continue with the default mmap handler */ -} - -const struct snd_malloc_ops snd_dma_sg_ops = { - .alloc = snd_dma_sg_alloc, - .free = snd_dma_sg_free, - .get_addr = snd_dma_sg_get_addr, - .get_page = snd_dma_sg_get_page, - .get_chunk_size = snd_dma_sg_get_chunk_size, - .mmap = snd_dma_sg_mmap, -}; -- 2.26.2

1 0

[PATCH v0 42/42] notifier: Return an error when callback is already registered
by Borislav Petkov 08 Nov '21

08 Nov '21

From: Borislav Petkov <bp(a)suse.de> The notifier registration routine doesn't return a proper error value when a callback has already been registered, leading people to track whether that registration has happened at the call site: https://lore.kernel.org/amd-gfx/20210512013058.6827-1-mukul.joshi@amd.com/ Which is unnecessary. Return -EEXIST to signal that case so that callers can act accordingly. Enforce callers to check the return value, leading to loud screaming during build: arch/x86/kernel/cpu/mce/core.c: In function ‘mce_register_decode_chain’: arch/x86/kernel/cpu/mce/core.c:167:2: error: ignoring return value of \ ‘blocking_notifier_chain_register’, declared with attribute warn_unused_result [-Werror=unused-result] blocking_notifier_chain_register(&x86_mce_decoder_chain, nb); ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Drop the WARN too, while at it. Suggested-by: Thomas Gleixner <tglx(a)linutronix.de> Signed-off-by: Borislav Petkov <bp(a)suse.de> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Ayush Sawal <ayush.sawal(a)chelsio.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Rohit Maheshwari <rohitm(a)chelsio.com> Cc: Steven Rostedt <rostedt(a)goodmis.org> Cc: Vinay Kumar Yadav <vinay.yadav(a)chelsio.com> Cc: alsa-devel(a)alsa-project.org Cc: bcm-kernel-feedback-list(a)broadcom.com Cc: intel-gfx(a)lists.freedesktop.org Cc: intel-gvt-dev(a)lists.freedesktop.org Cc: linux-alpha(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-clk(a)vger.kernel.org Cc: linux-crypto(a)vger.kernel.org Cc: linux-edac(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: linux-hyperv(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-leds(a)vger.kernel.org Cc: linux-mips(a)vger.kernel.org Cc: linux-parisc(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-remoteproc(a)vger.kernel.org Cc: linux-renesas-soc(a)vger.kernel.org Cc: linux-s390(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: linux-sh(a)vger.kernel.org Cc: linux-staging(a)lists.linux.dev Cc: linux-tegra(a)vger.kernel.org Cc: linux-um(a)lists.infradead.org Cc: linux-usb(a)vger.kernel.org Cc: linux-xtensa(a)linux-xtensa.org Cc: netdev(a)vger.kernel.org Cc: openipmi-developer(a)lists.sourceforge.net Cc: rcu(a)vger.kernel.org Cc: sparclinux(a)vger.kernel.org Cc: x86(a)kernel.org Cc: xen-devel(a)lists.xenproject.org --- include/linux/notifier.h | 8 ++++---- kernel/notifier.c | 36 +++++++++++++++++++----------------- 2 files changed, 23 insertions(+), 21 deletions(-) diff --git a/include/linux/notifier.h b/include/linux/notifier.h index 87069b8459af..45cc5a8d0fd8 100644 --- a/include/linux/notifier.h +++ b/include/linux/notifier.h @@ -141,13 +141,13 @@ extern void srcu_init_notifier_head(struct srcu_notifier_head *nh); #ifdef __KERNEL__ -extern int atomic_notifier_chain_register(struct atomic_notifier_head *nh, +extern int __must_check atomic_notifier_chain_register(struct atomic_notifier_head *nh, struct notifier_block *nb); -extern int blocking_notifier_chain_register(struct blocking_notifier_head *nh, +extern int __must_check blocking_notifier_chain_register(struct blocking_notifier_head *nh, struct notifier_block *nb); -extern int raw_notifier_chain_register(struct raw_notifier_head *nh, +extern int __must_check raw_notifier_chain_register(struct raw_notifier_head *nh, struct notifier_block *nb); -extern int srcu_notifier_chain_register(struct srcu_notifier_head *nh, +extern int __must_check srcu_notifier_chain_register(struct srcu_notifier_head *nh, struct notifier_block *nb); extern int atomic_notifier_chain_unregister(struct atomic_notifier_head *nh, diff --git a/kernel/notifier.c b/kernel/notifier.c index b8251dc0bc0f..451ef3f73ad2 100644 --- a/kernel/notifier.c +++ b/kernel/notifier.c @@ -20,13 +20,11 @@ BLOCKING_NOTIFIER_HEAD(reboot_notifier_list); */ static int notifier_chain_register(struct notifier_block **nl, - struct notifier_block *n) + struct notifier_block *n) { while ((*nl) != NULL) { - if (unlikely((*nl) == n)) { - WARN(1, "double register detected"); - return 0; - } + if (unlikely((*nl) == n)) + return -EEXIST; if (n->priority > (*nl)->priority) break; nl = &((*nl)->next); @@ -134,10 +132,11 @@ static int notifier_call_chain_robust(struct notifier_block **nl, * * Adds a notifier to an atomic notifier chain. * - * Currently always returns zero. + * Returns 0 on success, %-EEXIST on error. */ -int atomic_notifier_chain_register(struct atomic_notifier_head *nh, - struct notifier_block *n) +int __must_check +atomic_notifier_chain_register(struct atomic_notifier_head *nh, + struct notifier_block *n) { unsigned long flags; int ret; @@ -216,10 +215,11 @@ NOKPROBE_SYMBOL(atomic_notifier_call_chain); * Adds a notifier to a blocking notifier chain. * Must be called in process context. * - * Currently always returns zero. + * Returns 0 on success, %-EEXIST on error. */ -int blocking_notifier_chain_register(struct blocking_notifier_head *nh, - struct notifier_block *n) +int __must_check +blocking_notifier_chain_register(struct blocking_notifier_head *nh, + struct notifier_block *n) { int ret; @@ -335,10 +335,11 @@ EXPORT_SYMBOL_GPL(blocking_notifier_call_chain); * Adds a notifier to a raw notifier chain. * All locking must be provided by the caller. * - * Currently always returns zero. + * Returns 0 on success, %-EEXIST on error. */ -int raw_notifier_chain_register(struct raw_notifier_head *nh, - struct notifier_block *n) +int __must_check +raw_notifier_chain_register(struct raw_notifier_head *nh, + struct notifier_block *n) { return notifier_chain_register(&nh->head, n); } @@ -406,10 +407,11 @@ EXPORT_SYMBOL_GPL(raw_notifier_call_chain); * Adds a notifier to an SRCU notifier chain. * Must be called in process context. * - * Currently always returns zero. + * Returns 0 on success, %-EEXIST on error. */ -int srcu_notifier_chain_register(struct srcu_notifier_head *nh, - struct notifier_block *n) +int __must_check +srcu_notifier_chain_register(struct srcu_notifier_head *nh, + struct notifier_block *n) { int ret; -- 2.29.2

3 8

[PATCH v0 00/42] notifiers: Return an error when callback is already registered
by Borislav Petkov 08 Nov '21

08 Nov '21

From: Borislav Petkov <bp(a)suse.de> Hi all, this is a huge patchset for something which is really trivial - it changes the notifier registration routines to return an error value if a notifier callback is already present on the respective list of callbacks. For more details scroll to the last patch. Everything before it is converting the callers to check the return value of the registration routines and issue a warning, instead of the WARN() notifier_chain_register() does now. Before the last patch has been applied, though, that checking is a NOP which would make the application of those patches trivial - every maintainer can pick a patch at her/his discretion - only the last one enables the build warnings and that one will be queued only after the preceding patches have all been merged so that there are no build warnings. Due to the sheer volume of the patches, I have addressed the respective patch and the last one, which enables the warning, with addressees for each maintained area so as not to spam people unnecessarily. If people prefer I carry some through tip, instead, I'll gladly do so - your call. And, if you think the warning messages need to be more precise, feel free to adjust them before committing. Thanks! Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Ayush Sawal <ayush.sawal(a)chelsio.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Rohit Maheshwari <rohitm(a)chelsio.com> Cc: Steven Rostedt <rostedt(a)goodmis.org> Cc: Vinay Kumar Yadav <vinay.yadav(a)chelsio.com> Cc: alsa-devel(a)alsa-project.org Cc: bcm-kernel-feedback-list(a)broadcom.com Cc: intel-gfx(a)lists.freedesktop.org Cc: intel-gvt-dev(a)lists.freedesktop.org Cc: linux-alpha(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-clk(a)vger.kernel.org Cc: linux-crypto(a)vger.kernel.org Cc: linux-edac(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: linux-hyperv(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-leds(a)vger.kernel.org Cc: linux-mips(a)vger.kernel.org Cc: linux-parisc(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-remoteproc(a)vger.kernel.org Cc: linux-renesas-soc(a)vger.kernel.org Cc: linux-s390(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: linux-sh(a)vger.kernel.org Cc: linux-staging(a)lists.linux.dev Cc: linux-tegra(a)vger.kernel.org Cc: linux-um(a)lists.infradead.org Cc: linux-usb(a)vger.kernel.org Cc: linux-xtensa(a)linux-xtensa.org Cc: netdev(a)vger.kernel.org Cc: openipmi-developer(a)lists.sourceforge.net Cc: rcu(a)vger.kernel.org Cc: sparclinux(a)vger.kernel.org Cc: x86(a)kernel.org Cc: xen-devel(a)lists.xenproject.org Borislav Petkov (42): x86: Check notifier registration return value xen/x86: Check notifier registration return value impi: Check notifier registration return value clk: renesas: Check notifier registration return value dca: Check notifier registration return value firmware: Check notifier registration return value drm/i915: Check notifier registration return value Drivers: hv: vmbus: Check notifier registration return value iio: proximity: cros_ec: Check notifier registration return value leds: trigger: Check notifier registration return value misc: Check notifier registration return value ethernet: chelsio: Check notifier registration return value power: reset: Check notifier registration return value remoteproc: Check notifier registration return value scsi: target: Check notifier registration return value USB: Check notifier registration return value drivers: video: Check notifier registration return value drivers/xen: Check notifier registration return value kernel/hung_task: Check notifier registration return value rcu: Check notifier registration return value tracing: Check notifier registration return value net: fib_notifier: Check notifier registration return value ASoC: soc-jack: Check notifier registration return value staging: olpc_dcon: Check notifier registration return value arch/um: Check notifier registration return value alpha: Check notifier registration return value bus: brcmstb_gisb: Check notifier registration return value soc: bcm: brcmstb: pm: pm-arm: Check notifier registration return value arm64: Check notifier registration return value soc/tegra: Check notifier registration return value parisc: Check notifier registration return value macintosh/adb: Check notifier registration return value mips: Check notifier registration return value powerpc: Check notifier registration return value sh: Check notifier registration return value s390: Check notifier registration return value sparc: Check notifier registration return value xtensa: Check notifier registration return value crypto: ccree - check notifier registration return value EDAC/altera: Check notifier registration return value power: supply: ab8500: Check notifier registration return value notifier: Return an error when callback is already registered arch/alpha/kernel/setup.c | 5 +-- arch/arm64/kernel/setup.c | 6 ++-- arch/mips/kernel/relocate.c | 6 ++-- arch/mips/sgi-ip22/ip22-reset.c | 4 ++- arch/mips/sgi-ip32/ip32-reset.c | 4 ++- arch/parisc/kernel/pdc_chassis.c | 5 +-- arch/powerpc/kernel/setup-common.c | 12 ++++--- arch/s390/kernel/ipl.c | 4 ++- arch/s390/kvm/kvm-s390.c | 7 ++-- arch/sh/kernel/cpu/sh4a/setup-sh7724.c | 11 +++--- arch/sparc/kernel/sstate.c | 6 ++-- arch/um/drivers/mconsole_kern.c | 6 ++-- arch/um/kernel/um_arch.c | 5 +-- arch/x86/kernel/cpu/mce/core.c | 3 +- arch/x86/kernel/cpu/mce/dev-mcelog.c | 3 +- arch/x86/kernel/setup.c | 7 ++-- arch/x86/xen/enlighten.c | 4 ++- arch/xtensa/platforms/iss/setup.c | 3 +- drivers/bus/brcmstb_gisb.c | 6 ++-- drivers/char/ipmi/ipmi_msghandler.c | 3 +- drivers/clk/renesas/clk-div6.c | 4 ++- drivers/clk/renesas/rcar-cpg-lib.c | 4 ++- drivers/crypto/ccree/cc_fips.c | 4 ++- drivers/dca/dca-core.c | 3 +- drivers/edac/altera_edac.c | 6 ++-- drivers/firmware/arm_scmi/notify.c | 3 +- drivers/firmware/google/gsmi.c | 6 ++-- drivers/gpu/drm/i915/gvt/scheduler.c | 6 ++-- drivers/hv/vmbus_drv.c | 4 +-- .../iio/proximity/cros_ec_mkbp_proximity.c | 3 +- drivers/leds/trigger/ledtrig-activity.c | 6 ++-- drivers/leds/trigger/ledtrig-heartbeat.c | 6 ++-- drivers/leds/trigger/ledtrig-panic.c | 4 +-- drivers/macintosh/adbhid.c | 4 +-- drivers/misc/ibmasm/heartbeat.c | 3 +- drivers/misc/pvpanic/pvpanic.c | 3 +- .../chelsio/inline_crypto/chtls/chtls_main.c | 5 ++- drivers/parisc/power.c | 5 +-- drivers/power/reset/ltc2952-poweroff.c | 6 ++-- drivers/power/supply/ab8500_charger.c | 8 ++--- drivers/remoteproc/qcom_common.c | 3 +- drivers/remoteproc/qcom_sysmon.c | 4 ++- drivers/remoteproc/remoteproc_core.c | 4 ++- drivers/s390/char/con3215.c | 5 ++- drivers/s390/char/con3270.c | 5 ++- drivers/s390/char/sclp_con.c | 4 ++- drivers/s390/char/sclp_vt220.c | 4 ++- drivers/s390/char/zcore.c | 4 ++- drivers/soc/bcm/brcmstb/pm/pm-arm.c | 5 +-- drivers/soc/tegra/ari-tegra186.c | 7 ++-- drivers/staging/olpc_dcon/olpc_dcon.c | 4 ++- drivers/target/tcm_fc/tfc_conf.c | 4 ++- drivers/usb/core/notify.c | 3 +- drivers/video/console/dummycon.c | 3 +- drivers/video/fbdev/hyperv_fb.c | 5 +-- drivers/xen/manage.c | 3 +- drivers/xen/xenbus/xenbus_probe.c | 8 +++-- include/linux/notifier.h | 8 ++--- kernel/hung_task.c | 3 +- kernel/notifier.c | 36 ++++++++++--------- kernel/rcu/tree_stall.h | 4 ++- kernel/trace/trace.c | 4 +-- net/core/fib_notifier.c | 4 ++- sound/soc/soc-jack.c | 3 +- 64 files changed, 222 insertions(+), 118 deletions(-) -- 2.29.2

3 5

[PATCH v2 00/10] Fix kcontrol put callback in Tegra drivers
by Sameer Pujar 08 Nov '21

08 Nov '21

This series fixes kcontrol put callback in some of the Tegra drivers which are used on platforms based on Tegra210 and later. The callback is expected to return 1 whenever the HW update is done. This idea is suggested by Jaroslav. Similar suggestion came from Mark during review of series [0] and drivers under this were updated to return 1, but missed to take care of duplicate updates. This series updates all concerned drivers to return proper values and duplicate updates are filtered out. I have added 'Suggested-by" tags accordingly. [0] https://lore.kernel.org/linux-arm-kernel/20210913142307.GF4283@sirena.org.u… Changelog ========= v1->v2: ------- * ADMAIF, I2S, DMIC and DSPK drivers updated to take care of duplicate updates. * Similarly new patches are added for AHUB, MVC, SFC, AMX, ADX and Mixer drivers. Sameer Pujar (10): ASoC: tegra: Fix kcontrol put callback in ADMAIF ASoC: tegra: Fix kcontrol put callback in I2S ASoC: tegra: Fix kcontrol put callback in DMIC ASoC: tegra: Fix kcontrol put callback in DSPK ASoC: tegra: Fix kcontrol put callback in AHUB ASoC: tegra: Fix kcontrol put callback in MVC ASoC: tegra: Fix kcontrol put callback in SFC ASoC: tegra: Fix kcontrol put callback in AMX ASoC: tegra: Fix kcontrol put callback in ADX ASoC: tegra: Fix kcontrol put callback in Mixer sound/soc/tegra/tegra186_dspk.c | 33 ++++++++++++++++++++++++++------- sound/soc/tegra/tegra210_admaif.c | 23 ++++++++++++++++++----- sound/soc/tegra/tegra210_adx.c | 3 +++ sound/soc/tegra/tegra210_ahub.c | 11 +++++++---- sound/soc/tegra/tegra210_amx.c | 3 +++ sound/soc/tegra/tegra210_dmic.c | 35 +++++++++++++++++++++++++++-------- sound/soc/tegra/tegra210_i2s.c | 26 +++++++++++++++++++++++++- sound/soc/tegra/tegra210_mixer.c | 3 +++ sound/soc/tegra/tegra210_mvc.c | 18 ++++++++++++++++-- sound/soc/tegra/tegra210_sfc.c | 23 +++++++++++++++++------ 10 files changed, 145 insertions(+), 33 deletions(-) -- 2.7.4

4 16

[PATCH] ALSA: memalloc: Use proper SG helpers for noncontig allocations
by Takashi Iwai 08 Nov '21

08 Nov '21

The recently introduced non-contiguous page allocation support helpers are using the simplified code to calculate the page and DMA address based on the vmalloc helpers, but this isn't quite right as the vmap is valid only for the direct DMA. This patch corrects those accessors to use the proper SG helpers instead. Fixes: a25684a95646 ("ALSA: memalloc: Support for non-contiguous page allocation") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 64 +++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 61 insertions(+), 3 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index ad4a76b47b92..8c362bb5b81a 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -552,15 +552,73 @@ static void snd_dma_noncontig_sync(struct snd_dma_buffer *dmab, } } +static inline void snd_dma_noncontig_iter_set(struct snd_dma_buffer *dmab, + struct sg_page_iter *piter, + size_t offset) +{ + struct sg_table *sgt = dmab->private_data; + + __sg_page_iter_start(piter, sgt->sgl, sgt->orig_nents, + offset >> PAGE_SHIFT); +} + +static dma_addr_t snd_dma_noncontig_get_addr(struct snd_dma_buffer *dmab, + size_t offset) +{ + struct sg_dma_page_iter iter; + + snd_dma_noncontig_iter_set(dmab, &iter.base, offset); + __sg_page_iter_dma_next(&iter); + return sg_page_iter_dma_address(&iter) + offset % PAGE_SIZE; +} + +static struct page *snd_dma_noncontig_get_page(struct snd_dma_buffer *dmab, + size_t offset) +{ + struct sg_page_iter iter; + + snd_dma_noncontig_iter_set(dmab, &iter, offset); + __sg_page_iter_next(&iter); + return sg_page_iter_page(&iter); +} + +static unsigned int +snd_dma_noncontig_get_chunk_size(struct snd_dma_buffer *dmab, + unsigned int ofs, unsigned int size) +{ + struct sg_dma_page_iter iter; + unsigned int start, end; + unsigned long addr; + + start = ALIGN_DOWN(ofs, PAGE_SIZE); + end = ofs + size - 1; /* the last byte address */ + snd_dma_noncontig_iter_set(dmab, &iter.base, start); + if (!__sg_page_iter_dma_next(&iter)) + return 0; + /* check page continuity */ + addr = sg_page_iter_dma_address(&iter); + for (;;) { + start += PAGE_SIZE; + if (start > end) + break; + addr += PAGE_SIZE; + if (!__sg_page_iter_dma_next(&iter) || + sg_page_iter_dma_address(&iter) != addr) + return start - ofs; + } + /* ok, all on continuous pages */ + return size; +} + static const struct snd_malloc_ops snd_dma_noncontig_ops = { .alloc = snd_dma_noncontig_alloc, .free = snd_dma_noncontig_free, .mmap = snd_dma_noncontig_mmap, .sync = snd_dma_noncontig_sync, /* re-use vmalloc helpers for get_* ops */ - .get_addr = snd_dma_vmalloc_get_addr, - .get_page = snd_dma_vmalloc_get_page, - .get_chunk_size = snd_dma_vmalloc_get_chunk_size, + .get_addr = snd_dma_noncontig_get_addr, + .get_page = snd_dma_noncontig_get_page, + .get_chunk_size = snd_dma_noncontig_get_chunk_size, }; /* -- 2.26.2

1 0

[PATCH] ALSA: pci: rme: Fix unaligned buffer addresses
by Takashi Iwai 08 Nov '21

08 Nov '21

The recent fix for setting up the DMA buffer type on RME drivers tried to address the non-standard memory managements and changed the DMA buffer information to the standard snd_dma_buffer object that is allocated at the probe time. However, I overlooked that the RME drivers handle the buffer addresses based on 64k alignment, and the previous conversion broke that silently. This patch is an attempt to fix the regression. The snd_dma_buffer objects are copied to the original data with the correction to the aligned accesses, and those are passed to snd_pcm_set_runtime_buffer() helpers instead. The original snd_dma_buffer objects are managed by devres, hence they'll be released automagically. Fixes: 0899a7a23047 ("ALSA: pci: rme: Set up buffer type properly") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/rme9652/hdsp.c | 41 +++++++++++++++++++++---------------- sound/pci/rme9652/rme9652.c | 41 +++++++++++++++++++++---------------- 2 files changed, 46 insertions(+), 36 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 75aa2ea733a5..96c12dfb24cf 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -468,8 +468,11 @@ struct hdsp { unsigned char ss_out_channels; u32 io_loopback; /* output loopback channel states*/ - struct snd_dma_buffer *capture_dma_buf; - struct snd_dma_buffer *playback_dma_buf; + /* DMA buffers; those are copied instances from the original snd_dma_buf + * objects (which are managed via devres) for the address alignments + */ + struct snd_dma_buffer capture_dma_buf; + struct snd_dma_buffer playback_dma_buf; unsigned char *capture_buffer; /* suitably aligned address */ unsigned char *playback_buffer; /* suitably aligned address */ @@ -3764,30 +3767,32 @@ static void snd_hdsp_proc_init(struct hdsp *hdsp) static int snd_hdsp_initialize_memory(struct hdsp *hdsp) { - unsigned long pb_bus, cb_bus; + struct snd_dma_buffer *capture_dma, *playback_dma; - hdsp->capture_dma_buf = - snd_hammerfall_get_buffer(hdsp->pci, HDSP_DMA_AREA_BYTES); - hdsp->playback_dma_buf = - snd_hammerfall_get_buffer(hdsp->pci, HDSP_DMA_AREA_BYTES); - if (!hdsp->capture_dma_buf || !hdsp->playback_dma_buf) { + capture_dma = snd_hammerfall_get_buffer(hdsp->pci, HDSP_DMA_AREA_BYTES); + playback_dma = snd_hammerfall_get_buffer(hdsp->pci, HDSP_DMA_AREA_BYTES); + if (!capture_dma || !playback_dma) { dev_err(hdsp->card->dev, "%s: no buffers available\n", hdsp->card_name); return -ENOMEM; } - /* Align to bus-space 64K boundary */ + /* copy to the own data for alignment */ + hdsp->capture_dma_buf = *capture_dma; + hdsp->playback_dma_buf = *playback_dma; - cb_bus = ALIGN(hdsp->capture_dma_buf->addr, 0x10000ul); - pb_bus = ALIGN(hdsp->playback_dma_buf->addr, 0x10000ul); + /* Align to bus-space 64K boundary */ + hdsp->capture_dma_buf.addr = ALIGN(capture_dma->addr, 0x10000ul); + hdsp->playback_dma_buf.addr = ALIGN(playback_dma->addr, 0x10000ul); /* Tell the card where it is */ + hdsp_write(hdsp, HDSP_inputBufferAddress, hdsp->capture_dma_buf.addr); + hdsp_write(hdsp, HDSP_outputBufferAddress, hdsp->playback_dma_buf.addr); - hdsp_write(hdsp, HDSP_inputBufferAddress, cb_bus); - hdsp_write(hdsp, HDSP_outputBufferAddress, pb_bus); - - hdsp->capture_buffer = hdsp->capture_dma_buf->area + (cb_bus - hdsp->capture_dma_buf->addr); - hdsp->playback_buffer = hdsp->playback_dma_buf->area + (pb_bus - hdsp->playback_dma_buf->addr); + hdsp->capture_dma_buf.area += hdsp->capture_dma_buf.addr - capture_dma->addr; + hdsp->playback_dma_buf.area += hdsp->playback_dma_buf.addr - playback_dma->addr; + hdsp->capture_buffer = hdsp->capture_dma_buf.area; + hdsp->playback_buffer = hdsp->playback_dma_buf.area; return 0; } @@ -4507,7 +4512,7 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); runtime->hw = snd_hdsp_playback_subinfo; - snd_pcm_set_runtime_buffer(substream, hdsp->playback_dma_buf); + snd_pcm_set_runtime_buffer(substream, &hdsp->playback_dma_buf); hdsp->playback_pid = current->pid; hdsp->playback_substream = substream; @@ -4583,7 +4588,7 @@ static int snd_hdsp_capture_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); runtime->hw = snd_hdsp_capture_subinfo; - snd_pcm_set_runtime_buffer(substream, hdsp->capture_dma_buf); + snd_pcm_set_runtime_buffer(substream, &hdsp->capture_dma_buf); hdsp->capture_pid = current->pid; hdsp->capture_substream = substream; diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index e76f737ac9e8..7755e19aa776 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -208,8 +208,11 @@ struct snd_rme9652 { unsigned char ds_channels; unsigned char ss_channels; /* different for hammerfall/hammerfall-light */ - struct snd_dma_buffer *playback_dma_buf; - struct snd_dma_buffer *capture_dma_buf; + /* DMA buffers; those are copied instances from the original snd_dma_buf + * objects (which are managed via devres) for the address alignments + */ + struct snd_dma_buffer playback_dma_buf; + struct snd_dma_buffer capture_dma_buf; unsigned char *capture_buffer; /* suitably aligned address */ unsigned char *playback_buffer; /* suitably aligned address */ @@ -1719,30 +1722,32 @@ static void snd_rme9652_card_free(struct snd_card *card) static int snd_rme9652_initialize_memory(struct snd_rme9652 *rme9652) { - unsigned long pb_bus, cb_bus; + struct snd_dma_buffer *capture_dma, *playback_dma; - rme9652->capture_dma_buf = - snd_hammerfall_get_buffer(rme9652->pci, RME9652_DMA_AREA_BYTES); - rme9652->playback_dma_buf = - snd_hammerfall_get_buffer(rme9652->pci, RME9652_DMA_AREA_BYTES); - if (!rme9652->capture_dma_buf || !rme9652->playback_dma_buf) { + capture_dma = snd_hammerfall_get_buffer(rme9652->pci, RME9652_DMA_AREA_BYTES); + playback_dma = snd_hammerfall_get_buffer(rme9652->pci, RME9652_DMA_AREA_BYTES); + if (!capture_dma || !playback_dma) { dev_err(rme9652->card->dev, "%s: no buffers available\n", rme9652->card_name); return -ENOMEM; } - /* Align to bus-space 64K boundary */ + /* copy to the own data for alignment */ + rme9652->capture_dma_buf = *capture_dma; + rme9652->playback_dma_buf = *playback_dma; - cb_bus = ALIGN(rme9652->capture_dma_buf->addr, 0x10000ul); - pb_bus = ALIGN(rme9652->playback_dma_buf->addr, 0x10000ul); + /* Align to bus-space 64K boundary */ + rme9652->capture_dma_buf.addr = ALIGN(capture_dma->addr, 0x10000ul); + rme9652->playback_dma_buf.addr = ALIGN(playback_dma->addr, 0x10000ul); /* Tell the card where it is */ + rme9652_write(rme9652, RME9652_rec_buffer, rme9652->capture_dma_buf.addr); + rme9652_write(rme9652, RME9652_play_buffer, rme9652->playback_dma_buf.addr); - rme9652_write(rme9652, RME9652_rec_buffer, cb_bus); - rme9652_write(rme9652, RME9652_play_buffer, pb_bus); - - rme9652->capture_buffer = rme9652->capture_dma_buf->area + (cb_bus - rme9652->capture_dma_buf->addr); - rme9652->playback_buffer = rme9652->playback_dma_buf->area + (pb_bus - rme9652->playback_dma_buf->addr); + rme9652->capture_dma_buf.area += rme9652->capture_dma_buf.addr - capture_dma->addr; + rme9652->playback_dma_buf.area += rme9652->playback_dma_buf.addr - playback_dma->addr; + rme9652->capture_buffer = rme9652->capture_dma_buf.area; + rme9652->playback_buffer = rme9652->playback_dma_buf.area; return 0; } @@ -2259,7 +2264,7 @@ static int snd_rme9652_playback_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); runtime->hw = snd_rme9652_playback_subinfo; - snd_pcm_set_runtime_buffer(substream, rme9652->playback_dma_buf); + snd_pcm_set_runtime_buffer(substream, &rme9652->playback_dma_buf); if (rme9652->capture_substream == NULL) { rme9652_stop(rme9652); @@ -2318,7 +2323,7 @@ static int snd_rme9652_capture_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); runtime->hw = snd_rme9652_capture_subinfo; - snd_pcm_set_runtime_buffer(substream, rme9652->capture_dma_buf); + snd_pcm_set_runtime_buffer(substream, &rme9652->capture_dma_buf); if (rme9652->playback_substream == NULL) { rme9652_stop(rme9652); -- 2.31.1

1 0

[PATCH v4 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 08 Nov '21

08 Nov '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. This patch set depends on the dt-bindings header patch -- https://patchwork.kernel.org/project/alsa-devel/list/?series=543829 Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver Changes Since V3: -- Change audio jack playpause key value. Changes Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Changes Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 170 ++++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7280.c | 343 +++++++++++++++++++++ 4 files changed, 527 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

5 13

[PATCH v2] ASoC: Intel: add sof-nau8825 machine driver
by Mac Chiang 08 Nov '21

08 Nov '21

From: David Lin <CTLIN0(a)nuvoton.com> The machine driver is a generic machine driver for SOF with nau8825 codec w or w/o speaker additionally. Depending on the SOC HDMI, DMIC, Bluetooth offload support are added dynamically. Only add information related to SOF since the machine driver was only tested with SOF. There are currently 4 i2s machine variants of ADL. This supports the headphone NUA8825(SSP0) alone or with smart or dumb speakers. Board 2,3,4 use SSP2 for Bluetooth offload support except board 1. Board 1 : NAU8825 + RT1019P(SSP2) Board 2 : NAU8825 + MAX98373(SSP1) Board 3 : NAU8825 + MAX98360A(SSP1) Board 4 : NAU8825 Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> Co-developed-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- 1. remove hdac_hdmi relevant codes support 2. remove duplicated max_98360a_dai_link calls. Instead of checking no_amp condition 3. alter module license name to "GPL" 4. due to rt1019 non-i2c mode, so change to SND_SOC_RT1015P in KConfig --- sound/soc/intel/boards/Kconfig | 19 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/sof_nau8825.c | 651 ++++++++++++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 35 + 4 files changed, 707 insertions(+) create mode 100644 sound/soc/intel/boards/sof_nau8825.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5a225a4ce604..31d2f10672fd 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -526,6 +526,25 @@ config SND_SOC_INTEL_SOF_ES8336_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_SOF_NAU8825_MACH + tristate "SOF with nau8825 codec in I2S Mode" + depends on I2C && ACPI && GPIOLIB + depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ + (MFD_INTEL_LPSS || COMPILE_TEST)) + select SND_SOC_NAU8825 + select SND_SOC_RT1015P + select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_HDA_DSP_COMMON + select SND_SOC_INTEL_SOF_MAXIM_COMMON + help + This adds support for ASoC machine driver for SOF platforms + with nau8825 codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL if (SND_SOC_SOF_COMETLAKE && SND_SOC_SOF_HDA_LINK) diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 9ee8ed864f5d..3ea273d27168 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -22,6 +22,7 @@ snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o snd-soc-sof_rt5682-objs := sof_rt5682.o sof_realtek_common.o snd-soc-sof_cs42l42-objs := sof_cs42l42.o snd-soc-sof_es8336-objs := sof_es8336.o +snd-soc-sof_nau8825-objs := sof_nau8825.o sof_realtek_common.o snd-soc-cml_rt1011_rt5682-objs := cml_rt1011_rt5682.o snd-soc-kbl_da7219_max98357a-objs := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-objs := kbl_da7219_max98927.o @@ -44,6 +45,7 @@ snd-soc-sof-sdw-objs += sof_sdw.o \ obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH) += snd-soc-sof_cs42l42.o obj-$(CONFIG_SND_SOC_INTEL_SOF_ES8336_MACH) += snd-soc-sof_es8336.o +obj-$(CONFIG_SND_SOC_INTEL_SOF_NAU8825_MACH) += snd-soc-sof_nau8825.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c new file mode 100644 index 000000000000..33de043b66c6 --- /dev/null +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -0,0 +1,651 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright(c) 2021 Intel Corporation. +// Copyright(c) 2021 Nuvoton Corporation. + +/* + * Intel SOF Machine Driver with Nuvoton headphone codec NAU8825 + * and speaker codec RT1019P MAX98360a or MAX98373 + */ +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/dmi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/sof.h> +#include <sound/soc-acpi.h> +#include "../../codecs/nau8825.h" +#include "../common/soc-intel-quirks.h" +#include "hda_dsp_common.h" +#include "sof_realtek_common.h" +#include "sof_maxim_common.h" + +#define NAME_SIZE 32 + +#define SOF_NAU8825_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) +#define SOF_NAU8825_SSP_CODEC_MASK (GENMASK(2, 0)) +#define SOF_SPEAKER_AMP_PRESENT BIT(3) +#define SOF_NAU8825_SSP_AMP_SHIFT 4 +#define SOF_NAU8825_SSP_AMP_MASK (GENMASK(6, 4)) +#define SOF_NAU8825_SSP_AMP(quirk) \ + (((quirk) << SOF_NAU8825_SSP_AMP_SHIFT) & SOF_NAU8825_SSP_AMP_MASK) +#define SOF_NAU8825_NUM_HDMIDEV_SHIFT 7 +#define SOF_NAU8825_NUM_HDMIDEV_MASK (GENMASK(9, 7)) +#define SOF_NAU8825_NUM_HDMIDEV(quirk) \ + (((quirk) << SOF_NAU8825_NUM_HDMIDEV_SHIFT) & SOF_NAU8825_NUM_HDMIDEV_MASK) + +/* BT audio offload: reserve 3 bits for future */ +#define SOF_BT_OFFLOAD_SSP_SHIFT 10 +#define SOF_BT_OFFLOAD_SSP_MASK (GENMASK(12, 10)) +#define SOF_BT_OFFLOAD_SSP(quirk) \ + (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) +#define SOF_SSP_BT_OFFLOAD_PRESENT BIT(13) +#define SOF_RT1019P_SPEAKER_AMP_PRESENT BIT(14) +#define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(15) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(16) + +static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0); + +struct sof_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct sof_card_private { + struct clk *mclk; + struct snd_soc_jack sof_headset; + struct list_head hdmi_pcm_list; +}; + +static int sof_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = asoc_rtd_to_codec(rtd, 0); + struct sof_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + /* dai_link id is 1:1 mapped to the PCM device */ + pcm->device = rtd->dai_link->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int sof_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + + struct snd_soc_jack *jack; + int ret; + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3, + &ctx->sof_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->sof_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static void sof_nau8825_codec_exit(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; + + snd_soc_component_set_jack(component, NULL, NULL); +} + +static int sof_nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + int clk_freq, ret; + + clk_freq = sof_dai_get_bclk(rtd); /* BCLK freq */ + + if (clk_freq <= 0) { + dev_err(rtd->dev, "get bclk freq failed: %d\n", clk_freq); + return -EINVAL; + } + + /* Configure clock for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_BLK, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set BCLK clock %d\n", ret); + return ret; + } + + /* Configure pll for codec */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, clk_freq, + params_rate(params) * 256); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set BCLK: %d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops sof_nau8825_ops = { + .hw_params = sof_nau8825_hw_params, +}; + +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:00:1f.3" + } +}; + +static int sof_card_late_probe(struct snd_soc_card *card) +{ + struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_dapm_context *dapm = &card->dapm; + 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"); + snd_soc_dapm_disable_pin(dapm, "Right Spk"); + err = snd_soc_dapm_sync(dapm); + if (err < 0) + return err; + } + + return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); +} + +static const struct snd_kcontrol_new sof_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_kcontrol_new speaker_controls[] = { + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget sof_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_widget speaker_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_widget dmic_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route sof_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* other jacks */ + { "MIC", NULL, "Headset Mic" }, +}; + +static const struct snd_soc_dapm_route speaker_map[] = { + /* speaker */ + { "Spk", NULL, "Speaker" }, +}; + +static const struct snd_soc_dapm_route dmic_map[] = { + /* digital mics */ + {"DMic", NULL, "SoC DMIC"}, +}; + +static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, speaker_widgets, + ARRAY_SIZE(speaker_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, speaker_controls, + ARRAY_SIZE(speaker_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map, + ARRAY_SIZE(speaker_map)); + + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} + +static int dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets, + ARRAY_SIZE(dmic_widgets)); + if (ret) { + dev_err(card->dev, "DMic widget addition failed: %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map, + ARRAY_SIZE(dmic_map)); + + if (ret) + dev_err(card->dev, "DMic map addition failed: %d\n", ret); + + return ret; +} + +/* sof audio machine driver for nau8825 codec */ +static struct snd_soc_card sof_audio_card_nau8825 = { + .name = "nau8825", /* the sof- prefix is added by the core */ + .owner = THIS_MODULE, + .controls = sof_controls, + .num_controls = ARRAY_SIZE(sof_controls), + .dapm_widgets = sof_widgets, + .num_dapm_widgets = ARRAY_SIZE(sof_widgets), + .dapm_routes = sof_map, + .num_dapm_routes = ARRAY_SIZE(sof_map), + .fully_routed = true, + .late_probe = sof_card_late_probe, +}; + +static struct snd_soc_dai_link_component nau8825_component[] = { + { + .name = "i2c-10508825:00", + .dai_name = "nau8825-hifi", + } +}; + +static struct snd_soc_dai_link_component dmic_component[] = { + { + .name = "dmic-codec", + .dai_name = "dmic-hifi", + } +}; + +static struct snd_soc_dai_link_component rt1019p_component[] = { + { + .name = "RTL1019:00", + .dai_name = "HiFi", + } +}; + +static struct snd_soc_dai_link_component dummy_component[] = { + { + .name = "snd-soc-dummy", + .dai_name = "snd-soc-dummy-dai", + } +}; + +static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, + int ssp_codec, + int ssp_amp, + int dmic_be_num, + int hdmi_num) +{ + struct snd_soc_dai_link_component *idisp_components; + struct snd_soc_dai_link_component *cpus; + struct snd_soc_dai_link *links; + int i, id = 0; + + links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * + sof_audio_card_nau8825.num_links, GFP_KERNEL); + cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component) * + sof_audio_card_nau8825.num_links, GFP_KERNEL); + if (!links || !cpus) + goto devm_err; + + /* codec SSP */ + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_codec); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].codecs = nau8825_component; + links[id].num_codecs = ARRAY_SIZE(nau8825_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_nau8825_codec_init; + links[id].exit = sof_nau8825_codec_exit; + links[id].ops = &sof_nau8825_ops; + links[id].dpcm_playback = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_codec); + if (!links[id].cpus->dai_name) + goto devm_err; + + id++; + + /* dmic */ + if (dmic_be_num > 0) { + /* at least we have dmic01 */ + links[id].name = "dmic01"; + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = "DMIC01 Pin"; + links[id].init = dmic_init; + if (dmic_be_num > 1) { + /* set up 2 BE links at most */ + links[id + 1].name = "dmic16k"; + links[id + 1].cpus = &cpus[id + 1]; + links[id + 1].cpus->dai_name = "DMIC16k Pin"; + dmic_be_num = 2; + } + } + + for (i = 0; i < dmic_be_num; i++) { + links[id].id = id; + links[id].num_cpus = 1; + links[id].codecs = dmic_component; + links[id].num_codecs = ARRAY_SIZE(dmic_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].ignore_suspend = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + id++; + } + + /* HDMI */ + if (hdmi_num > 0) { + idisp_components = devm_kzalloc(dev, + sizeof(struct snd_soc_dai_link_component) * + hdmi_num, GFP_KERNEL); + if (!idisp_components) + goto devm_err; + } + for (i = 1; i <= hdmi_num; i++) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d", i); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d Pin", i); + if (!links[id].cpus->dai_name) + goto devm_err; + + idisp_components[i - 1].name = "ehdaudio0D2"; + idisp_components[i - 1].dai_name = devm_kasprintf(dev, + GFP_KERNEL, + "intel-hdmi-hifi%d", + i); + if (!idisp_components[i - 1].dai_name) + goto devm_err; + + links[id].codecs = &idisp_components[i - 1]; + links[id].num_codecs = 1; + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].init = sof_hdmi_init; + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + id++; + } + + /* speaker amp */ + if (sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) { + links[id].name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d-Codec", ssp_amp); + if (!links[id].name) + goto devm_err; + + links[id].id = id; + if (sof_nau8825_quirk & SOF_RT1019P_SPEAKER_AMP_PRESENT) { + links[id].codecs = rt1019p_component; + links[id].num_codecs = ARRAY_SIZE(rt1019p_component); + links[id].init = speaker_codec_init; + } else if (sof_nau8825_quirk & + SOF_MAX98373_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98373_components; + links[id].num_codecs = ARRAY_SIZE(max_98373_components); + links[id].init = max_98373_spk_codec_init; + links[id].ops = &max_98373_ops; + /* feedback stream */ + links[id].dpcm_capture = 1; + } else if (sof_nau8825_quirk & + SOF_MAX98360A_SPEAKER_AMP_PRESENT) { + max_98360a_dai_link(&links[id]); + } else { + goto devm_err; + } + + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].dpcm_playback = 1; + links[id].no_pcm = 1; + links[id].cpus = &cpus[id]; + links[id].num_cpus = 1; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_amp); + if (!links[id].cpus->dai_name) + goto devm_err; + id++; + } + + /* BT audio offload */ + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { + int port = (sof_nau8825_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> + SOF_BT_OFFLOAD_SSP_SHIFT; + + links[id].id = id; + links[id].cpus = &cpus[id]; + links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", port); + if (!links[id].cpus->dai_name) + goto devm_err; + links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); + if (!links[id].name) + goto devm_err; + links[id].codecs = dummy_component; + links[id].num_codecs = ARRAY_SIZE(dummy_component); + links[id].platforms = platform_component; + links[id].num_platforms = ARRAY_SIZE(platform_component); + links[id].dpcm_playback = 1; + links[id].dpcm_capture = 1; + links[id].no_pcm = 1; + links[id].num_cpus = 1; + } + + return links; +devm_err: + return NULL; +} + +static int sof_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *dai_links; + struct snd_soc_acpi_mach *mach; + struct sof_card_private *ctx; + int dmic_be_num, hdmi_num; + int ret, ssp_amp, ssp_codec; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + + if (pdev->id_entry && pdev->id_entry->driver_data) + sof_nau8825_quirk = (unsigned long)pdev->id_entry->driver_data; + + mach = pdev->dev.platform_data; + + /* A speaker amp might not be present when the quirk claims one is. + * Detect this via whether the machine driver match includes quirk_data. + */ + if ((sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) + sof_nau8825_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + + dev_dbg(&pdev->dev, "sof_nau8825_quirk = %lx\n", sof_nau8825_quirk); + + /* default number of DMIC DAI's */ + dmic_be_num = 2; + hdmi_num = (sof_nau8825_quirk & SOF_NAU8825_NUM_HDMIDEV_MASK) >> + SOF_NAU8825_NUM_HDMIDEV_SHIFT; + /* default number of HDMI DAI's */ + if (!hdmi_num) + hdmi_num = 3; + + ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >> + SOF_NAU8825_SSP_AMP_SHIFT; + + ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK; + + /* compute number of dai links */ + sof_audio_card_nau8825.num_links = 1 + dmic_be_num + hdmi_num; + + if (sof_nau8825_quirk & SOF_SPEAKER_AMP_PRESENT) + sof_audio_card_nau8825.num_links++; + + if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) + max_98373_set_codec_conf(&sof_audio_card_nau8825); + + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) + sof_audio_card_nau8825.num_links++; + + dai_links = sof_card_dai_links_create(&pdev->dev, ssp_codec, ssp_amp, + dmic_be_num, hdmi_num); + if (!dai_links) + return -ENOMEM; + + sof_audio_card_nau8825.dai_link = dai_links; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + sof_audio_card_nau8825.dev = &pdev->dev; + + /* set platform name for each dailink */ + ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_nau8825, + mach->mach_params.platform); + if (ret) + return ret; + + snd_soc_card_set_drvdata(&sof_audio_card_nau8825, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &sof_audio_card_nau8825); +} + +static const struct platform_device_id board_ids[] = { + { + .name = "sof_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, + { + .name = "adl_rt1019p_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1019P_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(2) | + SOF_NAU8825_NUM_HDMIDEV(4)), + }, + { + .name = "adl_max98373_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98373_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { + /* The limitation of length of char array, shorten the name */ + .name = "adl_mx98360a_nau8825", + .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_NAU8825_SSP_AMP(1) | + SOF_NAU8825_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, + { } +}; +MODULE_DEVICE_TABLE(platform, board_ids); + +static struct platform_driver sof_audio = { + .probe = sof_audio_probe, + .driver = { + .name = "sof_nau8825", + .pm = &snd_soc_pm_ops, + }, + .id_table = board_ids, +}; +module_platform_driver(sof_audio) + +/* Module information */ +MODULE_DESCRIPTION("SOF Audio Machine driver for NAU8825"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index f5b21a95d222..98dc41b682cf 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -285,6 +285,11 @@ static const struct snd_soc_acpi_codecs adl_max98360a_amp = { .codecs = {"MX98360A"} }; +static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { + .num_codecs = 1, + .codecs = {"RTL1019"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -310,6 +315,36 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", }, + { + .id = "10508825", + .drv_name = "adl_rt1019p_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_rt1019p_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "adl_max98373_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98373_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98373-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "adl_mx98360a_nau8825", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-mx98360a-nau8825.tplg", + }, + { + .id = "10508825", + .drv_name = "sof_nau8825", + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-nau8825.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.20.1

4 3

[PATCH] ALSA: firewire-motu: add support for MOTU Track 16
by Takashi Sakamoto 08 Nov '21

08 Nov '21

Mark of the Unicorn designed Track 16 2011 as one of models in third generation of its FireWire series. The model is already discontinued. It consists of below ICs: * Texas Instruments TSB41AB1 * Microchip (SMSC) USB3300 * Xilinx Spartan-3A FPGA, XC3S700A * Texas Instruments TMS320C6722 * Microchip (Atmel) AT91SAM SAM7S512 It supports sampling transfer frequency up to 192.0 kHz. The packet format differs depending on both of current sampling transfer frequency and the type of signal in optical interfaces. The model supports transmission of PCM frames as well as MIDI messages. The model supports command mechanism to configure internal DSP. Hardware meter information is available in the first 2 chunks of each data block of tx packet. This commit adds support for it. $ cd linux-firewire-tools/src $ python crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04107d95 bus_info_length 4, crc_length 16, crc 32149 404 31333934 bus_name "1394" 408 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 40c 0001f200 company_id 0001f2 | 410 000a83c4 device_id 00000a83c4 | EUI-64 0001f200000a83c4 root directory ----------------------------------------------------------------- 414 0004ef04 directory_length 4, crc 61188 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 d1000002 --> unit directory at 428 424 8d000005 --> eui-64 leaf at 438 unit directory at 428 ----------------------------------------------------------------- 428 00035b04 directory_length 3, crc 23300 42c 120001f2 specifier id 430 13000039 version 434 17102800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 0002b25f leaf_length 2, crc 45663 43c 0001f200 company_id 0001f2 | 440 000a83c4 device_id 00000a83c4 | EUI-64 0001f200000a83c4 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 3 +++ sound/firewire/motu/motu-protocol-v3.c | 17 ++++++++++++++--- sound/firewire/motu/motu.c | 1 + sound/firewire/motu/motu.h | 1 + 4 files changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index fd109bea4c53..22b6c779682a 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -169,6 +169,7 @@ config SND_FIREWIRE_MOTU * 828 * 896 * 828mk2 + * 896hd * Traveler * Ultralite * 8pre @@ -176,7 +177,9 @@ config SND_FIREWIRE_MOTU * 828mk3 (Hybrid) * Ultralite mk3 (FireWire only) * Ultralite mk3 (Hybrid) + * Traveler mk3 * Audio Express + * Track 16 * 4pre To compile this driver as a module, choose M here: the module diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 1371f38505d0..8a0426920a76 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -189,7 +189,7 @@ int snd_motu_protocol_v3_switch_fetching_mode(struct snd_motu *motu, sizeof(reg)); } -static int detect_packet_formats_dual_opt_iface(struct snd_motu *motu, u32 data) +static int detect_packet_formats_with_opt_ifaces(struct snd_motu *motu, u32 data) { if (data & V3_ENABLE_OPT_IN_IFACE_A) { if (data & V3_NO_ADAT_OPT_IN_IFACE_A) { @@ -261,8 +261,9 @@ int snd_motu_protocol_v3_cache_packet_formats(struct snd_motu *motu) if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid || - motu->spec == &snd_motu_spec_traveler_mk3) - return detect_packet_formats_dual_opt_iface(motu, data); + motu->spec == &snd_motu_spec_traveler_mk3 || + motu->spec == &snd_motu_spec_track16) + return detect_packet_formats_with_opt_ifaces(motu, data); else return 0; } @@ -317,6 +318,16 @@ const struct snd_motu_spec snd_motu_spec_audio_express = { .rx_fixed_pcm_chunks = {10, 10, 0}, }; +const struct snd_motu_spec snd_motu_spec_track16 = { + .name = "Track16", + .protocol_version = SND_MOTU_PROTOCOL_V3, + .flags = SND_MOTU_SPEC_RX_MIDI_3RD_Q | + SND_MOTU_SPEC_TX_MIDI_3RD_Q | + SND_MOTU_SPEC_COMMAND_DSP, + .tx_fixed_pcm_chunks = {14, 14, 14}, + .rx_fixed_pcm_chunks = {6, 6, 6}, +}; + const struct snd_motu_spec snd_motu_spec_4pre = { .name = "4pre", .protocol_version = SND_MOTU_PROTOCOL_V3, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 080728dd278f..f8b7fe38751c 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -173,6 +173,7 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3_hybrid), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), + SND_MOTU_DEV_ENTRY(0x000039, &snd_motu_spec_track16), SND_MOTU_DEV_ENTRY(0x000045, &snd_motu_spec_4pre), { } }; diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 6ad1feab478b..4189f2192284 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -141,6 +141,7 @@ extern const struct snd_motu_spec snd_motu_spec_828mk3_hybrid; extern const struct snd_motu_spec snd_motu_spec_traveler_mk3; extern const struct snd_motu_spec snd_motu_spec_ultralite_mk3; extern const struct snd_motu_spec snd_motu_spec_audio_express; +extern const struct snd_motu_spec snd_motu_spec_track16; extern const struct snd_motu_spec snd_motu_spec_4pre; int amdtp_motu_init(struct amdtp_stream *s, struct fw_unit *unit, -- 2.32.0

2 1

[PATCH v0 23/42] ASoC: soc-jack: Check notifier registration return value
by Borislav Petkov 08 Nov '21

08 Nov '21

From: Borislav Petkov <bp(a)suse.de> Avoid homegrown notifier registration checks. No functional changes. Signed-off-by: Borislav Petkov <bp(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/soc/soc-jack.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index d798765d168c..0875f5022066 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -181,7 +181,8 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins); void snd_soc_jack_notifier_register(struct snd_soc_jack *jack, struct notifier_block *nb) { - blocking_notifier_chain_register(&jack->notifier, nb); + if (blocking_notifier_chain_register(&jack->notifier, nb)) + pr_warn("Jack status notifier already registered\n"); } EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_register); -- 2.29.2

1 0

Support TC Helicon GoXLR
by Anthony Bourguignon 08 Nov '21

08 Nov '21

Hi everyone, I’m the owner of a GoXLR : https://www.tc-helicon.com/product.html?modelCode=P0CQK . It is a usb sound mixer that is not supported officially with Linux. The main issue is that the GoXLR is a multi inputs / multi outputs device, but it is not seen as one. It works great under Windows thanks to a driver. With Linux, it is seen as a single usb sound device with a single multichannel output. Right now, I managed with other people to separate those channels with pulseaudio or jack or even pipewire. For exemple, Front Left and Front Right are the stereo output named "System". Rear left and rear right are another stereo output name Game. There are other outputs, and it is the same mechanism for the inputs. But I’ve seen that a few usb devices do have quirks for them to behave as expected, without the need of configuration with pulse or pipewire. But from what I understand, it’s only to disable subdevices or reorder outputs. Am I wrong ? Is it possible to add a quirk for a device like the GoXLR and change the multichannel output into multiple subdevices with alsa ? Thanks a lot if you can help me and the others owners. Anthony

1 0

ucm2: Add support for RT5650 on MediaTek SoCs
by GitHub pull_request - opened 08 Nov '21

08 Nov '21

alsa-project/alsa-ucm-conf pull request #117 was opened from kholk: At least one MT8173 machine uses a RT5650 codec and that's Google Elm: add a UCM2 configuration for this codec. This was tested on Acer Chromebook R13 (MT8173 Elm). Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/117 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/117.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: PCM: Fix NULL dereference at mmap checks
by Takashi Iwai 07 Nov '21

07 Nov '21

The recent refactoring of mmap handling caused Oops on some devices that don't use the standard memory allocations. This patch addresses it by allowing snd_dma_buffer_mmap() helper to receive the NULL pointer dmab argument (and return an error appropriately). Fixes: a202bd1ad86d ("ALSA: core: Move mmap handler into memalloc ops") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 0145dabda56d..de7e9d2df39d 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -183,8 +183,11 @@ EXPORT_SYMBOL_GPL(snd_devm_alloc_dir_pages); int snd_dma_buffer_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { - const struct snd_malloc_ops *ops = snd_dma_get_ops(dmab); + const struct snd_malloc_ops *ops; + if (!dmab) + return -ENOENT; + ops = snd_dma_get_ops(dmab); if (ops && ops->mmap) return ops->mmap(dmab, area); else -- 2.31.1

1 0

[Bug][5.16-rc0] Between commits dcd68326d29b and 7ddb58cb0eca, was broken play sound via the DisplayPort and HDMI.
by Mikhail Gavrilov 07 Nov '21

07 Nov '21

Hi! Between commits dcd68326d29b and 7ddb58cb0eca, was broken play sound via the DisplayPort and HDMI. When selected DisplayPort or HDMI in a gnome control center instead of sound in speakers in the kernel log pops errors: [ 69.228324] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00000 flags=0x0000] [ 69.228672] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00180 flags=0x0000] [ 69.229321] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00280 flags=0x0000] [ 69.229988] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00380 flags=0x0000] [ 69.230653] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00480 flags=0x0000] [ 69.231320] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00580 flags=0x0000] [ 69.231986] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00680 flags=0x0000] [ 69.232651] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00780 flags=0x0000] [ 69.233319] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00880 flags=0x0000] [ 69.233986] snd_hda_intel 0000:0b:00.1: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x001b address=0x1b7c00980 flags=0x0000] [ 77.693031] amd_iommu_report_page_fault: 5527 callbacks suppressed I tried to bisect the bad commit but didn't find the exact commit. $ git bisect log git bisect start # good: [dcd68326d29b62f3039e4f4d23d3e38f24d37360] Merge tag 'devicetree-for-5.16' of git://git.kernel.org/pub/scm/linux/kernel/git/robh/linux git bisect good dcd68326d29b62f3039e4f4d23d3e38f24d37360 # bad: [7ddb58cb0ecae8e8b6181d736a87667cc9ab8389] Merge tag 'clk-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/clk/linux git bisect bad 7ddb58cb0ecae8e8b6181d736a87667cc9ab8389 # bad: [2219b0ceefe835b92a8a74a73fe964aa052742a2] Merge tag 'soc-5.16' of git://git.kernel.org/pub/scm/linux/kernel/git/soc/soc git bisect bad 2219b0ceefe835b92a8a74a73fe964aa052742a2 # good: [2003c44e28ac9759200a78dda20c5f695949e3f4] ASoC: cs42l42: Prevent NULL pointer deref in interrupt handler git bisect good 2003c44e28ac9759200a78dda20c5f695949e3f4 # bad: [ff0700f03609b9f0defacd4ce96d9519d721e0a2] Merge tag 'sound-5.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound git bisect bad ff0700f03609b9f0defacd4ce96d9519d721e0a2 # good: [956ac4f1f53c58f87f38fd79d4c87e9146245c09] Merge series "ASoC: Sanity checks and soc-topology updates" from Cezary Rojewski <cezary.rojewski(a)intel.com>: git bisect good 956ac4f1f53c58f87f38fd79d4c87e9146245c09 # skip: [634ec0b2906efd46f6f57977e172aa3470aca432] ALSA: firewire-motu: notify event for parameter change in register DSP model git bisect skip 634ec0b2906efd46f6f57977e172aa3470aca432 # bad: [407359d44ed33974137b9158da356d53f999dcf2] ALSA: firewire-motu: export meter information to userspace as float value git bisect bad 407359d44ed33974137b9158da356d53f999dcf2 # good: [c18c4966033e6473a472fb65fbd5a6441603fbf7] ALSA: pcm: Unify snd_pcm_delay() and snd_pcm_hwsync() git bisect good c18c4966033e6473a472fb65fbd5a6441603fbf7 # skip: [73325f60e2ed28f04032d43c2828b73776cfefd0] ALSA: memalloc: Support for non-coherent page allocation git bisect skip 73325f60e2ed28f04032d43c2828b73776cfefd0 # skip: [4c9eda8f37f9523f1d2ccbb442ce641e8c981c9f] ALSA: firewire-motu: queue event for parameter change in register DSP model git bisect skip 4c9eda8f37f9523f1d2ccbb442ce641e8c981c9f # skip: [58b62ab7025912ce1be36e3ba19d49620a0161b6] ALSA: firewire-motu: add ioctl command to read cached hardware meter git bisect skip 58b62ab7025912ce1be36e3ba19d49620a0161b6 # bad: [5aec579e08e4f2be7103ae264ac8f34883eb9273] ALSA: uapi: Fix a C++ style comment in asound.h git bisect bad 5aec579e08e4f2be7103ae264ac8f34883eb9273 # skip: [7d843c494a9b69d07bc0588124599e3f665a1496] ALSA: firewire-motu: parse messages for input parameters in register DSP model git bisect skip 7d843c494a9b69d07bc0588124599e3f665a1496 # skip: [ca15a09ccc5bd2731c5fcb667e6ea3bbbf8f5772] ALSA: firewire-motu: add ioctl command to read cached parameters in register DSP model git bisect skip ca15a09ccc5bd2731c5fcb667e6ea3bbbf8f5772 # skip: [dc36a9755a572781903d79f8437d109b72662da5] ALSA: firewire-motu: parse messages for mixer source parameters in register-DSP model git bisect skip dc36a9755a572781903d79f8437d109b72662da5 # skip: [41cc23389f5fc64bdac78b73935a44bd5abc990d] ALSA: firewire-motu: parse messages for line input parameters in register DSP model git bisect skip 41cc23389f5fc64bdac78b73935a44bd5abc990d # skip: [a25684a956468ee8bbbee44649e41e5d447e5adc] ALSA: memalloc: Support for non-contiguous page allocation git bisect skip a25684a956468ee8bbbee44649e41e5d447e5adc # skip: [6ca81d2b6305a884da441fd0281ff01afd5f8c7e] ALSA: firewire-motu: parse messages for output parameters in register DSP model git bisect skip 6ca81d2b6305a884da441fd0281ff01afd5f8c7e # skip: [90b28f3bb85c39b11daf29d473ef21a935c70ec5] ALSA: firewire-motu: add message parser for meter information in command DSP model git bisect skip 90b28f3bb85c39b11daf29d473ef21a935c70ec5 # skip: [3c05f1477e62ea5a0a8797ba6a545b1dc751fb31] ALSA: ISA: not for M68K git bisect skip 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 # skip: [2d9ea39917a4e4293bc2caea902c7059a330b611] ALSA: memalloc: Convert x86 SG-buffer handling with non-contiguous type git bisect skip 2d9ea39917a4e4293bc2caea902c7059a330b611 # skip: [ce69bed5557b05dd1918556d4e90c293382155ae] ALSA: firewire-motu: parse messages for mixer output parameters in register DSP model git bisect skip ce69bed5557b05dd1918556d4e90c293382155ae # bad: [b15706471abe916a16a38bee4434612998d869d2] ALSA: firewire: Fix C++ style comments in uapi header git bisect bad b15706471abe916a16a38bee4434612998d869d2 # skip: [bea36afa102e37d5e4d9ea519f14d1c92d512e45] ALSA: firewire-motu: add message parser to gather meter information in register DSP model git bisect skip bea36afa102e37d5e4d9ea519f14d1c92d512e45 # only skipped commits left to test # possible first bad commit: [b15706471abe916a16a38bee4434612998d869d2] ALSA: firewire: Fix C++ style comments in uapi header # possible first bad commit: [2d9ea39917a4e4293bc2caea902c7059a330b611] ALSA: memalloc: Convert x86 SG-buffer handling with non-contiguous type # possible first bad commit: [73325f60e2ed28f04032d43c2828b73776cfefd0] ALSA: memalloc: Support for non-coherent page allocation # possible first bad commit: [a25684a956468ee8bbbee44649e41e5d447e5adc] ALSA: memalloc: Support for non-contiguous page allocation # possible first bad commit: [3c05f1477e62ea5a0a8797ba6a545b1dc751fb31] ALSA: ISA: not for M68K # possible first bad commit: [634ec0b2906efd46f6f57977e172aa3470aca432] ALSA: firewire-motu: notify event for parameter change in register DSP model # possible first bad commit: [4c9eda8f37f9523f1d2ccbb442ce641e8c981c9f] ALSA: firewire-motu: queue event for parameter change in register DSP model # possible first bad commit: [ca15a09ccc5bd2731c5fcb667e6ea3bbbf8f5772] ALSA: firewire-motu: add ioctl command to read cached parameters in register DSP model # possible first bad commit: [7d843c494a9b69d07bc0588124599e3f665a1496] ALSA: firewire-motu: parse messages for input parameters in register DSP model # possible first bad commit: [41cc23389f5fc64bdac78b73935a44bd5abc990d] ALSA: firewire-motu: parse messages for line input parameters in register DSP model # possible first bad commit: [6ca81d2b6305a884da441fd0281ff01afd5f8c7e] ALSA: firewire-motu: parse messages for output parameters in register DSP model # possible first bad commit: [ce69bed5557b05dd1918556d4e90c293382155ae] ALSA: firewire-motu: parse messages for mixer output parameters in register DSP model # possible first bad commit: [dc36a9755a572781903d79f8437d109b72662da5] ALSA: firewire-motu: parse messages for mixer source parameters in register-DSP model # possible first bad commit: [58b62ab7025912ce1be36e3ba19d49620a0161b6] ALSA: firewire-motu: add ioctl command to read cached hardware meter # possible first bad commit: [90b28f3bb85c39b11daf29d473ef21a935c70ec5] ALSA: firewire-motu: add message parser for meter information in command DSP model # possible first bad commit: [bea36afa102e37d5e4d9ea519f14d1c92d512e45] ALSA: firewire-motu: add message parser to gather meter information in register DSP model I hope it helps. The full kernel log is attached to this message. -- Best Regards, Mike Gavrilov.

2 1

[PATCH] sound/soc: adds TAS5754M digital input amplifier component driver
by joerg＠hifiberry.com 07 Nov '21

07 Nov '21

From: Joerg Schambacher <joerg(a)hifiberry.com> Adds a minimum component driver to run the amplifier in I2S master mode only from standard audio clocks. Therefore, it only allows 44.1, 88.2, 176.4, 48, 96 and 192ksps with 16, 20, 24 and 32 bits sample size. Digital volume control and the -6dB switch are supported. Signed-off-by: Joerg Schambacher <joerg(a)hifiberry.com> --- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5754m.c | 547 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas5754m.h | 259 +++++++++++++++++ 4 files changed, 816 insertions(+) create mode 100644 sound/soc/codecs/tas5754m.c create mode 100644 sound/soc/codecs/tas5754m.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233a269d..cf0584948fcf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -210,6 +210,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 + imply SND_SOC_TAS5754M imply SND_SOC_TAS6424 imply SND_SOC_TDA7419 imply SND_SOC_TFA9879 @@ -1419,6 +1420,13 @@ config SND_SOC_TAS5720 Enable support for Texas Instruments TAS5720L/M high-efficiency mono Class-D audio power amplifiers. +config SND_SOC_TAS5754M + tristate "Texas Instruments TAS5754M Digital Input Audio amplifier" + depends on I2C + help + Enable support for Texas Instruments TAS5754M digital input + Class-D audio power amplifiers. + config SND_SOC_TAS6424 tristate "Texas Instruments TAS6424 Quad-Channel Audio amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8dcea2c4604a..39984900258a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -227,6 +227,7 @@ snd-soc-sti-sas-objs := sti-sas.o snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o snd-soc-tas5720-objs := tas5720.o +snd-soc-tas5754m-objs := tas5754m.o snd-soc-tas6424-objs := tas6424.o snd-soc-tda7419-objs := tda7419.o snd-soc-tas2770-objs := tas2770.o @@ -555,6 +556,7 @@ obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o +obj-$(CONFIG_SND_SOC_TAS5754M) += snd-soc-tas5754m.o obj-$(CONFIG_SND_SOC_TAS6424) += snd-soc-tas6424.o obj-$(CONFIG_SND_SOC_TDA7419) += snd-soc-tda7419.o obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o diff --git a/sound/soc/codecs/tas5754m.c b/sound/soc/codecs/tas5754m.c new file mode 100644 index 000000000000..19b16a2c4817 --- /dev/null +++ b/sound/soc/codecs/tas5754m.c @@ -0,0 +1,547 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Driver for the TAS5754M Audio Amplifier in Master mode (only) + * supports only standard audio frequencies 44.1 to 192 ksps + * + * Author: Joerg Schambacher <joerg(a)hifiberry.com> + * with fragments from Andy Liu <andy-liu(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/clk.h> +#include <linux/slab.h> +#include <linux/of.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/delay.h> + +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> + +#include "tas5754m.h" + +#define TAS5754M_RATES (SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_176400) +#define TAS5754M_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + + +static const struct reg_sequence tas5754m_init_sequence[] = { + { TAS5754M_RESET, 0x00 }, + { TAS5754M_MUTE, 0x11 }, + { TAS5754M_POWER, 0x00 }, + { TAS5754M_PLL_EN, 0x00 }, + { TAS5754M_RESET, 0x00 }, + { TAS5754M_GPIO_OUTPUT_3, 0x02 }, + { TAS5754M_GPIO_OUTPUT_4, 0x02 }, + { TAS5754M_GPIO_OUTPUT_6, 0x02 }, + { TAS5754M_GPIO_EN, 0x2c }, + { TAS5754M_GPIO_CONTROL_1, 0x04 }, + { TAS5754M_BCLK_LRCLK_CFG, 0x11 }, + { TAS5754M_MASTER_MODE, 0x7c }, + { TAS5754M_ERROR_DETECT, 0x77 }, + { TAS5754M_PLL_EN, 0x01 }, + { TAS5754M_PLL_REF, 0x00 }, + { TAS5754M_PLL_COEFF_0, 0x03 }, + { TAS5754M_PLL_COEFF_1, 0x0c }, + { TAS5754M_PLL_COEFF_2, 0x00 }, + { TAS5754M_PLL_COEFF_3, 0x00 }, + { TAS5754M_PLL_COEFF_4, 0x00 }, + { TAS5754M_DAC_REF, 0x30 }, + { TAS5754M_DSP_CLKDIV, 0x01 }, + { TAS5754M_DAC_CLKDIV, 0x0f }, + { TAS5754M_NCP_CLKDIV, 0x03 }, + { TAS5754M_OSR_CLKDIV, 0x00 }, + { TAS5754M_FS_SPEED_MODE, 0x00 }, + { TAS5754M_MASTER_CLKDIV_1, 0x0f }, + { TAS5754M_MASTER_CLKDIV_2, 0x1f }, + { TAS5754M_I2S_1, 0x00 }, + { TAS5754M_I2S_2, 0x01 }, + { TAS5754M_PLL_EN, 0x01 }, + { TAS5754M_MASTER_MODE, 0x7f }, + { TAS5754M_MUTE, 0x11 }, +}; + +static const struct reg_default tas5754m_reg_defaults[] = { + { TAS5754M_RESET, 0x00 }, + { TAS5754M_POWER, 0x00 }, + { TAS5754M_MUTE, 0x00 }, + { TAS5754M_DSP, 0x00 }, + { TAS5754M_PLL_REF, 0x00 }, + { TAS5754M_DAC_REF, 0x00 }, + { TAS5754M_DAC_ROUTING, 0x11 }, + { TAS5754M_DSP_PROGRAM, 0x01 }, + { TAS5754M_CLKDET, 0x00 }, + { TAS5754M_AUTO_MUTE, 0x00 }, + { TAS5754M_ERROR_DETECT, 0x00 }, + { TAS5754M_DIGITAL_VOLUME_1, 0x00 }, + { TAS5754M_DIGITAL_VOLUME_2, 0x30 }, + { TAS5754M_DIGITAL_VOLUME_3, 0x30 }, + { TAS5754M_DIGITAL_MUTE_1, 0x22 }, + { TAS5754M_DIGITAL_MUTE_2, 0x00 }, + { TAS5754M_DIGITAL_MUTE_3, 0x07 }, + { TAS5754M_OUTPUT_AMPLITUDE, 0x00 }, + { TAS5754M_ANALOG_GAIN_CTRL, 0x00 }, + { TAS5754M_UNDERVOLTAGE_PROT, 0x00 }, + { TAS5754M_ANALOG_MUTE_CTRL, 0x00 }, + { TAS5754M_ANALOG_GAIN_BOOST, 0x00 }, + { TAS5754M_VCOM_CTRL_1, 0x00 }, + { TAS5754M_VCOM_CTRL_2, 0x01 }, + { TAS5754M_BCLK_LRCLK_CFG, 0x00 }, + { TAS5754M_MASTER_MODE, 0x7c }, + { TAS5754M_GPIO_DACIN, 0x00 }, + { TAS5754M_GPIO_PLLIN, 0x00 }, + { TAS5754M_SYNCHRONIZE, 0x10 }, + { TAS5754M_PLL_COEFF_0, 0x00 }, + { TAS5754M_PLL_COEFF_1, 0x00 }, + { TAS5754M_PLL_COEFF_2, 0x00 }, + { TAS5754M_PLL_COEFF_3, 0x00 }, + { TAS5754M_PLL_COEFF_4, 0x00 }, + { TAS5754M_DSP_CLKDIV, 0x00 }, + { TAS5754M_DAC_CLKDIV, 0x00 }, + { TAS5754M_NCP_CLKDIV, 0x00 }, + { TAS5754M_OSR_CLKDIV, 0x00 }, + { TAS5754M_MASTER_CLKDIV_1, 0x00 }, + { TAS5754M_MASTER_CLKDIV_2, 0x00 }, + { TAS5754M_FS_SPEED_MODE, 0x00 }, + { TAS5754M_IDAC_1, 0x01 }, + { TAS5754M_IDAC_2, 0x00 }, +}; + +static bool tas5754m_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5754M_RESET: + case TAS5754M_POWER: + case TAS5754M_MUTE: + case TAS5754M_PLL_EN: + case TAS5754M_SPI_MISO_FUNCTION: + case TAS5754M_DSP: + case TAS5754M_GPIO_EN: + case TAS5754M_BCLK_LRCLK_CFG: + case TAS5754M_DSP_GPIO_INPUT: + case TAS5754M_MASTER_MODE: + case TAS5754M_PLL_REF: + case TAS5754M_DAC_REF: + case TAS5754M_GPIO_DACIN: + case TAS5754M_GPIO_PLLIN: + case TAS5754M_SYNCHRONIZE: + case TAS5754M_PLL_COEFF_0: + case TAS5754M_PLL_COEFF_1: + case TAS5754M_PLL_COEFF_2: + case TAS5754M_PLL_COEFF_3: + case TAS5754M_PLL_COEFF_4: + case TAS5754M_DSP_CLKDIV: + case TAS5754M_DAC_CLKDIV: + case TAS5754M_NCP_CLKDIV: + case TAS5754M_OSR_CLKDIV: + case TAS5754M_MASTER_CLKDIV_1: + case TAS5754M_MASTER_CLKDIV_2: + case TAS5754M_FS_SPEED_MODE: + case TAS5754M_IDAC_1: + case TAS5754M_IDAC_2: + case TAS5754M_ERROR_DETECT: + case TAS5754M_I2S_1: + case TAS5754M_I2S_2: + case TAS5754M_DAC_ROUTING: + case TAS5754M_DSP_PROGRAM: + case TAS5754M_CLKDET: + case TAS5754M_AUTO_MUTE: + case TAS5754M_DIGITAL_VOLUME_1: + case TAS5754M_DIGITAL_VOLUME_2: + case TAS5754M_DIGITAL_VOLUME_3: + case TAS5754M_DIGITAL_MUTE_1: + case TAS5754M_DIGITAL_MUTE_2: + case TAS5754M_DIGITAL_MUTE_3: + case TAS5754M_GPIO_OUTPUT_1: + case TAS5754M_GPIO_OUTPUT_2: + case TAS5754M_GPIO_OUTPUT_3: + case TAS5754M_GPIO_OUTPUT_4: + case TAS5754M_GPIO_OUTPUT_5: + case TAS5754M_GPIO_OUTPUT_6: + case TAS5754M_GPIO_CONTROL_1: + case TAS5754M_GPIO_CONTROL_2: + case TAS5754M_OVERFLOW: + case TAS5754M_RATE_DET_1: + case TAS5754M_RATE_DET_2: + case TAS5754M_RATE_DET_3: + case TAS5754M_RATE_DET_4: + case TAS5754M_CLOCK_STATUS: + case TAS5754M_ANALOG_MUTE_DET: + case TAS5754M_GPIN: + case TAS5754M_DIGITAL_MUTE_DET: + case TAS5754M_OUTPUT_AMPLITUDE: + case TAS5754M_ANALOG_GAIN_CTRL: + case TAS5754M_UNDERVOLTAGE_PROT: + case TAS5754M_ANALOG_MUTE_CTRL: + case TAS5754M_ANALOG_GAIN_BOOST: + case TAS5754M_VCOM_CTRL_1: + case TAS5754M_VCOM_CTRL_2: + case TAS5754M_CRAM_CTRL: + case TAS5754M_FLEX_A: + case TAS5754M_FLEX_B: + return true; + default: + return reg < 0x7f; + } +} + +static bool tas5754m_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5754M_PLL_EN: + case TAS5754M_OVERFLOW: + case TAS5754M_RATE_DET_1: + case TAS5754M_RATE_DET_2: + case TAS5754M_RATE_DET_3: + case TAS5754M_RATE_DET_4: + case TAS5754M_CLOCK_STATUS: + case TAS5754M_ANALOG_MUTE_DET: + case TAS5754M_GPIN: + case TAS5754M_DIGITAL_MUTE_DET: + case TAS5754M_CRAM_CTRL: + return true; + default: + return reg < 0x7f; + } +} + +struct tas5754m_priv { + struct regmap *regmap; + struct clk *sclk; +}; + +static const struct regmap_range_cfg tas5754m_range = { + .name = "Pages", + .range_min = TAS5754M_VIRT_BASE, + .range_max = TAS5754M_MAX_REGISTER, + .selector_reg = TAS5754M_PAGE, + .selector_mask = 0x7f, + .window_start = 0, + .window_len = 128, +}; + +const struct regmap_config tas5754m_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .ranges = &tas5754m_range, + .num_ranges = 1, + .max_register = TAS5754M_MAX_REGISTER, + + .reg_defaults = tas5754m_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5754m_reg_defaults), + .readable_reg = tas5754m_readable, + .volatile_reg = tas5754m_volatile, + + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(tas5754m_regmap); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); +static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0); + +static const struct snd_kcontrol_new tas5754m_controls[] = { +SOC_DOUBLE_R_TLV("Digital Playback Volume", TAS5754M_DIGITAL_VOLUME_2, + TAS5754M_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), +SOC_DOUBLE_TLV("Analog Playback Volume", TAS5754M_ANALOG_GAIN_CTRL, + TAS5754M_LAGN_SHIFT, TAS5754M_RAGN_SHIFT, 1, 1, analog_tlv), +}; + +static int tas5754m_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct tas5754m_priv *tas5754m = + snd_soc_component_get_drvdata(component); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + ret = regmap_update_bits(tas5754m->regmap, + TAS5754M_POWER, TAS5754M_RQST, 0); + if (ret != 0) { + dev_err(component->dev, + "Failed to remove standby: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_OFF: + ret = regmap_update_bits(tas5754m->regmap, + TAS5754M_POWER, TAS5754M_RQST, TAS5754M_RQST); + if (ret != 0) { + dev_err(component->dev, + "Failed to request standby: %d\n", ret); + return ret; + } + break; + } + + return 0; +} + +static int tas5754m_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 tas5754m_priv *tas5754m = + snd_soc_component_get_drvdata(component); + unsigned long bclk; + unsigned long mclk; + int sample_len; + int bclk_div; + int lrclk_div; + int alen; + int ret; + + switch (params_width(params)) { + case 16: + sample_len = 16; + alen = TAS5754M_ALEN_16; + break; + case 20: + sample_len = 32; + alen = TAS5754M_ALEN_20; + break; + case 24: + sample_len = 32; + alen = TAS5754M_ALEN_24; + break; + case 32: + sample_len = 32; + alen = TAS5754M_ALEN_32; + break; + default: + dev_err(component->dev, "Unsupported sample size: %d\n", + params_width(params)); + return -EINVAL; + } + ret = regmap_update_bits(tas5754m->regmap, TAS5754M_I2S_1, alen, alen); + if (ret != 0) { + dev_err(component->dev, + "Cannot set sample size: %d\n", ret); + return ret; + } + + switch (params_rate(params)) { + case 44100: + case 48000: + ret = regmap_write(tas5754m->regmap, + TAS5754M_FS_SPEED_MODE, TAS5754M_FSSP_48KHZ); + break; + case 88200: + case 96000: + ret = regmap_write(tas5754m->regmap, + TAS5754M_FS_SPEED_MODE, TAS5754M_FSSP_96KHZ); + break; + case 176400: + case 192000: + ret = regmap_write(tas5754m->regmap, + TAS5754M_FS_SPEED_MODE, TAS5754M_FSSP_192KHZ); + break; + default: + dev_err(component->dev, "Sample rate not supported: %d\n", + params_rate(params)); + return -EINVAL; + } + if (ret != 0) { + dev_err(component->dev, "Failed to config PLL\n"); + return ret; + } + + + mclk = clk_get_rate(tas5754m->sclk); + bclk = sample_len * 2 * params_rate(params); + bclk_div = mclk / bclk; + lrclk_div = sample_len * 2; + + // stop LR / SCLK clocks + ret = regmap_write(tas5754m->regmap, TAS5754M_MASTER_MODE, 0x7c); + + // set SCLK divider + ret |= regmap_write(tas5754m->regmap, TAS5754M_MASTER_CLKDIV_1, + bclk_div - 1); + + // set LRCLK divider + ret |= regmap_write(tas5754m->regmap, TAS5754M_MASTER_CLKDIV_2, + lrclk_div - 1); + + // restart LR / SCLK clocks + ret |= regmap_write(tas5754m->regmap, TAS5754M_MASTER_MODE, 0x7f); + if (ret != 0) { + dev_err(component->dev, "Failed to config PLL\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_component_driver tas5754m_soc_component = { + .set_bias_level = tas5754m_set_bias_level, + .idle_bias_on = true, + .controls = tas5754m_controls, + .num_controls = ARRAY_SIZE(tas5754m_controls), +}; + +static int tas5754m_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *component = dai->component; + + if (mute) { + snd_soc_component_write(component, TAS5754M_MUTE, 0x11); + } else { + usleep_range(1000, 2000); + snd_soc_component_write(component, TAS5754M_MUTE, 0x00); + } + return 0; +} + +static const struct snd_soc_dai_ops tas5754m_dai_ops = { + .mute_stream = tas5754m_mute, + .hw_params = tas5754m_hw_params, +}; + +static struct snd_soc_dai_driver tas5754m_dai = { + .name = "tas5754m-amplifier", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TAS5754M_RATES, + .formats = TAS5754M_FORMATS, + }, + .ops = &tas5754m_dai_ops, +}; + +static int tas5754m_probe(struct device *dev, struct regmap *regmap) +{ + struct tas5754m_priv *tas5754m; + int ret; + + tas5754m = devm_kzalloc(dev, sizeof(struct tas5754m_priv), GFP_KERNEL); + if (!tas5754m) + return -ENOMEM; + + dev_set_drvdata(dev, tas5754m); + tas5754m->regmap = regmap; + + ret = regmap_multi_reg_write(regmap, tas5754m_init_sequence, + ARRAY_SIZE(tas5754m_init_sequence)); + + if (ret != 0) { + dev_err(dev, "Failed to initialize TAS5754M: %d\n", ret); + goto err; + } + + tas5754m->sclk = devm_clk_get(dev, NULL); + if (PTR_ERR(tas5754m->sclk) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + goto err; + } + if (!IS_ERR(tas5754m->sclk)) { + ret = clk_prepare_enable(tas5754m->sclk); + if (ret != 0) { + dev_err(dev, "Failed to enable SCLK: %d\n", ret); + goto err; + } + } + + ret = snd_soc_register_component(dev, + &tas5754m_soc_component, &tas5754m_dai, 1); + if (ret != 0) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err; + } + + return 0; + +err: + return ret; + +} + +static int tas5754m_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) +{ + struct regmap *regmap; + struct regmap_config config = tas5754m_regmap; + + /* enable auto-increment mode */ + config.read_flag_mask = 0x80; + config.write_flag_mask = 0x80; + + regmap = devm_regmap_init_i2c(i2c, &config); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return tas5754m_probe(&i2c->dev, regmap); +} + +static int tas5754m_remove(struct device *dev) +{ + snd_soc_unregister_component(dev); + + return 0; +} + +static int tas5754m_i2c_remove(struct i2c_client *i2c) +{ + tas5754m_remove(&i2c->dev); + + return 0; +} + +static const struct i2c_device_id tas5754m_i2c_id[] = { + { "tas5754m", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5754m_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id tas5754m_of_match[] = { + { .compatible = "ti,tas5754m", }, + { .compatible = "ti,tas5756m", }, + { } +}; +MODULE_DEVICE_TABLE(of, tas5754m_of_match); +#endif + +static struct i2c_driver tas5754m_i2c_driver = { + .probe = tas5754m_i2c_probe, + .remove = tas5754m_i2c_remove, + .id_table = tas5754m_i2c_id, + .driver = { + .name = "tas5754m", + .of_match_table = tas5754m_of_match, + }, +}; + +module_i2c_driver(tas5754m_i2c_driver); + +MODULE_AUTHOR("Joerg Schambacher <joerg(a)hifiberry.com>"); +MODULE_DESCRIPTION("TAS5754M Audio Amplifier Driver - Master mode only"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas5754m.h b/sound/soc/codecs/tas5754m.h new file mode 100644 index 000000000000..492b8abede6c --- /dev/null +++ b/sound/soc/codecs/tas5754m.h @@ -0,0 +1,259 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Driver for the TAS5754M DAC+amplifier combo devices + * + * Author: (copied from pcm512x.h) + * Mark Brown <broonie(a)kernel.org> + * Copyright 2014 Linaro Ltd + */ + +#ifndef _SND_SOC_TAS5754M +#define _SND_SOC_TAS5754M + +#include <linux/pm.h> +#include <linux/regmap.h> + +#define TAS5754M_VIRT_BASE 0x000 +#define TAS5754M_PAGE_LEN 0x80 +#define TAS5754M_PAGE_BASE(n) (TAS5754M_VIRT_BASE + (TAS5754M_PAGE_LEN * n)) + +#define TAS5754M_PAGE 0 + +#define TAS5754M_RESET (TAS5754M_PAGE_BASE(0) + 1) +#define TAS5754M_POWER (TAS5754M_PAGE_BASE(0) + 2) +#define TAS5754M_MUTE (TAS5754M_PAGE_BASE(0) + 3) +#define TAS5754M_PLL_EN (TAS5754M_PAGE_BASE(0) + 4) +#define TAS5754M_SPI_MISO_FUNCTION (TAS5754M_PAGE_BASE(0) + 6) +#define TAS5754M_DSP (TAS5754M_PAGE_BASE(0) + 7) +#define TAS5754M_GPIO_EN (TAS5754M_PAGE_BASE(0) + 8) +#define TAS5754M_BCLK_LRCLK_CFG (TAS5754M_PAGE_BASE(0) + 9) +#define TAS5754M_DSP_GPIO_INPUT (TAS5754M_PAGE_BASE(0) + 10) +#define TAS5754M_MASTER_MODE (TAS5754M_PAGE_BASE(0) + 12) +#define TAS5754M_PLL_REF (TAS5754M_PAGE_BASE(0) + 13) +#define TAS5754M_DAC_REF (TAS5754M_PAGE_BASE(0) + 14) +#define TAS5754M_GPIO_DACIN (TAS5754M_PAGE_BASE(0) + 16) +#define TAS5754M_GPIO_PLLIN (TAS5754M_PAGE_BASE(0) + 18) +#define TAS5754M_SYNCHRONIZE (TAS5754M_PAGE_BASE(0) + 19) +#define TAS5754M_PLL_COEFF_0 (TAS5754M_PAGE_BASE(0) + 20) +#define TAS5754M_PLL_COEFF_1 (TAS5754M_PAGE_BASE(0) + 21) +#define TAS5754M_PLL_COEFF_2 (TAS5754M_PAGE_BASE(0) + 22) +#define TAS5754M_PLL_COEFF_3 (TAS5754M_PAGE_BASE(0) + 23) +#define TAS5754M_PLL_COEFF_4 (TAS5754M_PAGE_BASE(0) + 24) +#define TAS5754M_DSP_CLKDIV (TAS5754M_PAGE_BASE(0) + 27) +#define TAS5754M_DAC_CLKDIV (TAS5754M_PAGE_BASE(0) + 28) +#define TAS5754M_NCP_CLKDIV (TAS5754M_PAGE_BASE(0) + 29) +#define TAS5754M_OSR_CLKDIV (TAS5754M_PAGE_BASE(0) + 30) +#define TAS5754M_MASTER_CLKDIV_1 (TAS5754M_PAGE_BASE(0) + 32) +#define TAS5754M_MASTER_CLKDIV_2 (TAS5754M_PAGE_BASE(0) + 33) +#define TAS5754M_FS_SPEED_MODE (TAS5754M_PAGE_BASE(0) + 34) +#define TAS5754M_IDAC_1 (TAS5754M_PAGE_BASE(0) + 35) +#define TAS5754M_IDAC_2 (TAS5754M_PAGE_BASE(0) + 36) +#define TAS5754M_ERROR_DETECT (TAS5754M_PAGE_BASE(0) + 37) +#define TAS5754M_I2S_1 (TAS5754M_PAGE_BASE(0) + 40) +#define TAS5754M_I2S_2 (TAS5754M_PAGE_BASE(0) + 41) +#define TAS5754M_DAC_ROUTING (TAS5754M_PAGE_BASE(0) + 42) +#define TAS5754M_DSP_PROGRAM (TAS5754M_PAGE_BASE(0) + 43) +#define TAS5754M_CLKDET (TAS5754M_PAGE_BASE(0) + 44) +#define TAS5754M_AUTO_MUTE (TAS5754M_PAGE_BASE(0) + 59) +#define TAS5754M_DIGITAL_VOLUME_1 (TAS5754M_PAGE_BASE(0) + 60) +#define TAS5754M_DIGITAL_VOLUME_2 (TAS5754M_PAGE_BASE(0) + 61) +#define TAS5754M_DIGITAL_VOLUME_3 (TAS5754M_PAGE_BASE(0) + 62) +#define TAS5754M_DIGITAL_MUTE_1 (TAS5754M_PAGE_BASE(0) + 63) +#define TAS5754M_DIGITAL_MUTE_2 (TAS5754M_PAGE_BASE(0) + 64) +#define TAS5754M_DIGITAL_MUTE_3 (TAS5754M_PAGE_BASE(0) + 65) +#define TAS5754M_GPIO_OUTPUT_1 (TAS5754M_PAGE_BASE(0) + 80) +#define TAS5754M_GPIO_OUTPUT_2 (TAS5754M_PAGE_BASE(0) + 81) +#define TAS5754M_GPIO_OUTPUT_3 (TAS5754M_PAGE_BASE(0) + 82) +#define TAS5754M_GPIO_OUTPUT_4 (TAS5754M_PAGE_BASE(0) + 83) +#define TAS5754M_GPIO_OUTPUT_5 (TAS5754M_PAGE_BASE(0) + 84) +#define TAS5754M_GPIO_OUTPUT_6 (TAS5754M_PAGE_BASE(0) + 85) +#define TAS5754M_GPIO_CONTROL_1 (TAS5754M_PAGE_BASE(0) + 86) +#define TAS5754M_GPIO_CONTROL_2 (TAS5754M_PAGE_BASE(0) + 87) +#define TAS5754M_OVERFLOW (TAS5754M_PAGE_BASE(0) + 90) +#define TAS5754M_RATE_DET_1 (TAS5754M_PAGE_BASE(0) + 91) +#define TAS5754M_RATE_DET_2 (TAS5754M_PAGE_BASE(0) + 92) +#define TAS5754M_RATE_DET_3 (TAS5754M_PAGE_BASE(0) + 93) +#define TAS5754M_RATE_DET_4 (TAS5754M_PAGE_BASE(0) + 94) +#define TAS5754M_CLOCK_STATUS (TAS5754M_PAGE_BASE(0) + 95) +#define TAS5754M_ANALOG_MUTE_DET (TAS5754M_PAGE_BASE(0) + 108) +#define TAS5754M_GPIN (TAS5754M_PAGE_BASE(0) + 119) +#define TAS5754M_DIGITAL_MUTE_DET (TAS5754M_PAGE_BASE(0) + 120) + +#define TAS5754M_OUTPUT_AMPLITUDE (TAS5754M_PAGE_BASE(1) + 1) +#define TAS5754M_ANALOG_GAIN_CTRL (TAS5754M_PAGE_BASE(1) + 2) +#define TAS5754M_UNDERVOLTAGE_PROT (TAS5754M_PAGE_BASE(1) + 5) +#define TAS5754M_ANALOG_MUTE_CTRL (TAS5754M_PAGE_BASE(1) + 6) +#define TAS5754M_ANALOG_GAIN_BOOST (TAS5754M_PAGE_BASE(1) + 7) +#define TAS5754M_VCOM_CTRL_1 (TAS5754M_PAGE_BASE(1) + 8) +#define TAS5754M_VCOM_CTRL_2 (TAS5754M_PAGE_BASE(1) + 9) + +#define TAS5754M_CRAM_CTRL (TAS5754M_PAGE_BASE(44) + 1) + +#define TAS5754M_FLEX_A (TAS5754M_PAGE_BASE(253) + 63) +#define TAS5754M_FLEX_B (TAS5754M_PAGE_BASE(253) + 64) + +#define TAS5754M_MAX_REGISTER (TAS5754M_PAGE_BASE(253) + 64) + +/* Page 0, Register 1 - reset */ +#define TAS5754M_RSTR (1 << 0) +#define TAS5754M_RSTM (1 << 4) + +/* Page 0, Register 2 - power */ +#define TAS5754M_RQPD (1 << 0) +#define TAS5754M_RQPD_SHIFT 0 +#define TAS5754M_RQST (1 << 4) +#define TAS5754M_RQST_SHIFT 4 + +/* Page 0, Register 3 - mute */ +#define TAS5754M_RQMR (1 << 0) +#define TAS5754M_RQMR_SHIFT 0 +#define TAS5754M_RQML (1 << 4) +#define TAS5754M_RQML_SHIFT 4 + +/* Page 0, Register 4 - PLL */ +#define TAS5754M_PLLE (1 << 0) +#define TAS5754M_PLLE_SHIFT 0 +#define TAS5754M_PLCK (1 << 4) +#define TAS5754M_PLCK_SHIFT 4 + +/* Page 0, Register 7 - DSP */ +#define TAS5754M_SDSL (1 << 0) +#define TAS5754M_SDSL_SHIFT 0 +#define TAS5754M_DEMP (1 << 4) +#define TAS5754M_DEMP_SHIFT 4 + +/* Page 0, Register 8 - GPIO output enable */ +#define TAS5754M_G1OE (1 << 0) +#define TAS5754M_G2OE (1 << 1) +#define TAS5754M_G3OE (1 << 2) +#define TAS5754M_G4OE (1 << 3) +#define TAS5754M_G5OE (1 << 4) +#define TAS5754M_G6OE (1 << 5) + +/* Page 0, Register 9 - BCK, LRCLK configuration */ +#define TAS5754M_LRKO (1 << 0) +#define TAS5754M_LRKO_SHIFT 0 +#define TAS5754M_BCKO (1 << 4) +#define TAS5754M_BCKO_SHIFT 4 +#define TAS5754M_BCKP (1 << 5) +#define TAS5754M_BCKP_SHIFT 5 + +/* Page 0, Register 12 - Master mode BCK, LRCLK reset */ +#define TAS5754M_RLRK (1 << 0) +#define TAS5754M_RLRK_SHIFT 0 +#define TAS5754M_RBCK (1 << 1) +#define TAS5754M_RBCK_SHIFT 1 + +/* Page 0, Register 13 - PLL reference */ +#define TAS5754M_SREF (7 << 4) +#define TAS5754M_SREF_SHIFT 4 +#define TAS5754M_SREF_SCK (0 << 4) +#define TAS5754M_SREF_BCK (1 << 4) +#define TAS5754M_SREF_GPIO (3 << 4) + +/* Page 0, Register 14 - DAC reference */ +#define TAS5754M_SDAC (7 << 4) +#define TAS5754M_SDAC_SHIFT 4 +#define TAS5754M_SDAC_MCK (0 << 4) +#define TAS5754M_SDAC_PLL (1 << 4) +#define TAS5754M_SDAC_SCK (3 << 4) +#define TAS5754M_SDAC_BCK (4 << 4) +#define TAS5754M_SDAC_GPIO (5 << 4) + +/* Page 0, Register 16, 18 - GPIO source for DAC, PLL */ +#define TAS5754M_GREF (7 << 0) +#define TAS5754M_GREF_SHIFT 0 +#define TAS5754M_GREF_GPIO1 (0 << 0) +#define TAS5754M_GREF_GPIO2 (1 << 0) +#define TAS5754M_GREF_GPIO3 (2 << 0) +#define TAS5754M_GREF_GPIO4 (3 << 0) +#define TAS5754M_GREF_GPIO5 (4 << 0) +#define TAS5754M_GREF_GPIO6 (5 << 0) + +/* Page 0, Register 19 - synchronize */ +#define TAS5754M_RQSY (1 << 0) +#define TAS5754M_RQSY_RESUME (0 << 0) +#define TAS5754M_RQSY_HALT (1 << 0) + +/* Page 0, Register 34 - fs speed mode */ +#define TAS5754M_FSSP (3 << 0) +#define TAS5754M_FSSP_SHIFT 0 +#define TAS5754M_FSSP_48KHZ (0 << 0) +#define TAS5754M_FSSP_96KHZ (1 << 0) +#define TAS5754M_FSSP_192KHZ (2 << 0) +#define TAS5754M_FSSP_384KHZ (3 << 0) + +/* Page 0, Register 37 - Error detection */ +#define TAS5754M_IPLK (1 << 0) +#define TAS5754M_DCAS (1 << 1) +#define TAS5754M_IDCM (1 << 2) +#define TAS5754M_IDCH (1 << 3) +#define TAS5754M_IDSK (1 << 4) +#define TAS5754M_IDBK (1 << 5) +#define TAS5754M_IDFS (1 << 6) + +/* Page 0, Register 40 - I2S configuration */ +#define TAS5754M_ALEN (3 << 0) +#define TAS5754M_ALEN_SHIFT 0 +#define TAS5754M_ALEN_16 (0 << 0) +#define TAS5754M_ALEN_20 (1 << 0) +#define TAS5754M_ALEN_24 (2 << 0) +#define TAS5754M_ALEN_32 (3 << 0) +#define TAS5754M_AFMT (3 << 4) +#define TAS5754M_AFMT_SHIFT 4 +#define TAS5754M_AFMT_I2S (0 << 4) +#define TAS5754M_AFMT_DSP (1 << 4) +#define TAS5754M_AFMT_RTJ (2 << 4) +#define TAS5754M_AFMT_LTJ (3 << 4) + +/* Page 0, Register 42 - DAC routing */ +#define TAS5754M_AUPR_SHIFT 0 +#define TAS5754M_AUPL_SHIFT 4 + +/* Page 0, Register 59 - auto mute */ +#define TAS5754M_ATMR_SHIFT 0 +#define TAS5754M_ATML_SHIFT 4 + +/* Page 0, Register 63 - ramp rates */ +#define TAS5754M_VNDF_SHIFT 6 +#define TAS5754M_VNDS_SHIFT 4 +#define TAS5754M_VNUF_SHIFT 2 +#define TAS5754M_VNUS_SHIFT 0 + +/* Page 0, Register 64 - emergency ramp rates */ +#define TAS5754M_VEDF_SHIFT 6 +#define TAS5754M_VEDS_SHIFT 4 + +/* Page 0, Register 65 - Digital mute enables */ +#define TAS5754M_ACTL_SHIFT 2 +#define TAS5754M_AMLE_SHIFT 1 +#define TAS5754M_AMRE_SHIFT 0 + +/* Page 0, Register 80-85, GPIO output selection */ +#define TAS5754M_GxSL (31 << 0) +#define TAS5754M_GxSL_SHIFT 0 +#define TAS5754M_GxSL_OFF (0 << 0) +#define TAS5754M_GxSL_DSP (1 << 0) +#define TAS5754M_GxSL_REG (2 << 0) +#define TAS5754M_GxSL_AMUTB (3 << 0) +#define TAS5754M_GxSL_AMUTL (4 << 0) +#define TAS5754M_GxSL_AMUTR (5 << 0) +#define TAS5754M_GxSL_CLKI (6 << 0) +#define TAS5754M_GxSL_SDOUT (7 << 0) +#define TAS5754M_GxSL_ANMUL (8 << 0) +#define TAS5754M_GxSL_ANMUR (9 << 0) +#define TAS5754M_GxSL_PLLLK (10 << 0) +#define TAS5754M_GxSL_CPCLK (11 << 0) +#define TAS5754M_GxSL_UV0_7 (14 << 0) +#define TAS5754M_GxSL_UV0_3 (15 << 0) +#define TAS5754M_GxSL_PLLCK (16 << 0) + +/* Page 1, Register 2 - analog volume control */ +#define TAS5754M_RAGN_SHIFT 0 +#define TAS5754M_LAGN_SHIFT 4 + +/* Page 1, Register 7 - analog boost control */ +#define TAS5754M_AGBR_SHIFT 0 +#define TAS5754M_AGBL_SHIFT 4 + +#endif -- 2.17.1

2 2

[PATCH] ALSA: hda/realtek: Add quirk for ASUS UX550VE
by Takashi Iwai 07 Nov '21

07 Nov '21

ASUS UX550VE (SSID 1043:1970) requires a similar workaround for managing the routing of the 4 speakers like some other ASUS models. Add a corresponding quirk entry for fixing it. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212641 Cc: <stable(a)vger.kernel.org> 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 3601749f2df6..a6098c942aeb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8698,6 +8698,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), + SND_PCI_QUIRK(0x1043, 0x1970, "ASUS UX550VE", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1982, "ASUS B1400CEPE", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), -- 2.26.2

1 0

Re: [Alsa-user] USB audio on riscv64
by Kip Warner 07 Nov '21

07 Nov '21

On Sat, 2021-11-06 at 14:50 -0700, chris hermansen wrote: > To allow remote users to use ALSA, you need to add those users to the > audio group, however this is not recommended by default (see note > below). Thanks Chris. For anyone else having a similar problem, adding myself to the audio group, like so, fixed the problem: $ sudo usermod -a -G audio kip -- Kip Warner -- Senior Software Engineer OpenPGP signed/encrypted mail preferred https://www.thevertigo.com

1 0

Re: [Alsa-user] USB audio on riscv64
by Kip Warner 06 Nov '21

06 Nov '21

On Thu, 2021-11-04 at 08:40 +0000, Dmitri Seletski wrote: > I am just a user. > Can't help but wonder if usb snd modules were compiled in or loaded? Hey Dmitri, CC'ing the lists. When I plug in the device, I see the following on the bus: $ lsusb ... Bus 001 Device 003: ID 1908:2070 GEMBIRD Honk HK-5002 USB Speaker ... The kernel logs the following in kern.log: [ 291.681086] usb 1-2.3: new full-speed USB device number 3 using xhci_hcd [ 291.850299] usb 1-2.3: New USB device found, idVendor=1908, idProduct=2070, bcdDevice= 1.00 [ 291.850333] usb 1-2.3: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 291.850345] usb 1-2.3: Product: USB2.0 Device [ 291.850355] usb 1-2.3: Manufacturer: Generic [ 291.850365] usb 1-2.3: SerialNumber: 20130100ph0 [ 292.434209] mc: Linux media interface: v0.10 [ 292.734949] usb 1-2.3: 1:1: cannot get freq at ep 0x2 [ 292.802612] usb 1-2.3: 2:0: bogus dB values (-12800/-12700), disabling dB reporting [ 292.803733] usbcore: registered new interface driver snd-usb-audio The device still doesn't appear to be visible to ALSA: $ aplay -l aplay: device_list:276: no soundcards found... If I try playing an Ogg/Vorbis file with mpv, I see the following with no sound: $ mpv "/usr/share/games/lincity-ng/music/default/01 - pronobozo - lincity.ogg" (+) Audio --aid=1 (vorbis 2ch 44100Hz) AO: [pulse] 44100Hz stereo 2ch float A: 00:01:00 / 00:03:30 (28%) The device works fine on arm64 via a Raspberry Pi 4. -- Kip Warner -- Senior Software Engineer OpenPGP signed/encrypted mail preferred https://www.thevertigo.com

2 8

Need info: How can I have conditional includes in a topology conf file
by GitHub issues - opened 05 Nov '21

05 Nov '21

alsa-project/alsa-utils issue #118 was opened from ranj063: For example: I want to be able to conditionally include the DMIC pipelines/DAI's depending on whether the device has DMICs or not. Issue URL : https://github.com/alsa-project/alsa-utils/issues/118 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/9] ASoC: Ietel: Add Dell ADL products support
by Bard Liao 05 Nov '21

05 Nov '21

This series adds configurations for Dell ADL products. Gongjun Song (9): ASoC: Intel: sof_sdw: Add support for SKU 0AF3 product ASoC: Intel: soc-acpi: add SKU 0AF3 SoundWire configuration ASoC: Intel: sof_sdw: Add support for SKU 0B00 and 0B01 products ASoC: Intel: sof_sdw: Add support for SKU 0B11 product ASoC: Intel: sof_sdw: Add support for SKU 0B13 product ASoC: Intel: soc-acpi: add SKU 0B13 SoundWire configuration ASoC: Intel: sof_sdw: Add support for SKU 0B29 product ASoC: Intel: soc-acpi: add SKU 0B29 SoundWire configuration ASoC: Intel: sof_sdw: Add support for SKU 0B12 product sound/soc/intel/boards/sof_sdw.c | 69 ++++++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 105 ++++++++++++++++++ 2 files changed, 174 insertions(+) -- 2.17.1

2 10

[PATCH] ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk
by AngeloGioacchino Del Regno 05 Nov '21

05 Nov '21

Some RT5645 and RT5650 powered platforms are using "Ext Spk" instead of "Speaker", and this is also reflected in alsa-lib configurations for the generic RT5645 usecase manager configs. Rename the "Speaker" control to "Ext Spk" in order to be able to make the userspace reuse/inherit the same configurations also for this machine, along with the others. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index c28ebf891cb0..2cbf679f5c74 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -30,15 +30,15 @@ static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { }; static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { - {"Speaker", NULL, "SPOL"}, - {"Speaker", NULL, "SPOR"}, + {"Ext Spk", NULL, "SPOL"}, + {"Ext Spk", NULL, "SPOR"}, {"DMIC L1", NULL, "Int Mic"}, {"DMIC R1", NULL, "Int Mic"}, {"Headphone", NULL, "HPOL"}, @@ -48,7 +48,7 @@ static const struct snd_soc_dapm_route mt8173_rt5650_routes[] = { }; static const struct snd_kcontrol_new mt8173_rt5650_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), -- 2.33.1

2 1

[PATCH] ASoC: SOF: Intel: hda: fix hotplug when only codec is suspended
by Kai Vehmanen 05 Nov '21

05 Nov '21

If codec is in runtime suspend, but controller is not, hotplug events are missed as the codec has no way to alert the controller. Problem does not occur if both controller and codec are active, or when both are suspended. An easy way to reproduce is to play an audio stream on one codec (e.g. to HDMI/DP display codec), wait for other HDA codec to go to runtime suspend, and then plug in a headset to the suspended codec. The jack event is not reported correctly in this case. Another way to reproduce is to force controller to stay active with "snd_sof_pci.sof_pci_debug=0x1" Fix the issue by reconfiguring the WAKEEN register when powering up/down individual links, and handling control events in the interrupt handler. Fixes: 87fc20e4a0cb ("ASoC: SOF: Intel: hda: use hdac_ext fine-grained link management") Reported-by: Hui Wang <hui.wang(a)canonical.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(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> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-bus.c | 17 +++++++++++++++++ sound/soc/sof/intel/hda-dsp.c | 3 +-- sound/soc/sof/intel/hda.c | 16 ++++++++++++++++ 3 files changed, 34 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c index 30025d3c16b6..0862ff8b6627 100644 --- a/sound/soc/sof/intel/hda-bus.c +++ b/sound/soc/sof/intel/hda-bus.c @@ -10,6 +10,8 @@ #include <linux/io.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> +#include <sound/hda_codec.h> +#include <sound/hda_register.h> #include "../sof-priv.h" #include "hda.h" @@ -21,6 +23,18 @@ #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) +static void update_codec_wake_enable(struct hdac_bus *bus, unsigned int addr, bool link_power) +{ + unsigned int mask = snd_hdac_chip_readw(bus, WAKEEN); + + if (link_power) + mask &= ~BIT(addr); + else + mask |= BIT(addr); + + snd_hdac_chip_updatew(bus, WAKEEN, STATESTS_INT_MASK, mask); +} + static void sof_hda_bus_link_power(struct hdac_device *codec, bool enable) { struct hdac_bus *bus = codec->bus; @@ -41,6 +55,9 @@ static void sof_hda_bus_link_power(struct hdac_device *codec, bool enable) */ if (codec->addr == HDA_IDISP_ADDR && !enable) snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); + + /* WAKEEN needs to be set for disabled links */ + update_codec_wake_enable(bus, codec->addr, enable); } static const struct hdac_bus_ops bus_core_ops = { diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 623cf291e207..262a70791a8f 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -623,8 +623,7 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) hda_dsp_ipc_int_disable(sdev); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - if (runtime_suspend) - hda_codec_jack_wake_enable(sdev, true); + hda_codec_jack_wake_enable(sdev, runtime_suspend); /* power down all hda link */ snd_hdac_ext_bus_link_power_down_all(bus); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index f60e2c57d3d0..ef92cca7ae01 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -696,6 +696,20 @@ static int hda_init_caps(struct snd_sof_dev *sdev) return 0; } +static void hda_check_for_state_change(struct snd_sof_dev *sdev) +{ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + struct hdac_bus *bus = sof_to_bus(sdev); + unsigned int codec_mask; + + codec_mask = snd_hdac_chip_readw(bus, STATESTS); + if (codec_mask) { + hda_codec_jack_check(sdev); + snd_hdac_chip_writew(bus, STATESTS, codec_mask); + } +#endif +} + static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context) { struct snd_sof_dev *sdev = context; @@ -737,6 +751,8 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) if (hda_sdw_check_wakeen_irq(sdev)) hda_sdw_process_wakeen(sdev); + hda_check_for_state_change(sdev); + /* enable GIE interrupt */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, base-commit: 5d03907bbf9ccf10e0d2cfb4f4d312b7cc4274f4 -- 2.33.0

2 1

[PATCH] ASoC: DAPM: Cover regression by kctl change notification fix
by Takashi Iwai 05 Nov '21

05 Nov '21

The recent fix for DAPM to correct the kctl change notification by the commit 5af82c81b2c4 ("ASoC: DAPM: Fix missing kctl change notifications") caused other regressions since it changed the behavior of snd_soc_dapm_set_pin() that is called from several API functions. Formerly it returned always 0 for success, but now it returns 0 or 1. This patch addresses it, restoring the old behavior of snd_soc_dapm_set_pin() while keeping the fix in snd_soc_dapm_put_pin_switch(). Fixes: 5af82c81b2c4 ("ASoC: DAPM: Fix missing kctl change notifications") Reported-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 29 +++++++++++++++++++++++------ 1 file changed, 23 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2892b0aba151..b06c5682445c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2559,8 +2559,13 @@ static struct snd_soc_dapm_widget *dapm_find_widget( return NULL; } -static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, - const char *pin, int status) +/* + * set the DAPM pin status: + * returns 1 when the value has been updated, 0 when unchanged, or a negative + * error code; called from kcontrol put callback + */ +static int __snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, + const char *pin, int status) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); int ret = 0; @@ -2586,6 +2591,18 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, return ret; } +/* + * similar as __snd_soc_dapm_set_pin(), but returns 0 when successful; + * called from several API functions below + */ +static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, + const char *pin, int status) +{ + int ret = __snd_soc_dapm_set_pin(dapm, pin, status); + + return ret < 0 ? ret : 0; +} + /** * snd_soc_dapm_sync_unlocked - scan and power dapm paths * @dapm: DAPM context @@ -3589,10 +3606,10 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, const char *pin = (const char *)kcontrol->private_value; int ret; - if (ucontrol->value.integer.value[0]) - ret = snd_soc_dapm_enable_pin(&card->dapm, pin); - else - ret = snd_soc_dapm_disable_pin(&card->dapm, pin); + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + ret = __snd_soc_dapm_set_pin(&card->dapm, pin, + !!ucontrol->value.integer.value[0]); + mutex_unlock(&card->dapm_mutex); snd_soc_dapm_sync(&card->dapm); return ret; -- 2.31.1

3 4

[PATCH] ALSA: memalloc: Catch call with NULL snd_dma_buffer pointer
by Takashi Iwai 05 Nov '21

05 Nov '21

Although we've covered all calls with NULL dma buffer pointer, so far, there may be still some else in the wild. For catching such a case more easily, add a WARN_ON_ONCE() in snd_dma_get_ops(). Fixes: 37af81c5998f ("ALSA: core: Abstract memory alloc helpers") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index ea778f868cf3..ea9698cea2e3 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -631,6 +631,8 @@ static const struct snd_malloc_ops *dma_ops[] = { static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab) { + if (WARN_ON_ONCE(!dmab)) + return NULL; if (WARN_ON_ONCE(dmab->dev.type <= SNDRV_DMA_TYPE_UNKNOWN || dmab->dev.type >= ARRAY_SIZE(dma_ops))) return NULL; -- 2.31.1

1 0

The regression caused by snd_soc_dapm_enable_pin
by Yu-Hsuan Hsu 05 Nov '21

05 Nov '21

Hi Takashi, The patch 5af82c81b2c49cfb1cad84d9eb6eab0e3d1c4842(ASoC: DAPM: Fix missing kctl change notifications) caused the regression on some ChromeBook. The reason is that some drivers check the return value of snd_soc_dapm_enable_pin, like kabylake_ssp0_trigger(Which caused a regression). In addition, some SOF drivers may be also affected by this change(e.g. sof_sdw_max98373.c). Could you help to fix it? Thanks. Best, Yu-Hsuan

2 5

[PATCH v2] ALSA: timer: Unconditionally unlink slave instances, too
by Takashi Iwai 05 Nov '21

05 Nov '21

Like the previous fix (commit c0317c0e8709 "ALSA: timer: Fix use-after-free problem"), we have to unlink slave timer instances immediately at snd_timer_stop(), too. Otherwise it may leave a stale entry in the list if the slave instance is freed before actually running. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: return -EBUSY again if the instance isn't running sound/core/timer.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 4f9bab931951..b3214baa8919 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -665,23 +665,22 @@ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) static int snd_timer_stop_slave(struct snd_timer_instance *timeri, bool stop) { unsigned long flags; + bool running; spin_lock_irqsave(&slave_active_lock, flags); - if (!(timeri->flags & SNDRV_TIMER_IFLG_RUNNING)) { - spin_unlock_irqrestore(&slave_active_lock, flags); - return -EBUSY; - } + running = timeri->flags & SNDRV_TIMER_IFLG_RUNNING; timeri->flags &= ~SNDRV_TIMER_IFLG_RUNNING; if (timeri->timer) { spin_lock(&timeri->timer->lock); list_del_init(&timeri->ack_list); list_del_init(&timeri->active_list); - snd_timer_notify1(timeri, stop ? SNDRV_TIMER_EVENT_STOP : - SNDRV_TIMER_EVENT_PAUSE); + if (running) + snd_timer_notify1(timeri, stop ? SNDRV_TIMER_EVENT_STOP : + SNDRV_TIMER_EVENT_PAUSE); spin_unlock(&timeri->timer->lock); } spin_unlock_irqrestore(&slave_active_lock, flags); - return 0; + return running ? 0 : -EBUSY; } /* -- 2.26.2

1 0

[PATCH] ALSA: timer: Unconditionally unlink slave instances, too
by Takashi Iwai 05 Nov '21

05 Nov '21

Like the previous fix (commit c0317c0e8709 "ALSA: timer: Fix use-after-free problem"), we have to unlink slave timer instances immediately at snd_timer_stop(), too. Otherwise it may leave a stale entry in the list if the slave instance is freed before actually running. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 4f9bab931951..5749e5275ba5 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -665,19 +665,18 @@ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) static int snd_timer_stop_slave(struct snd_timer_instance *timeri, bool stop) { unsigned long flags; + bool running; spin_lock_irqsave(&slave_active_lock, flags); - if (!(timeri->flags & SNDRV_TIMER_IFLG_RUNNING)) { - spin_unlock_irqrestore(&slave_active_lock, flags); - return -EBUSY; - } + running = timeri->flags & SNDRV_TIMER_IFLG_RUNNING; timeri->flags &= ~SNDRV_TIMER_IFLG_RUNNING; if (timeri->timer) { spin_lock(&timeri->timer->lock); list_del_init(&timeri->ack_list); list_del_init(&timeri->active_list); - snd_timer_notify1(timeri, stop ? SNDRV_TIMER_EVENT_STOP : - SNDRV_TIMER_EVENT_PAUSE); + if (running) + snd_timer_notify1(timeri, stop ? SNDRV_TIMER_EVENT_STOP : + SNDRV_TIMER_EVENT_PAUSE); spin_unlock(&timeri->timer->lock); } spin_unlock_irqrestore(&slave_active_lock, flags); -- 2.26.2

1 1

[PATCH v3 2/2] ASoC: codec: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 04 Nov '21

04 Nov '21

New codec driver for Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1071 ++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 379 +++++++++++ 4 files changed, 1459 insertions(+) create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 326f2d611ad4..b81c4721c0d5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -220,6 +220,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TDA7419 imply SND_SOC_TFA9879 imply SND_SOC_TFA989X + imply SND_SOC_TLV320ADC3XXX imply SND_SOC_TLV320ADCX140 imply SND_SOC_TLV320AIC23_I2C imply SND_SOC_TLV320AIC23_SPI @@ -1486,6 +1487,12 @@ config SND_SOC_TFA989X Note that the driver currently bypasses the built-in "CoolFlux DSP" and does not support (hardware) volume control. +config SND_SOC_TLV320ADC3XXX + tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" + help + Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 + ADCs. + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9acfbcbfc46d..5112c12e6d76 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -238,6 +238,7 @@ snd-soc-tda7419-objs := tda7419.o snd-soc-tas2770-objs := tas2770.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o +snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o @@ -572,6 +573,7 @@ obj-$(CONFIG_SND_SOC_TDA7419) += snd-soc-tda7419.o obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TFA989X) += snd-soc-tfa989x.o +obj-$(CONFIG_SND_SOC_TLV320ADC3XXX) += snd-soc-tlv320adc3xxx.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c new file mode 100644 index 000000000000..64e86e3b687a --- /dev/null +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -0,0 +1,1071 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov +// +// History: +// +// Author: "Shahina Shaik" < shahina.s(a)mistralsolutions.com > +// Copyright: (C) 2010 Mistral Solutions Pvt Ltd. +// +// Author: Dongge wu <dgwu(a)ambarella.com> +// 2015/10/28 - [Dongge wu] Created file +// Copyright (C) 2014-2018, Ambarella, Inc. +// +// Author: Ricard Wanderlof <ricardw(a)axis.com> +// 2020/11/05: Fixing driver for Linux 4.14: more clocking modes, etc. +// 2021/09/03: Porting driver to Linux 5.4 and later. Added GPIO pin control, +// DAPM controlled clock chain, general cleanup. +// Copyright (C) 2021 Axis Communications AB +// + +#include <dt-bindings/sound/tlv320adc3xxx.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio/driver.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/cdev.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "tlv320adc3xxx.h" + +enum adc3xxx_type { + ADC3001 = 0, + ADC3101 +}; + +struct adc3xxx { + struct device *dev; + enum adc3xxx_type type; + struct clk *mclk; + struct regmap *regmap; + unsigned int pll_mode; + unsigned int rst_pin; + unsigned int rst_active; + unsigned int sysclk; + unsigned int gpio_cfg[ADC3XXX_GPIOS_MAX]; /* value+1 (0 => not set) */ + unsigned int micbias_vg[ADC3XXX_MICBIAS_PINS]; + int master; + u8 page_no; + int use_pll; +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif +}; + +static const unsigned int adc3xxx_gpio_ctrl_reg[ADC3XXX_GPIOS_MAX] = { + ADC3XXX_GPIO1_CTRL, + ADC3XXX_GPIO2_CTRL +}; + +static const unsigned int adc3xxx_micbias_shift[ADC3XXX_MICBIAS_PINS] = { + ADC3XXX_MICBIAS1_SHIFT, + ADC3XXX_MICBIAS2_SHIFT +}; + +static const struct reg_default adc3xxx_defaults[] = { + /* Page 0 */ + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, { 3, 0x00 }, + { 4, 0x00 }, { 5, 0x11 }, { 6, 0x04 }, { 7, 0x00 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x00 }, { 11, 0x00 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x00 }, { 15, 0x00 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x01 }, { 19, 0x01 }, + { 20, 0x80 }, { 21, 0x80 }, { 22, 0x04 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x01 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x02 }, { 30, 0x01 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x10 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x00 }, { 38, 0x02 }, { 39, 0x00 }, + { 40, 0x00 }, { 41, 0x00 }, { 42, 0x00 }, { 43, 0x00 }, + { 44, 0x00 }, { 45, 0x00 }, { 46, 0x00 }, { 47, 0x00 }, + { 48, 0x00 }, { 49, 0x00 }, { 50, 0x00 }, { 51, 0x00 }, + { 52, 0x00 }, { 53, 0x12 }, { 54, 0x00 }, { 55, 0x00 }, + { 56, 0x00 }, { 57, 0x00 }, { 58, 0x00 }, { 59, 0x44 }, + { 60, 0x00 }, { 61, 0x01 }, { 62, 0x00 }, { 63, 0x00 }, + { 64, 0x00 }, { 65, 0x00 }, { 66, 0x00 }, { 67, 0x00 }, + { 68, 0x00 }, { 69, 0x00 }, { 70, 0x00 }, { 71, 0x00 }, + { 72, 0x00 }, { 73, 0x00 }, { 74, 0x00 }, { 75, 0x00 }, + { 76, 0x00 }, { 77, 0x00 }, { 78, 0x00 }, { 79, 0x00 }, + { 80, 0x00 }, { 81, 0x00 }, { 82, 0x88 }, { 83, 0x00 }, + { 84, 0x00 }, { 85, 0x00 }, { 86, 0x00 }, { 87, 0x00 }, + { 88, 0x7f }, { 89, 0x00 }, { 90, 0x00 }, { 91, 0x00 }, + { 92, 0x00 }, { 93, 0x00 }, { 94, 0x00 }, { 95, 0x00 }, + { 96, 0x7f }, { 97, 0x00 }, { 98, 0x00 }, { 99, 0x00 }, + { 100, 0x00 }, { 101, 0x00 }, { 102, 0x00 }, { 103, 0x00 }, + { 104, 0x00 }, { 105, 0x00 }, { 106, 0x00 }, { 107, 0x00 }, + { 108, 0x00 }, { 109, 0x00 }, { 110, 0x00 }, { 111, 0x00 }, + { 112, 0x00 }, { 113, 0x00 }, { 114, 0x00 }, { 115, 0x00 }, + { 116, 0x00 }, { 117, 0x00 }, { 118, 0x00 }, { 119, 0x00 }, + { 120, 0x00 }, { 121, 0x00 }, { 122, 0x00 }, { 123, 0x00 }, + { 124, 0x00 }, { 125, 0x00 }, { 126, 0x00 }, { 127, 0x00 }, + + /* Page 1 */ + { 128, 0x00 }, { 129, 0x00 }, { 130, 0x00 }, { 131, 0x00 }, + { 132, 0x00 }, { 133, 0x00 }, { 134, 0x00 }, { 135, 0x00 }, + { 136, 0x00 }, { 137, 0x00 }, { 138, 0x00 }, { 139, 0x00 }, + { 140, 0x00 }, { 141, 0x00 }, { 142, 0x00 }, { 143, 0x00 }, + { 144, 0x00 }, { 145, 0x00 }, { 146, 0x00 }, { 147, 0x00 }, + { 148, 0x00 }, { 149, 0x00 }, { 150, 0x00 }, { 151, 0x00 }, + { 152, 0x00 }, { 153, 0x00 }, { 154, 0x00 }, { 155, 0x00 }, + { 156, 0x00 }, { 157, 0x00 }, { 158, 0x00 }, { 159, 0x00 }, + { 160, 0x00 }, { 161, 0x00 }, { 162, 0x00 }, { 163, 0x00 }, + { 164, 0x00 }, { 165, 0x00 }, { 166, 0x00 }, { 167, 0x00 }, + { 168, 0x00 }, { 169, 0x00 }, { 170, 0x00 }, { 171, 0x00 }, + { 172, 0x00 }, { 173, 0x00 }, { 174, 0x00 }, { 175, 0x00 }, + { 176, 0x00 }, { 177, 0x00 }, { 178, 0x00 }, { 179, 0x00 }, + { 180, 0xff }, { 181, 0x00 }, { 182, 0x3f }, { 183, 0xff }, + { 184, 0x00 }, { 185, 0x3f }, { 186, 0x00 }, { 187, 0x80 }, + { 188, 0x80 }, { 189, 0x00 }, { 190, 0x00 }, { 191, 0x00 }, +}; + +static bool adc3xxx_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADC3XXX_RESET: + return true; + default: + return false; + } +} + +static const struct regmap_range_cfg adc3xxx_ranges[] = { + { + .range_min = 0, + .range_max = 2 * ADC3XXX_PAGE_SIZE, + .selector_reg = ADC3XXX_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = ADC3XXX_PAGE_SIZE, + } +}; + +static const struct regmap_config adc3xxx_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = adc3xxx_defaults, + .num_reg_defaults = ARRAY_SIZE(adc3xxx_defaults), + + .volatile_reg = adc3xxx_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + + .ranges = adc3xxx_ranges, + .num_ranges = ARRAY_SIZE(adc3xxx_ranges), + .max_register = 2 * ADC3XXX_PAGE_SIZE, +}; + +struct adc3xxx_rate_divs { + u32 mclk; + u32 rate; + u8 pll_p; + u8 pll_r; + u8 pll_j; + u16 pll_d; + u8 nadc; + u8 madc; + u8 aosr; +}; + +/* + * PLL and Clock settings. + * If p member is 0, PLL is not used. + * The order of the entries in this table have the PLL entries before + * the non-PLL entries, so that the PLL modes are preferred unless + * the PLL mode setting says otherwise. + */ +static const struct adc3xxx_rate_divs adc3xxx_divs[] = { + /* mclk, rate, p, r, j, d, nadc, madc, aosr */ + /* 8k rate */ + { 12000000, 8000, 1, 1, 7, 1680, 42, 2, 128 }, + { 12288000, 8000, 1, 1, 7, 0000, 42, 2, 128 }, + /* 11.025k rate */ + { 12000000, 11025, 1, 1, 6, 8208, 29, 2, 128 }, + /* 16k rate */ + { 12000000, 16000, 1, 1, 7, 1680, 21, 2, 128 }, + { 12288000, 16000, 1, 1, 7, 0000, 21, 2, 128 }, + /* 22.05k rate */ + { 12000000, 22050, 1, 1, 7, 560, 15, 2, 128 }, + /* 32k rate */ + { 12000000, 32000, 1, 1, 8, 1920, 12, 2, 128 }, + { 12288000, 32000, 1, 1, 8, 0000, 12, 2, 128 }, + /* 44.1k rate */ + { 12000000, 44100, 1, 1, 7, 5264, 8, 2, 128 }, + /* 48k rate */ + { 12000000, 48000, 1, 1, 7, 1680, 7, 2, 128 }, + { 12288000, 48000, 1, 1, 7, 0000, 7, 2, 128 }, + { 24576000, 48000, 1, 1, 3, 5000, 7, 2, 128 }, /* With PLL */ + { 24576000, 48000, 0, 0, 0, 0000, 2, 2, 128 }, /* Without PLL */ + /* 88.2k rate */ + { 12000000, 88200, 1, 1, 7, 5264, 4, 4, 64 }, + /* 96k rate */ + { 12000000, 96000, 1, 1, 8, 1920, 4, 4, 64 }, +}; + +static int adc3xxx_get_divs(struct device *dev, int mclk, int rate, int pll_mode) +{ + int i; + + dev_dbg(dev, "mclk = %d, rate = %d, clock mode %u\n", + mclk, rate, pll_mode); + for (i = 0; i < ARRAY_SIZE(adc3xxx_divs); i++) { + const struct adc3xxx_rate_divs *mode = &adc3xxx_divs[i]; + + /* Skip this entry if it doesn't fulfill the intended clock + * mode requirement. We consider anything besides the two + * modes below to be the same as ADC3XXX_PLL_AUTO. + */ + if ((pll_mode == ADC3XXX_PLL_BYPASS && mode->pll_p) || + (pll_mode == ADC3XXX_PLL_ENABLE && !mode->pll_p)) + continue; + + if (mode->rate == rate && mode->mclk == mclk) + return i; + } + + dev_info(dev, "Master clock rate %d and sample rate %d is not supported\n", + mclk, rate); + return -EINVAL; +} + +static int adc3xxx_pll_delay(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* 10msec delay needed after PLL power-up to allow + * PLL and dividers to stabilize (datasheet p13). + */ + usleep_range(10000, 20000); + + return 0; +} + +static const char * const adc_softstepping_text[] = { "1 step", "2 step", "off" }; +static const SOC_ENUM_SINGLE_DECL(adc_softstepping_enum, ADC3XXX_ADC_DIGITAL, 0, + adc_softstepping_text); + +static const char * const multiplier_text[] = { "1", "2", "4", "8", "16", "32", "64", "128" }; +static const SOC_ENUM_SINGLE_DECL(left_agc_attack_mult_enum, + ADC3XXX_LEFT_CHN_AGC_4, 0, multiplier_text); +static const SOC_ENUM_SINGLE_DECL(right_agc_attack_mult_enum, + ADC3XXX_RIGHT_CHN_AGC_4, 0, multiplier_text); +static const SOC_ENUM_SINGLE_DECL(left_agc_decay_mult_enum, + ADC3XXX_LEFT_CHN_AGC_5, 0, multiplier_text); +static const SOC_ENUM_SINGLE_DECL(right_agc_decay_mult_enum, + ADC3XXX_RIGHT_CHN_AGC_5, 0, multiplier_text); + +static const char * const dither_dc_offset_text[] = { + "0mV", "15mV", "30mV", "45mV", "60mV", "75mV", "90mV", "105mV", + "-15mV", "-30mV", "-45mV", "-60mV", "-75mV", "-90mV", "-105mV" +}; +static const unsigned int dither_dc_offset_values[] = { + 0, 1, 2, 3, 4, 5, 6, 7, 9, 10, 11, 12, 13, 14, 15 +}; +static const SOC_VALUE_ENUM_DOUBLE_DECL(dither_dc_offset_enum, + ADC3XXX_DITHER_CTRL, + 4, 0, 0xf, dither_dc_offset_text, + dither_dc_offset_values); + +static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fine_tlv, -40, 10, 0); +/* AGC target: 8 values: -5.5, -8, -10, -12, -14, -17, -20, -24 dB */ +/* It would be nice to declare these in the order above, but empirically + * TLV_DB_SCALE_ITEM doesn't take lightly to the increment (second) parameter + * being negative, despite there being examples to the contrary in other + * drivers. So declare these in the order from lowest to highest, and + * set the invert flag in the SOC_DOUBLE_R_TLV declaration instead. + */ +static const DECLARE_TLV_DB_RANGE(agc_target_tlv, + 0, 0, TLV_DB_SCALE_ITEM(-2400, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(-2000, 300, 0), + 4, 6, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 7, 7, TLV_DB_SCALE_ITEM(-550, 0, 0)); +/* Since the 'disabled' value (mute) is at the highest value in the dB + * range (i.e. just before -32 dB) rather than the lowest, we need to resort + * to using a TLV_DB_RANGE in order to get the mute value in the right place. + */ +static const DECLARE_TLV_DB_RANGE(agc_thresh_tlv, + 0, 30, TLV_DB_SCALE_ITEM(-9000, 200, 0), + 31, 31, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +/* AGC hysteresis: 4 values: 1, 2, 4 dB, disabled (= mute) */ +static const DECLARE_TLV_DB_RANGE(agc_hysteresis_tlv, + 0, 1, TLV_DB_SCALE_ITEM(100, 100, 0), + 2, 2, TLV_DB_SCALE_ITEM(400, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +static const DECLARE_TLV_DB_SCALE(agc_max_tlv, 0, 50, 0); +/* Input attenuation: -6 dB or 0 dB */ +static const DECLARE_TLV_DB_SCALE(input_attenuation_tlv, -600, 600, 0); + +static const struct snd_kcontrol_new adc3xxx_snd_controls[] = { + SOC_DOUBLE_R_TLV("PGA Capture Volume", ADC3XXX_LEFT_APGA_CTRL, + ADC3XXX_RIGHT_APGA_CTRL, 0, 80, 0, pga_tlv), + SOC_DOUBLE("PGA Capture Switch", ADC3XXX_ADC_FGA, 7, 3, 1, 1), + SOC_DOUBLE_R("AGC Capture Switch", ADC3XXX_LEFT_CHN_AGC_1, + ADC3XXX_RIGHT_CHN_AGC_1, 7, 1, 0), + SOC_DOUBLE_R_TLV("AGC Target Level Capture Volume", ADC3XXX_LEFT_CHN_AGC_1, + ADC3XXX_RIGHT_CHN_AGC_2, 4, 0x07, 1, agc_target_tlv), + SOC_DOUBLE_R_TLV("AGC Noise Threshold Capture Volume", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 1, 0x1f, 1, agc_thresh_tlv), + SOC_DOUBLE_R_TLV("AGC Hysteresis Capture Volume", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 6, 3, 0, agc_hysteresis_tlv), + SOC_DOUBLE_R("AGC Clip Stepping Capture Switch", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 0, 1, 0), + /* + * Oddly enough, the data sheet says the default value + * for the left/right AGC maximum gain register field + * (ADC3XXX_LEFT/RIGHT_CHN_AGC_3 bits 0..6) is 0x7f = 127 + * (verified empirically) even though this value (indeed, above + * 0x50) is specified as 'Reserved. Do not use.' in the accompanying + * table in the data sheet. + */ + SOC_DOUBLE_R_TLV("AGC Maximum Capture Volume", ADC3XXX_LEFT_CHN_AGC_3, + ADC3XXX_RIGHT_CHN_AGC_3, 0, 0x50, 0, agc_max_tlv), + SOC_DOUBLE_R("AGC Attack Time", ADC3XXX_LEFT_CHN_AGC_4, + ADC3XXX_RIGHT_CHN_AGC_4, 3, 0x1f, 0), + /* Would like to have the multipliers as LR pairs, but there is + * no SOC_ENUM_foo which accepts two values in separate registers. + */ + SOC_ENUM("AGC Left Attack Time Multiplier", left_agc_attack_mult_enum), + SOC_ENUM("AGC Right Attack Time Multiplier", right_agc_attack_mult_enum), + SOC_DOUBLE_R("AGC Decay Time", ADC3XXX_LEFT_CHN_AGC_5, + ADC3XXX_RIGHT_CHN_AGC_5, 3, 0x1f, 0), + SOC_ENUM("AGC Left Decay Time Multiplier", left_agc_decay_mult_enum), + SOC_ENUM("AGC Right Decay Time Multiplier", right_agc_decay_mult_enum), + SOC_DOUBLE_R("AGC Noise Debounce", ADC3XXX_LEFT_CHN_AGC_6, + ADC3XXX_RIGHT_CHN_AGC_6, 0, 0x1f, 0), + SOC_DOUBLE_R("AGC Signal Debounce", ADC3XXX_LEFT_CHN_AGC_7, + ADC3XXX_RIGHT_CHN_AGC_7, 0, 0x0f, 0), + /* Read only register */ + SOC_DOUBLE_R_S_TLV("AGC Applied Capture Volume", ADC3XXX_LEFT_AGC_GAIN, + ADC3XXX_RIGHT_AGC_GAIN, 0, -24, 40, 6, 0, adc_tlv), + /* ADC soft stepping */ + SOC_ENUM("ADC Soft Stepping", adc_softstepping_enum), + /* Left/Right Input attenuation */ + SOC_SINGLE_TLV("Left Input IN_1L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_2L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_3L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_1R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_2L_3L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 6, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_1L_1R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_2R_3R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_1R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_2R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_3R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_1L Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_2R_3R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 6, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_1L_1R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_2L_3L Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 2, 1, 1, input_attenuation_tlv), + SOC_DOUBLE_R_S_TLV("ADC Volume Control Capture Volume", ADC3XXX_LADC_VOL, + ADC3XXX_RADC_VOL, 0, -24, 40, 6, 0, adc_tlv), + /* Empirically, the following doesn't work the way it's supposed + * to. Values 0, -0.1, -0.2 and -0.3 dB result in the same level, and + * -0.4 dB drops about 0.12 dB on a specific chip. + */ + SOC_DOUBLE_TLV("ADC Fine Volume Control Capture Volume", ADC3XXX_ADC_FGA, + 4, 0, 4, 1, adc_fine_tlv), + SOC_SINGLE("Left ADC Unselected CM Bias Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 6, 1, 0), + SOC_SINGLE("Right ADC Unselected CM Bias Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 6, 1, 0), + SOC_ENUM("Dither Control DC Offset", dither_dc_offset_enum), +}; + +/* Left input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Right input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1L Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Left Digital Mic input for left ADC */ +static const struct snd_kcontrol_new left_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Left ADC Capture Switch", + ADC3XXX_ADC_DIGITAL, 3, 0x1, 0), +}; + +/* Right Digital Mic input for Right ADC */ +static const struct snd_kcontrol_new right_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Right ADC Capture Switch", + ADC3XXX_ADC_DIGITAL, 2, 0x1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget adc3xxx_dapm_widgets[] = { + + /* Left Input Selection */ + SND_SOC_DAPM_MIXER("Left Input", SND_SOC_NOPM, 0, 0, + &left_input_mixer_controls[0], + ARRAY_SIZE(left_input_mixer_controls)), + /* Right Input Selection */ + SND_SOC_DAPM_MIXER("Right Input", SND_SOC_NOPM, 0, 0, + &right_input_mixer_controls[0], + ARRAY_SIZE(right_input_mixer_controls)), + /* PGA selection */ + SND_SOC_DAPM_PGA("Left PGA", ADC3XXX_LEFT_APGA_CTRL, 7, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right PGA", ADC3XXX_RIGHT_APGA_CTRL, 7, 1, NULL, 0), + + /* Digital Microphone Input Control for Left/Right ADC */ + SND_SOC_DAPM_MIXER("Left DMic Input", SND_SOC_NOPM, 0, 0, + &left_input_dmic_controls[0], + ARRAY_SIZE(left_input_dmic_controls)), + SND_SOC_DAPM_MIXER("Right DMic Input", SND_SOC_NOPM, 0, 0, + &right_input_dmic_controls[0], + ARRAY_SIZE(right_input_dmic_controls)), + + /* Left/Right ADC */ + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ADC3XXX_ADC_DIGITAL, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ADC3XXX_ADC_DIGITAL, 6, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("IN_1L"), + SND_SOC_DAPM_INPUT("IN_1R"), + SND_SOC_DAPM_INPUT("IN_2L"), + SND_SOC_DAPM_INPUT("IN_2R"), + SND_SOC_DAPM_INPUT("IN_3L"), + SND_SOC_DAPM_INPUT("IN_3R"), + SND_SOC_DAPM_INPUT("DIFL_1L_1R"), + SND_SOC_DAPM_INPUT("DIFL_2L_3L"), + SND_SOC_DAPM_INPUT("DIFL_2R_3R"), + SND_SOC_DAPM_INPUT("DIFR_1L_1R"), + SND_SOC_DAPM_INPUT("DIFR_2L_3L"), + SND_SOC_DAPM_INPUT("DIFR_2R_3R"), + SND_SOC_DAPM_INPUT("DMic_L"), + SND_SOC_DAPM_INPUT("DMic_R"), + + /* Digital audio interface output */ + SND_SOC_DAPM_AIF_OUT("AIF_OUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Clocks */ + SND_SOC_DAPM_SUPPLY("PLL_CLK", ADC3XXX_PLL_PROG_PR, ADC3XXX_ENABLE_PLL_SHIFT, + 0, adc3xxx_pll_delay, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_SUPPLY("ADC_CLK", ADC3XXX_ADC_NADC, ADC3XXX_ENABLE_NADC_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_MOD_CLK", ADC3XXX_ADC_MADC, ADC3XXX_ENABLE_MADC_SHIFT, + 0, NULL, 0), + + /* This refers to the generated BCLK in master mode. */ + SND_SOC_DAPM_SUPPLY("BCLK", ADC3XXX_BCLK_N_DIV, ADC3XXX_ENABLE_BCLK_SHIFT, + 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route adc3xxx_intercon[] = { + /* Left input selection from switches */ + { "Left Input", "IN_1L Capture Switch", "IN_1L" }, + { "Left Input", "IN_2L Capture Switch", "IN_2L" }, + { "Left Input", "IN_3L Capture Switch", "IN_3L" }, + { "Left Input", "DIF_2L_3L Capture Switch", "DIFL_2L_3L" }, + { "Left Input", "DIF_1L_1R Capture Switch", "DIFL_1L_1R" }, + { "Left Input", "DIF_2R_3R Capture Switch", "DIFL_2R_3R" }, + { "Left Input", "IN_1R Capture Switch", "IN_1R" }, + + /* Left input selection to left PGA */ + { "Left PGA", NULL, "Left Input" }, + + /* Left PGA to left ADC */ + { "Left ADC", NULL, "Left PGA" }, + + /* Right input selection from switches */ + { "Right Input", "IN_1R Capture Switch", "IN_1R" }, + { "Right Input", "IN_2R Capture Switch", "IN_2R" }, + { "Right Input", "IN_3R Capture Switch", "IN_3R" }, + { "Right Input", "DIF_2R_3R Capture Switch", "DIFR_2R_3R" }, + { "Right Input", "DIF_1L_1R Capture Switch", "DIFR_1L_1R" }, + { "Right Input", "DIF_2L_3L Capture Switch", "DIFR_2L_3L" }, + { "Right Input", "IN_1L Capture Switch", "IN_1L" }, + + /* Right input selection to right PGA */ + { "Right PGA", NULL, "Right Input" }, + + /* Right PGA to right ADC */ + { "Right ADC", NULL, "Right PGA" }, + + /* Left DMic Input selection from switch */ + { "Left DMic Input", "Left ADC Capture Switch", "DMic_L" }, + + /* Left DMic to left ADC */ + { "Left ADC", NULL, "Left DMic Input" }, + + /* Right DMic Input selection from switch */ + { "Right DMic Input", "Right ADC Capture Switch", "DMic_R" }, + + /* Right DMic to right ADC */ + { "Right ADC", NULL, "Right DMic Input" }, + + /* ADC to AIF output */ + { "AIF_OUT", NULL, "Left ADC" }, + { "AIF_OUT", NULL, "Right ADC" }, + + /* Clocking */ + { "ADC_MOD_CLK", NULL, "ADC_CLK" }, + { "Left ADC", NULL, "ADC_MOD_CLK" }, + { "Right ADC", NULL, "ADC_MOD_CLK" }, + + { "BCLK", NULL, "ADC_CLK" }, +}; + +static const struct snd_soc_dapm_route adc3xxx_pll_intercon[] = { + { "ADC_CLK", NULL, "PLL_CLK" }, +}; + +static const struct snd_soc_dapm_route adc3xxx_bclk_out_intercon[] = { + { "AIF_OUT", NULL, "BCLK" } +}; + +#ifdef CONFIG_GPIOLIB +static int adc3xxx_gpio_request(struct gpio_chip *chip, unsigned int offset) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + + if (offset >= ADC3XXX_GPIOS_MAX) + return -EINVAL; + + /* GPIO1 is offset 0, GPIO2 is offset 1 */ + /* We check here that the GPIO pins are either not configured in the + * DT, or that they purposely are set as outputs. + * (Input mode not yet implemented). + */ + if (adc3xxx->gpio_cfg[offset] != 0 && + adc3xxx->gpio_cfg[offset] != ADC3XXX_GPIO_GPO + 1) + return -EINVAL; + + return 0; +} + +static int adc3xxx_gpio_direction_out(struct gpio_chip *chip, + unsigned int offset, int value) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + + /* Set GPIO output function. */ + return regmap_update_bits(adc3xxx->regmap, + adc3xxx_gpio_ctrl_reg[offset], + ADC3XXX_GPIO_CTRL_CFG_MASK | + ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK, + ADC3XXX_GPIO_GPO << ADC3XXX_GPIO_CTRL_CFG_SHIFT | + !!value << ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_SHIFT); +} + +/* With only GPIO outputs configured, we never get the .direction_out call, + * so we set the output mode and output value in the same call. Hence + * .set in practice does the same thing as .direction_out . + */ +static void adc3xxx_gpio_set(struct gpio_chip *chip, unsigned int offset, + int value) +{ + (void) adc3xxx_gpio_direction_out(chip, offset, value); +} + +/* Even though we only support GPIO output for now, some GPIO clients + * want to read the current pin state using the .get callback. + */ +static int adc3xxx_gpio_get(struct gpio_chip *chip, unsigned int offset) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + unsigned int regval; + int ret; + + /* We only allow output pins, so just read the value set in the output + * pin register field. + */ + ret = regmap_read(adc3xxx->regmap, adc3xxx_gpio_ctrl_reg[offset], &regval); + if (ret) + return ret; + return !!(regval & ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK); +} + +static const struct gpio_chip adc3xxx_gpio_chip = { + .label = "adc3xxx", + .owner = THIS_MODULE, + .request = adc3xxx_gpio_request, + .direction_output = adc3xxx_gpio_direction_out, + .set = adc3xxx_gpio_set, + .get = adc3xxx_gpio_get, + .can_sleep = 1, +}; + +static void adc3xxx_free_gpio(struct adc3xxx *adc3xxx) +{ + gpiochip_remove(&adc3xxx->gpio_chip); +} + +#endif + +static void adc3xxx_init_gpio(struct adc3xxx *adc3xxx) +{ + int gpio, micbias; +#ifdef CONFIG_GPIOLIB + int ret; + + adc3xxx->gpio_chip = adc3xxx_gpio_chip; + adc3xxx->gpio_chip.ngpio = ADC3XXX_GPIOS_MAX; + adc3xxx->gpio_chip.parent = adc3xxx->dev; + adc3xxx->gpio_chip.base = -1; + + ret = gpiochip_add_data(&adc3xxx->gpio_chip, adc3xxx); + if (ret) + dev_err(adc3xxx->dev, "Failed to add gpios: %d\n", ret); +#endif + + /* Set up potential GPIO configuration from the devicetree. + * This allows us to set up things which are not software + * controllable GPIOs, such as PDM microphone I/O, + */ + for (gpio = 0; gpio < ADC3XXX_GPIOS_MAX; gpio++) { + unsigned int cfg = adc3xxx->gpio_cfg[gpio]; + + if (cfg) { + cfg--; /* actual value to use is stored +1 */ + regmap_update_bits(adc3xxx->regmap, + adc3xxx_gpio_ctrl_reg[gpio], + ADC3XXX_GPIO_CTRL_CFG_MASK, + cfg << ADC3XXX_GPIO_CTRL_CFG_SHIFT); + } + } + + /* Set up micbias voltage */ + for (micbias = 0; micbias < ADC3XXX_MICBIAS_PINS; micbias++) { + unsigned int vg = adc3xxx->micbias_vg[micbias]; + + regmap_update_bits(adc3xxx->regmap, + ADC3XXX_MICBIAS_CTRL, + ADC3XXX_MICBIAS_MASK << adc3xxx_micbias_shift[micbias], + vg << adc3xxx_micbias_shift[micbias]); + } +} + +static int adc3xxx_parse_dt_gpio(struct adc3xxx *adc3xxx, + const char *propname, unsigned int *cfg) +{ + struct device *dev = adc3xxx->dev; + struct device_node *np = dev->of_node; + unsigned int val; + + if (!of_property_read_u32(np, propname, &val)) { + if (val & ~15 || val == 7 || val >= 11) { + dev_err(dev, "Invalid property value for '%s'\n", propname); + return -EINVAL; + } + if (val == ADC3XXX_GPIO_GPI) + dev_warn(dev, "GPIO Input read not yet implemented\n"); + *cfg = val + 1; /* 0 => not set up, all others shifted +1 */ + } + return 0; +} + +static int adc3xxx_parse_dt_micbias(struct adc3xxx *adc3xxx, + const char *propname, unsigned int *vg) +{ + struct device *dev = adc3xxx->dev; + struct device_node *np = dev->of_node; + unsigned int val; + + if (!of_property_read_u32(np, propname, &val)) { + if (val >= ADC3XXX_MICBIAS_AVDD) { + dev_err(dev, "Invalid property value for '%s'\n", propname); + return -EINVAL; + } + *vg = val; + } + return 0; +} + +static int adc3xxx_parse_pll_mode(uint32_t val, unsigned int *pll_mode) +{ + if (val != ADC3XXX_PLL_ENABLE && val != ADC3XXX_PLL_BYPASS && + val != ADC3XXX_PLL_AUTO) + return -EINVAL; + + *pll_mode = val; + + return 0; +} + +static void adc3xxx_setup_pll(struct snd_soc_component *component, + int div_entry) +{ + int i = div_entry; + + /* P & R values */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_PR, + (adc3xxx_divs[i].pll_p << ADC3XXX_PLLP_SHIFT) | + (adc3xxx_divs[i].pll_r << ADC3XXX_PLLR_SHIFT)); + /* J value */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_J, + adc3xxx_divs[i].pll_j & ADC3XXX_PLLJ_MASK); + /* D value */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_LSB, + adc3xxx_divs[i].pll_d & ADC3XXX_PLLD_LSB_MASK); + snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_MSB, + (adc3xxx_divs[i].pll_d >> 8) & ADC3XXX_PLLD_MSB_MASK); +} + +static int adc3xxx_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 snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(dai->component); + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + int i, width = 16; + u8 iface_len, bdiv; + + i = adc3xxx_get_divs(component->dev, adc3xxx->sysclk, + params_rate(params), adc3xxx->pll_mode); + + if (i < 0) + return i; + + /* select data word length */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface_len = ADC3XXX_IFACE_16BITS; + width = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_len = ADC3XXX_IFACE_20BITS; + width = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_len = ADC3XXX_IFACE_24BITS; + width = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_len = ADC3XXX_IFACE_32BITS; + width = 32; + break; + default: + dev_err(component->dev, "Unsupported serial data format\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, ADC3XXX_INTERFACE_CTRL_1, + ADC3XXX_WLENGTH_MASK, iface_len); + if (adc3xxx_divs[i].pll_p) { /* If PLL used for this mode */ + adc3xxx_setup_pll(component, i); + snd_soc_component_write(component, ADC3XXX_CLKGEN_MUX, ADC3XXX_USE_PLL); + if (!adc3xxx->use_pll) { + snd_soc_dapm_add_routes(dapm, adc3xxx_pll_intercon, + ARRAY_SIZE(adc3xxx_pll_intercon)); + adc3xxx->use_pll = 1; + } + } else { + snd_soc_component_write(component, ADC3XXX_CLKGEN_MUX, ADC3XXX_NO_PLL); + if (adc3xxx->use_pll) { + snd_soc_dapm_del_routes(dapm, adc3xxx_pll_intercon, + ARRAY_SIZE(adc3xxx_pll_intercon)); + adc3xxx->use_pll = 0; + } + } + + /* NADC */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_NADC, + ADC3XXX_NADC_MASK, adc3xxx_divs[i].nadc); + /* MADC */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_MADC, + ADC3XXX_MADC_MASK, adc3xxx_divs[i].madc); + /* AOSR */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_AOSR, + ADC3XXX_AOSR_MASK, adc3xxx_divs[i].aosr); + /* BDIV N Value */ + /* BCLK is (by default) set up to be derived from ADC_CLK */ + bdiv = (adc3xxx_divs[i].aosr * adc3xxx_divs[i].madc) / (2 * width); + snd_soc_component_update_bits(component, ADC3XXX_BCLK_N_DIV, + ADC3XXX_BDIV_MASK, bdiv); + + return 0; +} + +static const char *adc3xxx_pll_mode_text(int pll_mode) +{ + switch (pll_mode) { + case ADC3XXX_PLL_AUTO: + return "PLL auto"; + case ADC3XXX_PLL_ENABLE: + return "PLL enable"; + case ADC3XXX_PLL_BYPASS: + return "PLL bypass"; + default: + break; + } + + return "PLL unknown"; +} + +static int adc3xxx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + int ret; + + ret = adc3xxx_parse_pll_mode(clk_id, &adc3xxx->pll_mode); + if (ret < 0) + return ret; + + adc3xxx->sysclk = freq; + dev_dbg(component->dev, "Set sysclk to %u Hz, %s\n", + freq, adc3xxx_pll_mode_text(adc3xxx->pll_mode)); + return 0; +} + +static int adc3xxx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + u8 clkdir = 0, format = 0; + int master = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + master = 1; + clkdir = ADC3XXX_BCLK_MASTER | ADC3XXX_WCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBC_CFC: + master = 0; + break; + default: + dev_err(component->dev, "Invalid DAI clock setup\n"); + return -EINVAL; + } + + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF: + format = ADC3XXX_FORMAT_DSP; + break; + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF: + format = ADC3XXX_FORMAT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_RJF; + break; + case SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_LJF; + break; + default: + dev_err(component->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* Add/del route enabling BCLK output as applicable */ + if (master && !adc3xxx->master) + snd_soc_dapm_add_routes(dapm, adc3xxx_bclk_out_intercon, + ARRAY_SIZE(adc3xxx_bclk_out_intercon)); + else if (!master && adc3xxx->master) + snd_soc_dapm_del_routes(dapm, adc3xxx_bclk_out_intercon, + ARRAY_SIZE(adc3xxx_bclk_out_intercon)); + adc3xxx->master = master; + + /* set clock direction and format */ + return snd_soc_component_update_bits(component, + ADC3XXX_INTERFACE_CTRL_1, + ADC3XXX_CLKDIR_MASK | ADC3XXX_FORMAT_MASK, + clkdir | format); +} + +static struct snd_soc_dai_ops adc3xxx_dai_ops = { + .hw_params = adc3xxx_hw_params, + .set_sysclk = adc3xxx_set_dai_sysclk, + .set_fmt = adc3xxx_set_dai_fmt, +}; + +struct snd_soc_dai_driver adc3xxx_dai = { + .name = "tlv320adc3xxx-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ADC3XXX_RATES, + .formats = ADC3XXX_FORMATS, + }, + .ops = &adc3xxx_dai_ops, +}; + +static const struct snd_soc_component_driver soc_component_dev_adc3xxx = { + .controls = adc3xxx_snd_controls, + .num_controls = ARRAY_SIZE(adc3xxx_snd_controls), + .dapm_widgets = adc3xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adc3xxx_dapm_widgets), + .dapm_routes = adc3xxx_intercon, + .num_dapm_routes = ARRAY_SIZE(adc3xxx_intercon), +}; + +static int adc3xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; + struct adc3xxx *adc3xxx = NULL; + enum of_gpio_flags flags; + int ret; + + adc3xxx = devm_kzalloc(dev, sizeof(struct adc3xxx), GFP_KERNEL); + if (!adc3xxx) + return -ENOMEM; + adc3xxx->dev = dev; + + adc3xxx->mclk = devm_clk_get(dev, NULL); + if (IS_ERR(adc3xxx->mclk)) { + /* + * The chip itself supports running off the BCLK either + * directly or via the PLL, but the driver does not (yet), so + * having a specified mclk is required. Otherwise, we could + * use the lack of a clocks property to indicate when BCLK is + * intended as the clock source. + */ + return PTR_ERR(adc3xxx->mclk); + } else if (adc3xxx->mclk) { + ret = clk_prepare_enable(adc3xxx->mclk); + if (ret < 0) + return ret; + dev_dbg(dev, "Enabled MCLK, freq %lu Hz\n", clk_get_rate(adc3xxx->mclk)); + } + + adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmdin-gpio1", &adc3xxx->gpio_cfg[0]); + adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmclk-gpio2", &adc3xxx->gpio_cfg[1]); + adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias1-vg", &adc3xxx->micbias_vg[0]); + adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias2-vg", &adc3xxx->micbias_vg[1]); + + adc3xxx->regmap = devm_regmap_init_i2c(i2c, &adc3xxx_regmap); + if (IS_ERR(adc3xxx->regmap)) { + ret = PTR_ERR(adc3xxx->regmap); + return ret; + } + + i2c_set_clientdata(i2c, adc3xxx); + + adc3xxx->rst_pin = of_get_named_gpio_flags(np, "reset-gpios", 0, &flags); + if (adc3xxx->rst_pin < 0 || !gpio_is_valid(adc3xxx->rst_pin)) + return -ENXIO; + + adc3xxx->rst_active = !(flags & OF_GPIO_ACTIVE_LOW); + adc3xxx->type = id->driver_data; + + ret = devm_gpio_request(dev, adc3xxx->rst_pin, "adc3xxx reset"); + if (ret < 0) { + dev_err(dev, "Failed to request rst_pin: %d\n", ret); + return ret; + } + + /* Reset codec chip */ + gpio_direction_output(adc3xxx->rst_pin, adc3xxx->rst_active); + usleep_range(2000, 100000); /* Requirement: > 10 ns (datasheet p13) */ + gpio_direction_output(adc3xxx->rst_pin, !adc3xxx->rst_active); + + /* Potentially set up pins used as GPIOs */ + adc3xxx_init_gpio(adc3xxx); + + ret = snd_soc_register_component(dev, + &soc_component_dev_adc3xxx, &adc3xxx_dai, 1); + if (ret < 0) + dev_err(dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int __exit adc3xxx_i2c_remove(struct i2c_client *client) +{ + struct adc3xxx *adc3xxx = i2c_get_clientdata(client); + + if (adc3xxx->mclk) + clk_disable_unprepare(adc3xxx->mclk); +#ifdef CONFIG_GPIOLIB + adc3xxx_free_gpio(adc3xxx); +#endif + snd_soc_unregister_component(&client->dev); + return 0; +} + +static const struct of_device_id tlv320adc3xxx_of_match[] = { + { .compatible = "ti,tlv320adc3001", }, + { .compatible = "ti,tlv320adc3101", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320adc3xxx_of_match); + +static const struct i2c_device_id adc3xxx_i2c_id[] = { + { "tlv320adc3001", ADC3001 }, + { "tlv320adc3101", ADC3101 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, adc3xxx_i2c_id); + +static struct i2c_driver adc3xxx_i2c_driver = { + .driver = { + .name = "tlv320adc3xxx-codec", + .of_match_table = tlv320adc3xxx_of_match, + }, + .probe = adc3xxx_i2c_probe, + .remove = adc3xxx_i2c_remove, + .id_table = adc3xxx_i2c_id, +}; + +module_i2c_driver(adc3xxx_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320ADC3xxx codec driver"); +MODULE_AUTHOR(" shahina.s(a)mistralsolutions.com "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/tlv320adc3xxx.h b/sound/soc/codecs/tlv320adc3xxx.h new file mode 100644 index 000000000000..82f82c3e7bc5 --- /dev/null +++ b/sound/soc/codecs/tlv320adc3xxx.h @@ -0,0 +1,379 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov + * + * History: + * + * Author: "Shahina Shaik" < shahina.s(a)mistralsolutions.com > + * Copyright: (C) 2010 Mistral Solutions Pvt Ltd. + * + * Author: Dongge wu <dgwu(a)ambarella.com> + * 2015/10/28 - [Dongge wu] Created file + * Copyright (C) 2014-2018, Ambarella, Inc. + * + * Author: Ricard Wanderlof <ricardw(a)axis.com> + * 2020/11/05: Fixing driver for Linux 4.14: more clocking modes, etc. + * 2021/09/03: Porting to Linux 5.4, with enhancements. + * Copyright (C) 2021 Axis Communications AB + * + */ + +#ifndef _ADC3XXX_H +#define _ADC3XXX_H + +/* 8 bit mask value */ +#define ADC3XXX_8BITS_MASK 0xFF + +#define ADC3XXX_PAGE_SIZE 128 +#define ADC3XXX_REG(page, reg) ((page * ADC3XXX_PAGE_SIZE) + reg) + +#define ADC3XXX_MICBIAS_PINS 2 + +/* Number of GPIO pins exposed via the gpiolib interface */ +#define ADC3XXX_GPIOS_MAX 2 + +/* + * PLL modes, to be used for clk_id for set_sysclk callback. + * + * The default behavior (AUTO) is to take the first matching entry in the clock + * table, which is intended to be the PLL based one if there is more than one. + * + * Setting the clock source using simple-card (clocks or + * system-clock-frequency property) sets clk_id = 0 = ADC3XXX_PLL_AUTO. + */ +#define ADC3XXX_PLL_AUTO 0 /* Use first available mode */ +#define ADC3XXX_PLL_ENABLE 1 /* Use PLL for clock generation */ +#define ADC3XXX_PLL_BYPASS 2 /* Don't use PLL for clock generation */ + + +/****************************************************************************/ +/* Page 0 Registers */ +/****************************************************************************/ + +/* Page select register */ +#define ADC3XXX_PAGE_SELECT ADC3XXX_REG(0, 0) +/* Software reset register */ +#define ADC3XXX_RESET ADC3XXX_REG(0, 1) + +/* 2-3 Reserved */ + +/* PLL programming register B */ +#define ADC3XXX_CLKGEN_MUX ADC3XXX_REG(0, 4) +/* PLL P and R-Val */ +#define ADC3XXX_PLL_PROG_PR ADC3XXX_REG(0, 5) +/* PLL J-Val */ +#define ADC3XXX_PLL_PROG_J ADC3XXX_REG(0, 6) +/* PLL D-Val MSB */ +#define ADC3XXX_PLL_PROG_D_MSB ADC3XXX_REG(0, 7) +/* PLL D-Val LSB */ +#define ADC3XXX_PLL_PROG_D_LSB ADC3XXX_REG(0, 8) + +/* 9-17 Reserved */ + +/* ADC NADC */ +#define ADC3XXX_ADC_NADC ADC3XXX_REG(0, 18) +/* ADC MADC */ +#define ADC3XXX_ADC_MADC ADC3XXX_REG(0, 19) +/* ADC AOSR */ +#define ADC3XXX_ADC_AOSR ADC3XXX_REG(0, 20) +/* ADC IADC */ +#define ADC3XXX_ADC_IADC ADC3XXX_REG(0, 21) + +/* 23-24 Reserved */ + +/* CLKOUT MUX */ +#define ADC3XXX_CLKOUT_MUX ADC3XXX_REG(0, 25) +/* CLOCKOUT M divider value */ +#define ADC3XXX_CLKOUT_M_DIV ADC3XXX_REG(0, 26) +/*Audio Interface Setting Register 1*/ +#define ADC3XXX_INTERFACE_CTRL_1 ADC3XXX_REG(0, 27) +/* Data Slot Offset (Ch_Offset_1) */ +#define ADC3XXX_CH_OFFSET_1 ADC3XXX_REG(0, 28) +/* ADC interface control 2 */ +#define ADC3XXX_INTERFACE_CTRL_2 ADC3XXX_REG(0, 29) +/* BCLK N Divider */ +#define ADC3XXX_BCLK_N_DIV ADC3XXX_REG(0, 30) +/* Secondary audio interface control 1 */ +#define ADC3XXX_INTERFACE_CTRL_3 ADC3XXX_REG(0, 31) +/* Secondary audio interface control 2 */ +#define ADC3XXX_INTERFACE_CTRL_4 ADC3XXX_REG(0, 32) +/* Secondary audio interface control 3 */ +#define ADC3XXX_INTERFACE_CTRL_5 ADC3XXX_REG(0, 33) +/* I2S sync */ +#define ADC3XXX_I2S_SYNC ADC3XXX_REG(0, 34) + +/* 35 Reserved */ + +/* ADC flag register */ +#define ADC3XXX_ADC_FLAG ADC3XXX_REG(0, 36) +/* Data slot offset 2 (Ch_Offset_2) */ +#define ADC3XXX_CH_OFFSET_2 ADC3XXX_REG(0, 37) +/* I2S TDM control register */ +#define ADC3XXX_I2S_TDM_CTRL ADC3XXX_REG(0, 38) + +/* 39-41 Reserved */ + +/* Interrupt flags (overflow) */ +#define ADC3XXX_INTR_FLAG_1 ADC3XXX_REG(0, 42) +/* Interrupt flags (overflow) */ +#define ADC3XXX_INTR_FLAG_2 ADC3XXX_REG(0, 43) + +/* 44 Reserved */ + +/* Interrupt flags ADC */ +#define ADC3XXX_INTR_FLAG_ADC1 ADC3XXX_REG(0, 45) + +/* 46 Reserved */ + +/* Interrupt flags ADC */ +#define ADC3XXX_INTR_FLAG_ADC2 ADC3XXX_REG(0, 47) +/* INT1 interrupt control */ +#define ADC3XXX_INT1_CTRL ADC3XXX_REG(0, 48) +/* INT2 interrupt control */ +#define ADC3XXX_INT2_CTRL ADC3XXX_REG(0, 49) + +/* 50 Reserved */ + +/* DMCLK/GPIO2 control */ +#define ADC3XXX_GPIO2_CTRL ADC3XXX_REG(0, 51) +/* DMDIN/GPIO1 control */ +#define ADC3XXX_GPIO1_CTRL ADC3XXX_REG(0, 52) +/* DOUT Control */ +#define ADC3XXX_DOUT_CTRL ADC3XXX_REG(0, 53) + +/* 54-56 Reserved */ + +/* ADC sync control 1 */ +#define ADC3XXX_SYNC_CTRL_1 ADC3XXX_REG(0, 57) +/* ADC sync control 2 */ +#define ADC3XXX_SYNC_CTRL_2 ADC3XXX_REG(0, 58) +/* ADC CIC filter gain control */ +#define ADC3XXX_CIC_GAIN_CTRL ADC3XXX_REG(0, 59) + +/* 60 Reserved */ + +/* ADC processing block selection */ +#define ADC3XXX_PRB_SELECT ADC3XXX_REG(0, 61) +/* Programmable instruction mode control bits */ +#define ADC3XXX_INST_MODE_CTRL ADC3XXX_REG(0, 62) + +/* 63-79 Reserved */ + +/* Digital microphone polarity control */ +#define ADC3XXX_MIC_POLARITY_CTRL ADC3XXX_REG(0, 80) +/* ADC Digital */ +#define ADC3XXX_ADC_DIGITAL ADC3XXX_REG(0, 81) +/* ADC Fine Gain Adjust */ +#define ADC3XXX_ADC_FGA ADC3XXX_REG(0, 82) +/* Left ADC Channel Volume Control */ +#define ADC3XXX_LADC_VOL ADC3XXX_REG(0, 83) +/* Right ADC Channel Volume Control */ +#define ADC3XXX_RADC_VOL ADC3XXX_REG(0, 84) +/* ADC phase compensation */ +#define ADC3XXX_ADC_PHASE_COMP ADC3XXX_REG(0, 85) +/* Left Channel AGC Control Register 1 */ +#define ADC3XXX_LEFT_CHN_AGC_1 ADC3XXX_REG(0, 86) +/* Left Channel AGC Control Register 2 */ +#define ADC3XXX_LEFT_CHN_AGC_2 ADC3XXX_REG(0, 87) +/* Left Channel AGC Control Register 3 */ +#define ADC3XXX_LEFT_CHN_AGC_3 ADC3XXX_REG(0, 88) +/* Left Channel AGC Control Register 4 */ +#define ADC3XXX_LEFT_CHN_AGC_4 ADC3XXX_REG(0, 89) +/* Left Channel AGC Control Register 5 */ +#define ADC3XXX_LEFT_CHN_AGC_5 ADC3XXX_REG(0, 90) +/* Left Channel AGC Control Register 6 */ +#define ADC3XXX_LEFT_CHN_AGC_6 ADC3XXX_REG(0, 91) +/* Left Channel AGC Control Register 7 */ +#define ADC3XXX_LEFT_CHN_AGC_7 ADC3XXX_REG(0, 92) +/* Left AGC gain */ +#define ADC3XXX_LEFT_AGC_GAIN ADC3XXX_REG(0, 93) +/* Right Channel AGC Control Register 1 */ +#define ADC3XXX_RIGHT_CHN_AGC_1 ADC3XXX_REG(0, 94) +/* Right Channel AGC Control Register 2 */ +#define ADC3XXX_RIGHT_CHN_AGC_2 ADC3XXX_REG(0, 95) +/* Right Channel AGC Control Register 3 */ +#define ADC3XXX_RIGHT_CHN_AGC_3 ADC3XXX_REG(0, 96) +/* Right Channel AGC Control Register 4 */ +#define ADC3XXX_RIGHT_CHN_AGC_4 ADC3XXX_REG(0, 97) +/* Right Channel AGC Control Register 5 */ +#define ADC3XXX_RIGHT_CHN_AGC_5 ADC3XXX_REG(0, 98) +/* Right Channel AGC Control Register 6 */ +#define ADC3XXX_RIGHT_CHN_AGC_6 ADC3XXX_REG(0, 99) +/* Right Channel AGC Control Register 7 */ +#define ADC3XXX_RIGHT_CHN_AGC_7 ADC3XXX_REG(0, 100) +/* Right AGC gain */ +#define ADC3XXX_RIGHT_AGC_GAIN ADC3XXX_REG(0, 101) + +/* 102-127 Reserved */ + +/****************************************************************************/ +/* Page 1 Registers */ +/****************************************************************************/ +/* 1-25 Reserved */ + +/* Dither control */ +#define ADC3XXX_DITHER_CTRL ADC3XXX_REG(1, 26) + +/* 27-50 Reserved */ + +/* MICBIAS Configuration Register */ +#define ADC3XXX_MICBIAS_CTRL ADC3XXX_REG(1, 51) +/* Left ADC input selection for Left PGA */ +#define ADC3XXX_LEFT_PGA_SEL_1 ADC3XXX_REG(1, 52) + +/* 53 Reserved */ + +/* Right ADC input selection for Left PGA */ +#define ADC3XXX_LEFT_PGA_SEL_2 ADC3XXX_REG(1, 54) +/* Right ADC input selection for right PGA */ +#define ADC3XXX_RIGHT_PGA_SEL_1 ADC3XXX_REG(1, 55) + +/* 56 Reserved */ + +/* Right ADC input selection for right PGA */ +#define ADC3XXX_RIGHT_PGA_SEL_2 ADC3XXX_REG(1, 57) + +/* 58 Reserved */ + +/* Left analog PGA settings */ +#define ADC3XXX_LEFT_APGA_CTRL ADC3XXX_REG(1, 59) +/* Right analog PGA settings */ +#define ADC3XXX_RIGHT_APGA_CTRL ADC3XXX_REG(1, 60) +/* ADC Low current Modes */ +#define ADC3XXX_LOW_CURRENT_MODES ADC3XXX_REG(1, 61) +/* ADC analog PGA flags */ +#define ADC3XXX_ANALOG_PGA_FLAGS ADC3XXX_REG(1, 62) + +/* 63-127 Reserved */ + +/****************************************************************************/ +/* Macros and definitions */ +/****************************************************************************/ + +#define ADC3XXX_RATES SNDRV_PCM_RATE_8000_96000 +#define ADC3XXX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/****************************************************************************/ +/* ADC3XXX Register bits */ +/****************************************************************************/ +/* PLL Enable bits */ +#define ADC3XXX_ENABLE_PLL_SHIFT 7 +#define ADC3XXX_ENABLE_PLL (1 << ADC3XXX_ENABLE_PLL_SHIFT) +#define ADC3XXX_ENABLE_NADC_SHIFT 7 +#define ADC3XXX_ENABLE_NADC (1 << ADC3XXX_ENABLE_NADC_SHIFT) +#define ADC3XXX_ENABLE_MADC_SHIFT 7 +#define ADC3XXX_ENABLE_MADC (1 << ADC3XXX_ENABLE_MADC_SHIFT) +#define ADC3XXX_ENABLE_BCLK_SHIFT 7 +#define ADC3XXX_ENABLE_BCLK (1 << ADC3XXX_ENABLE_BCLK_SHIFT) + +/* Power bits */ +#define ADC3XXX_LADC_PWR_ON 0x80 +#define ADC3XXX_RADC_PWR_ON 0x40 + +#define ADC3XXX_SOFT_RESET 0x01 +#define ADC3XXX_BCLK_MASTER 0x08 +#define ADC3XXX_WCLK_MASTER 0x04 + +/* Interface register masks */ +#define ADC3XXX_FORMAT_MASK 0xc0 +#define ADC3XXX_FORMAT_SHIFT 6 +#define ADC3XXX_WLENGTH_MASK 0x30 +#define ADC3XXX_WLENGTH_SHIFT 4 +#define ADC3XXX_CLKDIR_MASK 0x0c +#define ADC3XXX_CLKDIR_SHIFT 2 + +/* Interface register bit patterns */ +#define ADC3XXX_FORMAT_I2S (0 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_DSP (1 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_RJF (2 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_LJF (3 << ADC3XXX_FORMAT_SHIFT) + +#define ADC3XXX_IFACE_16BITS (0 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_20BITS (1 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_24BITS (2 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_32BITS (3 << ADC3XXX_WLENGTH_SHIFT) + +/* PLL P/R bit offsets */ +#define ADC3XXX_PLLP_SHIFT 4 +#define ADC3XXX_PLLR_SHIFT 0 +#define ADC3XXX_PLL_PR_MASK 0x7f +#define ADC3XXX_PLLJ_MASK 0x3f +#define ADC3XXX_PLLD_MSB_MASK 0x3f +#define ADC3XXX_PLLD_LSB_MASK 0xff +#define ADC3XXX_NADC_MASK 0x7f +#define ADC3XXX_MADC_MASK 0x7f +#define ADC3XXX_AOSR_MASK 0xff +#define ADC3XXX_IADC_MASK 0xff +#define ADC3XXX_BDIV_MASK 0x7f + +/* PLL_CLKIN bits */ +#define ADC3XXX_PLL_CLKIN_SHIFT 2 +#define ADC3XXX_PLL_CLKIN_MCLK 0x0 +#define ADC3XXX_PLL_CLKIN_BCLK 0x1 +#define ADC3XXX_PLL_CLKIN_ZERO 0x3 + +/* CODEC_CLKIN bits */ +#define ADC3XXX_CODEC_CLKIN_SHIFT 0 +#define ADC3XXX_CODEC_CLKIN_MCLK 0x0 +#define ADC3XXX_CODEC_CLKIN_BCLK 0x1 +#define ADC3XXX_CODEC_CLKIN_PLL_CLK 0x3 + +#define ADC3XXX_USE_PLL ((ADC3XXX_PLL_CLKIN_MCLK << ADC3XXX_PLL_CLKIN_SHIFT) | \ + (ADC3XXX_CODEC_CLKIN_PLL_CLK << ADC3XXX_CODEC_CLKIN_SHIFT)) +#define ADC3XXX_NO_PLL ((ADC3XXX_PLL_CLKIN_ZERO << ADC3XXX_PLL_CLKIN_SHIFT) | \ + (ADC3XXX_CODEC_CLKIN_MCLK << ADC3XXX_CODEC_CLKIN_SHIFT)) + +/* Analog PGA control bits */ +#define ADC3XXX_LPGA_MUTE 0x80 +#define ADC3XXX_RPGA_MUTE 0x80 + +#define ADC3XXX_LPGA_GAIN_MASK 0x7f +#define ADC3XXX_RPGA_GAIN_MASK 0x7f + +/* ADC current modes */ +#define ADC3XXX_ADC_LOW_CURR_MODE 0x01 + +/* Left ADC Input selection bits */ +#define ADC3XXX_LCH_SEL1_SHIFT 0 +#define ADC3XXX_LCH_SEL2_SHIFT 2 +#define ADC3XXX_LCH_SEL3_SHIFT 4 +#define ADC3XXX_LCH_SEL4_SHIFT 6 + +#define ADC3XXX_LCH_SEL1X_SHIFT 0 +#define ADC3XXX_LCH_SEL2X_SHIFT 2 +#define ADC3XXX_LCH_SEL3X_SHIFT 4 +#define ADC3XXX_LCH_COMMON_MODE 0x40 +#define ADC3XXX_BYPASS_LPGA 0x80 + +/* Right ADC Input selection bits */ +#define ADC3XXX_RCH_SEL1_SHIFT 0 +#define ADC3XXX_RCH_SEL2_SHIFT 2 +#define ADC3XXX_RCH_SEL3_SHIFT 4 +#define ADC3XXX_RCH_SEL4_SHIFT 6 + +#define ADC3XXX_RCH_SEL1X_SHIFT 0 +#define ADC3XXX_RCH_SEL2X_SHIFT 2 +#define ADC3XXX_RCH_SEL3X_SHIFT 4 +#define ADC3XXX_RCH_COMMON_MODE 0x40 +#define ADC3XXX_BYPASS_RPGA 0x80 + +/* MICBIAS control bits */ +#define ADC3XXX_MICBIAS_MASK 0x2 +#define ADC3XXX_MICBIAS1_SHIFT 5 +#define ADC3XXX_MICBIAS2_SHIFT 3 + +#define ADC3XXX_ADC_MAX_VOLUME 64 +#define ADC3XXX_ADC_POS_VOL 24 + +/* GPIO control bits (GPIO1_CTRL and GPIO2_CTRL) */ +#define ADC3XXX_GPIO_CTRL_CFG_MASK 0x3c +#define ADC3XXX_GPIO_CTRL_CFG_SHIFT 2 +#define ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK 0x01 +#define ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_SHIFT 0 +#define ADC3XXX_GPIO_CTRL_INPUT_VALUE_MASK 0x02 +#define ADC3XXX_GPIO_CTRL_INPUT_VALUE_SHIFT 1 + +#endif /* _ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[PATCH] ALSA: hda/realtek: Add a quirk for Acer Spin SP513-54N
by Jaroslav Kysela 04 Nov '21

04 Nov '21

Another model requires ALC255_FIXUP_ACER_MIC_NO_PRESENCE fixup. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211853 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- 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 e6b266416393..3601749f2df6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8496,6 +8496,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x141f, "Acer Spin SP513-54N", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x142b, "Acer Swift SF314-42", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), -- 2.31.1

2 1

topology: Add option to pass pre-processor definitions
by GitHub pull_request - opened 04 Nov '21

04 Nov '21

alsa-project/alsa-utils pull request #117 was opened from ranj063: Add a -D switch to be able to pass pre-processor definitions that will be used to expand arguments in the input config file. This will be useful to generate multiple topology binaries from the same input config file with different argument values. For example: if we had a pipeline config as follows: Object.Pipeline { volume-playback.1 { dynamic_pipeline $DYNAMIC_PIPELINE } } We can define an argument for DYNAMIC_PIPELINE as: @args [ DYNAMIC_PIPELINE ] @args.DYNAMIC_PIPELINE { type integer default 0 } And when pre-processin the conf file pass "-D DYNAMIC_PIPELINE=1" to enable dynamic pipeline in the topology. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/117 Patch URL : https://github.com/alsa-project/alsa-utils/pull/117.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] Revert "ALSA: memalloc: Convert x86 SG-buffer handling with non-contiguous type"
by Takashi Iwai 04 Nov '21

04 Nov '21

This reverts commit 2d9ea39917a4e4293bc2caea902c7059a330b611. We've got a regression report showing that the audio got broken the device over AMD IOMMU. The conversion assumed the wrong pointer / page mapping for the indirect mapping case, and we need to correct this urgently, so let's revert it for now. Fixes: 2d9ea39917a4 ("ALSA: memalloc: Convert x86 SG-buffer handling with non-contiguous type") Reported-and-tested-by: Alex Xu (Hello71) <alex_y_xu(a)yahoo.ca> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 14 +-- sound/core/Makefile | 1 + sound/core/memalloc.c | 51 +--------- sound/core/sgbuf.c | 201 +++++++++++++++++++++++++++++++++++++++ 4 files changed, 213 insertions(+), 54 deletions(-) create mode 100644 sound/core/sgbuf.c diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 653dfffb3ac8..1051b84e8579 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -36,6 +36,13 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */ #define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */ #define SNDRV_DMA_TYPE_DEV_WC 5 /* continuous write-combined */ +#ifdef CONFIG_SND_DMA_SGBUF +#define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */ +#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ +#else +#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ +#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC +#endif #ifdef CONFIG_GENERIC_ALLOCATOR #define SNDRV_DMA_TYPE_DEV_IRAM 4 /* generic device iram-buffer */ #else @@ -44,13 +51,6 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_VMALLOC 7 /* vmalloc'ed buffer */ #define SNDRV_DMA_TYPE_NONCONTIG 8 /* non-coherent SG buffer */ #define SNDRV_DMA_TYPE_NONCOHERENT 9 /* non-coherent buffer */ -#ifdef CONFIG_SND_DMA_SGBUF -#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_NONCONTIG -#define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ -#else -#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ -#define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC -#endif /* * info for buffer allocation diff --git a/sound/core/Makefile b/sound/core/Makefile index 350d704ced98..79e1407cd0de 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -19,6 +19,7 @@ snd-$(CONFIG_SND_JACK) += ctljack.o jack.o snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \ pcm_memory.o memalloc.o snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o +snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 99cd0f67daa1..ea778f868cf3 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -560,50 +560,6 @@ static const struct snd_malloc_ops snd_dma_noncontig_ops = { .get_chunk_size = snd_dma_vmalloc_get_chunk_size, }; -/* x86-specific SG-buffer with WC pages */ -#ifdef CONFIG_SND_DMA_SGBUF -#define vmalloc_to_virt(v) (unsigned long)page_to_virt(vmalloc_to_page(v)) - -static void *snd_dma_sg_wc_alloc(struct snd_dma_buffer *dmab, size_t size) -{ - void *p = snd_dma_noncontig_alloc(dmab, size); - size_t ofs; - - if (!p) - return NULL; - for (ofs = 0; ofs < size; ofs += PAGE_SIZE) - set_memory_uc(vmalloc_to_virt(p + ofs), 1); - return p; -} - -static void snd_dma_sg_wc_free(struct snd_dma_buffer *dmab) -{ - size_t ofs; - - for (ofs = 0; ofs < dmab->bytes; ofs += PAGE_SIZE) - set_memory_wb(vmalloc_to_virt(dmab->area + ofs), 1); - snd_dma_noncontig_free(dmab); -} - -static int snd_dma_sg_wc_mmap(struct snd_dma_buffer *dmab, - struct vm_area_struct *area) -{ - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); - /* FIXME: dma_mmap_noncontiguous() works? */ - return -ENOENT; /* continue with the default mmap handler */ -} - -const struct snd_malloc_ops snd_dma_sg_wc_ops = { - .alloc = snd_dma_sg_wc_alloc, - .free = snd_dma_sg_wc_free, - .mmap = snd_dma_sg_wc_mmap, - .sync = snd_dma_noncontig_sync, - .get_addr = snd_dma_vmalloc_get_addr, - .get_page = snd_dma_vmalloc_get_page, - .get_chunk_size = snd_dma_vmalloc_get_chunk_size, -}; -#endif /* CONFIG_SND_DMA_SGBUF */ - /* * Non-coherent pages allocator */ @@ -663,13 +619,14 @@ static const struct snd_malloc_ops *dma_ops[] = { [SNDRV_DMA_TYPE_DEV_WC] = &snd_dma_wc_ops, [SNDRV_DMA_TYPE_NONCONTIG] = &snd_dma_noncontig_ops, [SNDRV_DMA_TYPE_NONCOHERENT] = &snd_dma_noncoherent_ops, -#ifdef CONFIG_SND_DMA_SGBUF - [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_wc_ops, -#endif #ifdef CONFIG_GENERIC_ALLOCATOR [SNDRV_DMA_TYPE_DEV_IRAM] = &snd_dma_iram_ops, #endif /* CONFIG_GENERIC_ALLOCATOR */ #endif /* CONFIG_HAS_DMA */ +#ifdef CONFIG_SND_DMA_SGBUF + [SNDRV_DMA_TYPE_DEV_SG] = &snd_dma_sg_ops, + [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_ops, +#endif }; static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab) diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c new file mode 100644 index 000000000000..8352a5cdb19f --- /dev/null +++ b/sound/core/sgbuf.c @@ -0,0 +1,201 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * Scatter-Gather buffer + * + * Copyright (c) by Takashi Iwai <tiwai(a)suse.de> + */ + +#include <linux/slab.h> +#include <linux/mm.h> +#include <linux/vmalloc.h> +#include <linux/export.h> +#include <sound/memalloc.h> +#include "memalloc_local.h" + +struct snd_sg_page { + void *buf; + dma_addr_t addr; +}; + +struct snd_sg_buf { + int size; /* allocated byte size */ + int pages; /* allocated pages */ + int tblsize; /* allocated table size */ + struct snd_sg_page *table; /* address table */ + struct page **page_table; /* page table (for vmap/vunmap) */ + struct device *dev; +}; + +/* table entries are align to 32 */ +#define SGBUF_TBL_ALIGN 32 +#define sgbuf_align_table(tbl) ALIGN((tbl), SGBUF_TBL_ALIGN) + +static void snd_dma_sg_free(struct snd_dma_buffer *dmab) +{ + struct snd_sg_buf *sgbuf = dmab->private_data; + struct snd_dma_buffer tmpb; + int i; + + if (!sgbuf) + return; + + vunmap(dmab->area); + dmab->area = NULL; + + tmpb.dev.type = SNDRV_DMA_TYPE_DEV; + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) + tmpb.dev.type = SNDRV_DMA_TYPE_DEV_WC; + tmpb.dev.dev = sgbuf->dev; + for (i = 0; i < sgbuf->pages; i++) { + if (!(sgbuf->table[i].addr & ~PAGE_MASK)) + continue; /* continuous pages */ + tmpb.area = sgbuf->table[i].buf; + tmpb.addr = sgbuf->table[i].addr & PAGE_MASK; + tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT; + snd_dma_free_pages(&tmpb); + } + + kfree(sgbuf->table); + kfree(sgbuf->page_table); + kfree(sgbuf); + dmab->private_data = NULL; +} + +#define MAX_ALLOC_PAGES 32 + +static void *snd_dma_sg_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + struct snd_sg_buf *sgbuf; + unsigned int i, pages, chunk, maxpages; + struct snd_dma_buffer tmpb; + struct snd_sg_page *table; + struct page **pgtable; + int type = SNDRV_DMA_TYPE_DEV; + pgprot_t prot = PAGE_KERNEL; + void *area; + + dmab->private_data = sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); + if (!sgbuf) + return NULL; + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) { + type = SNDRV_DMA_TYPE_DEV_WC; +#ifdef pgprot_noncached + prot = pgprot_noncached(PAGE_KERNEL); +#endif + } + sgbuf->dev = dmab->dev.dev; + pages = snd_sgbuf_aligned_pages(size); + sgbuf->tblsize = sgbuf_align_table(pages); + table = kcalloc(sgbuf->tblsize, sizeof(*table), GFP_KERNEL); + if (!table) + goto _failed; + sgbuf->table = table; + pgtable = kcalloc(sgbuf->tblsize, sizeof(*pgtable), GFP_KERNEL); + if (!pgtable) + goto _failed; + sgbuf->page_table = pgtable; + + /* allocate pages */ + maxpages = MAX_ALLOC_PAGES; + while (pages > 0) { + chunk = pages; + /* don't be too eager to take a huge chunk */ + if (chunk > maxpages) + chunk = maxpages; + chunk <<= PAGE_SHIFT; + if (snd_dma_alloc_pages_fallback(type, dmab->dev.dev, + chunk, &tmpb) < 0) { + if (!sgbuf->pages) + goto _failed; + size = sgbuf->pages * PAGE_SIZE; + break; + } + chunk = tmpb.bytes >> PAGE_SHIFT; + for (i = 0; i < chunk; i++) { + table->buf = tmpb.area; + table->addr = tmpb.addr; + if (!i) + table->addr |= chunk; /* mark head */ + table++; + *pgtable++ = virt_to_page(tmpb.area); + tmpb.area += PAGE_SIZE; + tmpb.addr += PAGE_SIZE; + } + sgbuf->pages += chunk; + pages -= chunk; + if (chunk < maxpages) + maxpages = chunk; + } + + sgbuf->size = size; + area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, prot); + if (!area) + goto _failed; + return area; + + _failed: + snd_dma_sg_free(dmab); /* free the table */ + return NULL; +} + +static dma_addr_t snd_dma_sg_get_addr(struct snd_dma_buffer *dmab, + size_t offset) +{ + struct snd_sg_buf *sgbuf = dmab->private_data; + dma_addr_t addr; + + addr = sgbuf->table[offset >> PAGE_SHIFT].addr; + addr &= ~((dma_addr_t)PAGE_SIZE - 1); + return addr + offset % PAGE_SIZE; +} + +static struct page *snd_dma_sg_get_page(struct snd_dma_buffer *dmab, + size_t offset) +{ + struct snd_sg_buf *sgbuf = dmab->private_data; + unsigned int idx = offset >> PAGE_SHIFT; + + if (idx >= (unsigned int)sgbuf->pages) + return NULL; + return sgbuf->page_table[idx]; +} + +static unsigned int snd_dma_sg_get_chunk_size(struct snd_dma_buffer *dmab, + unsigned int ofs, + unsigned int size) +{ + struct snd_sg_buf *sg = dmab->private_data; + unsigned int start, end, pg; + + start = ofs >> PAGE_SHIFT; + end = (ofs + size - 1) >> PAGE_SHIFT; + /* check page continuity */ + pg = sg->table[start].addr >> PAGE_SHIFT; + for (;;) { + start++; + if (start > end) + break; + pg++; + if ((sg->table[start].addr >> PAGE_SHIFT) != pg) + return (start << PAGE_SHIFT) - ofs; + } + /* ok, all on continuous pages */ + return size; +} + +static int snd_dma_sg_mmap(struct snd_dma_buffer *dmab, + struct vm_area_struct *area) +{ + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return -ENOENT; /* continue with the default mmap handler */ +} + +const struct snd_malloc_ops snd_dma_sg_ops = { + .alloc = snd_dma_sg_alloc, + .free = snd_dma_sg_free, + .get_addr = snd_dma_sg_get_addr, + .get_page = snd_dma_sg_get_page, + .get_chunk_size = snd_dma_sg_get_chunk_size, + .mmap = snd_dma_sg_mmap, +}; -- 2.26.2

1 0

[PATCH] ALSA: firewire-motu: add support for MOTU Traveler mk3
by Takashi Sakamoto 04 Nov '21

04 Nov '21

Mark of the Unicorn (MOTU) shipped Traveler mk3 as one of models in third generation of its FireWire series, and discontinued it already. The model consists of below ICs: * Texas Instruments TSB41AB2 * Phillips Semiconductors PDI1394L40 * Altera cyclone EP1C3 * Texas Instruments TMS320VC5402 It supports sampling transfer frequency up to 192.0 kHz. The packet format differs depending on both of current sampling transfer frequency and whether to enable ADAT channels in rx/tx packets. The model supports transmission of PCM frames as well as MIDI messages. The model supports command mechanism to configure internal DSP. Hardware meter information is available in the first 2 chunks of each data block of tx packet. This commit adds support for it. $ cd linux-firewire-tools/src $ python crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 0410af0a bus_info_length 4, crc_length 16, crc 44810 404 31333934 bus_name "1394" 408 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 40c 0001f200 company_id 0001f2 | 410 00090911 device_id 0000090911 | EUI-64 0001f20000090911 root directory ----------------------------------------------------------------- 414 0004ef04 directory_length 4, crc 61188 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 d1000002 --> unit directory at 428 424 8d000005 --> eui-64 leaf at 438 unit directory at 428 ----------------------------------------------------------------- 428 00031733 directory_length 3, crc 5939 42c 120001f2 specifier id 430 1300001b version 434 17102800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 00028484 leaf_length 2, crc 33924 43c 0001f200 company_id 0001f2 | 440 00090911 device_id 0000090911 | EUI-64 0001f20000090911 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v3.c | 22 +++++++++++++++++++--- sound/firewire/motu/motu.c | 1 + sound/firewire/motu/motu.h | 1 + 3 files changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 05608e8ca0bc..1371f38505d0 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -16,6 +16,7 @@ #define V3_CLOCK_SRC_INTERNAL 0x00 #define V3_CLOCK_SRC_WORD_ON_BNC 0x01 #define V3_CLOCK_SRC_SPH 0x02 +#define V3_CLOCK_SRC_AESEBU_ON_XLR 0x08 #define V3_CLOCK_SRC_SPDIF_ON_COAX 0x10 #define V3_CLOCK_SRC_OPT_IFACE_A 0x18 #define V3_CLOCK_SRC_OPT_IFACE_B 0x19 @@ -126,6 +127,9 @@ int snd_motu_protocol_v3_get_clock_source(struct snd_motu *motu, case V3_CLOCK_SRC_SPH: *src = SND_MOTU_CLOCK_SOURCE_SPH; break; + case V3_CLOCK_SRC_AESEBU_ON_XLR: + *src = SND_MOTU_CLOCK_SOURCE_AESEBU_ON_XLR; + break; case V3_CLOCK_SRC_SPDIF_ON_COAX: *src = SND_MOTU_CLOCK_SOURCE_SPDIF_ON_COAX; break; @@ -185,7 +189,7 @@ int snd_motu_protocol_v3_switch_fetching_mode(struct snd_motu *motu, sizeof(reg)); } -static int detect_packet_formats_828mk3(struct snd_motu *motu, u32 data) +static int detect_packet_formats_dual_opt_iface(struct snd_motu *motu, u32 data) { if (data & V3_ENABLE_OPT_IN_IFACE_A) { if (data & V3_NO_ADAT_OPT_IN_IFACE_A) { @@ -255,8 +259,10 @@ int snd_motu_protocol_v3_cache_packet_formats(struct snd_motu *motu) motu->spec->rx_fixed_pcm_chunks, sizeof(motu->rx_packet_formats.pcm_chunks)); - if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid) - return detect_packet_formats_828mk3(motu, data); + if (motu->spec == &snd_motu_spec_828mk3_fw || + motu->spec == &snd_motu_spec_828mk3_hybrid || + motu->spec == &snd_motu_spec_traveler_mk3) + return detect_packet_formats_dual_opt_iface(motu, data); else return 0; } @@ -281,6 +287,16 @@ const struct snd_motu_spec snd_motu_spec_828mk3_hybrid = { .rx_fixed_pcm_chunks = {14, 14, 14}, // Additional 4 dummy chunks at higher rate. }; +const struct snd_motu_spec snd_motu_spec_traveler_mk3 = { + .name = "TravelerMk3", + .protocol_version = SND_MOTU_PROTOCOL_V3, + .flags = SND_MOTU_SPEC_RX_MIDI_3RD_Q | + SND_MOTU_SPEC_TX_MIDI_3RD_Q | + SND_MOTU_SPEC_COMMAND_DSP, + .tx_fixed_pcm_chunks = {18, 14, 10}, + .rx_fixed_pcm_chunks = {14, 14, 10}, +}; + const struct snd_motu_spec snd_motu_spec_ultralite_mk3 = { .name = "UltraLiteMk3", .protocol_version = SND_MOTU_PROTOCOL_V3, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 5fc7ae475537..080728dd278f 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -169,6 +169,7 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3_fw), // FireWire only. SND_MOTU_DEV_ENTRY(0x000019, &snd_motu_spec_ultralite_mk3), // FireWire only. + SND_MOTU_DEV_ENTRY(0x00001b, &snd_motu_spec_traveler_mk3), SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3_hybrid), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index 79704ae6a73e..6ad1feab478b 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -138,6 +138,7 @@ extern const struct snd_motu_spec snd_motu_spec_8pre; extern const struct snd_motu_spec snd_motu_spec_828mk3_fw; extern const struct snd_motu_spec snd_motu_spec_828mk3_hybrid; +extern const struct snd_motu_spec snd_motu_spec_traveler_mk3; extern const struct snd_motu_spec snd_motu_spec_ultralite_mk3; extern const struct snd_motu_spec snd_motu_spec_audio_express; extern const struct snd_motu_spec snd_motu_spec_4pre; -- 2.32.0

2 1

[PATCH] ASoC: SOF:control: Fix variable type in snd_sof_refresh_control()
by Peter Ujfalusi 04 Nov '21

04 Nov '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> The second parameter for snd_sof_ipc_set_get_comp_data() is ipc_cmd, not ipc_ctrl_type and the type is u32. Fixes: 756bbe4205bc6 ("ASoC: SOF: Handle control change notification from firmware") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> --- sound/soc/sof/control.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 58bb89af4de1..bb1dfe4f6d40 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -69,7 +69,7 @@ static void snd_sof_refresh_control(struct snd_sof_control *scontrol) { struct sof_ipc_ctrl_data *cdata = scontrol->control_data; struct snd_soc_component *scomp = scontrol->scomp; - enum sof_ipc_ctrl_type ctrl_type; + u32 ipc_cmd; int ret; if (!scontrol->comp_data_dirty) @@ -79,9 +79,9 @@ static void snd_sof_refresh_control(struct snd_sof_control *scontrol) return; if (scontrol->cmd == SOF_CTRL_CMD_BINARY) - ctrl_type = SOF_IPC_COMP_GET_DATA; + ipc_cmd = SOF_IPC_COMP_GET_DATA; else - ctrl_type = SOF_IPC_COMP_GET_VALUE; + ipc_cmd = SOF_IPC_COMP_GET_VALUE; /* set the ABI header values */ cdata->data->magic = SOF_ABI_MAGIC; @@ -89,7 +89,7 @@ static void snd_sof_refresh_control(struct snd_sof_control *scontrol) /* refresh the component data from DSP */ scontrol->comp_data_dirty = false; - ret = snd_sof_ipc_set_get_comp_data(scontrol, ctrl_type, + ret = snd_sof_ipc_set_get_comp_data(scontrol, ipc_cmd, SOF_CTRL_TYPE_VALUE_CHAN_GET, scontrol->cmd, false); if (ret < 0) { -- 2.33.1

3 2

[PATCH v3 0/2] Codec driver for TLV320ADC3001 and 3101
by Ricard Wanderlof 04 Nov '21

04 Nov '21

This is a codec driver for the TLV320ADC3001 and TLV320ADC3101 2 channel audio ADC chips from Texas Instruments. Based on an old driver for these chips that I was referred to by Texas Instruments, but which apparently was never upstreamed, I've taken patches from various incarnations, ported them and augmented them with various new features in order to make a working driver. There is currently no support for the on-chip "miniDSP". Although the TLV320ADC3001 and TLV320ADC3101 have similarities with other TI codec chips for which there are existing drivers, upon closer inspection I decided the differences were too great to warrant attempting to modify an existing driver, especially without access to all supported chips and resources for testing them all. There are certainly things that could be improved, among others in the area of PLL register value calculations, but it has been tested on the Axis ARTPEC-8 platform and has been found to work satisfactorily at least with a TLV320ADC3101 operating as an I2S slave at 48 kHz with 32 bit bit depth, 1 or 2 channels, both with the on-chip PLL enabled and disabled. I have not tested this on a TLV320ADC3001 chip; in the original driver the only differences were related to the now-removed initialization code, so I've just retained the compatible strings for both chips. Limitations: - GPIO pins on chip can only be set up as outputs. - Using BCLK as an input to clock chain is supported by the hw but not by driver. Changes v2->v3: Fix a couple of issues reported by the kernel test robot (lkp(a)intel.com). Changes v1->v2: Large portions of the driver have been reworked from v1 based on remarks from Mark. More specifically: - Remove an unneeded #ifdef and EXPORT_SYMBOL_GPL. - Add ADC3XXX_ prefix to all macro names to avoid namespace contention. - Clean up certain macro names with mixed lower case - upper case names. - Do away with the D0..D7 macros representing bit values. - Fix general formatting: spaces before and after {, }, replace multi spaces with tabs, maximum 2 blank lines anywhere, and other minor issues. - Page register does not need to be volatile. - Rather than use 0 / -6dB enum for input attenuator, use standard TLV. - The dither offset control contains reserved values so use VALUE_ENUM instead of straight ENUM. - Replace vector of SOC_ENUMs with individual variables. - Fix ALSA control names: - Use Volume at the end of all gain controls, and remove "Gain" where applicable. - Use "Switch" at the end of all switch controls, either adding it or changing "switch" into "Switch" as applicable. - Switches and Volumes related to capture should be set up as such, so add "Capture" to widget names where appropriate. This is mostly important for the PGA Capture and Switch widgets, not just for consistency, but also because amixer simple Switch controls expect different names (cap/nocap instead of on/off) for control values for capture. - Make widget names have capital first letters for all consitutent words. - When resuming, the registers will not have changed, so no need to call snd_soc_component_cache_sync(). - Move reset gpio init to I2C probe - Remove needless _priv at end of private struct name. - Make hex constants lower case - _CBM_CFM/_CBS_CFS -> _CBP_CFP/_CBC_CFC - Instead of using device specific ALSA controls to control the two GPIO pins, use GPIOLIB. GPIO pin functions still can be set up in the devicetree as the pins can be used for other functions which are not GPIO related. - Limitation: The GPIO pins can only be used as outputs, even though the hardware alternatively supports using them as inputs. - Handle GPIO configs in a vector rather than separately. - Use more appropriate module_i2c_driver instead of generic module_init/module_exit. - Remove useless initialization data. - As the ADC mute function had no control, add it. - Clean up GPIO initialization which was done both in the I2C probe and in the component probe. - Remove probe, remove, suspend and resume callbacks, which either do not do anything at all anymore, or uselessly just do set_bias_level. - The PLL should not be configured in the devicetree, so move the corresponding definitions to tlv320adc3xxx.h where they can be picked up by a machine driver if needed. - Default to PLL AUTO mode. - Use "reset-gpios" instead of too-generic "gpios" for dt property for configuring codec reset pin. - Use function for printing PLL mode rather than macro. - dev_info is too verbose for most of the printouts, change them to dev_dbg. - Add support for configuring micbias voltage in DT. - Remove ALSA control for mic bias. - Move PLL and NADC, MADC divisor enable to DAPM, including 10 ms post PLL start up delay via POST_PMU event. - ADC is powered on using DAPM, so needless to do it in set_bias_level, thus remove it. - Let DAPM handle BCLK divisor enable. With this, we can remove set_bias_level as it doesn't do anything anymore. - Fix incorrect argument order to a snd_soc_component_update_bits call. - Although the chip itself can operate using BCLK as the clock source, either directly to the divider chain or via the PLL, this is not yet supported in the driver, so require the clocks property for the MCLK for now. - Move PLL rate structure out of .h file where it does not belong. /Ricard Ricard Wanderlöf (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 +++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1071 +++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 379 ++++++ 6 files changed, 1624 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

Creative Soundblaster AE-7 Headphones Issue
by Emre Fidan 04 Nov '21

04 Nov '21

Hi There, Headphones connected to Creative Soundblaster AE-7 are normally auto-detected on alsamixer, but sink shows the headphone port as not available. alsamixer can change speakers to headphones and vice-versa, however, the audio silences on the headphones after a couple of seconds unless you keep system sound settings/easyeffects/pavucontol opened or minimized. acpi_listen reports Line-In plugged/unplugged instead of headphones, trying to reassign the with hdajackretask does the sensing (acpi_listen shows headphones plugged/unplugged), however, no sound from headphones this time at all. Any suggestions? Thanks. -- Emre Fidan

1 0

HDA-Intel: fixes for Headphone+LO variant (Realtek ALC233)
by GitHub pull_request - opened 04 Nov '21

04 Nov '21

alsa-project/alsa-ucm-conf pull request #116 was opened from perexg: Issue: https://github.com/alsa-project/alsa-ucm-conf/issues/114 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/116 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/116.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Intel SOF HDA: Lenovo m920sff support (Realtek ALC 233)
by GitHub issues - edited 04 Nov '21

04 Nov '21

alsa-project/alsa-ucm-conf issue #114 was edited from perexg: Description: An issue with the headphone detection (Speaker/Headphone auto-mute). More info: https://github.com/thesofproject/linux/issues/3088 and https://bugzilla.suse.com/show_bug.cgi?id=1188685 Mixer: ``` Simple mixer control 'Master',0 Simple mixer control 'Headphone',0 # Switch only Simple mixer control 'Headphone+LO',0 # Volume only Simple mixer control 'Speaker',0 Simple mixer control 'Front Mic',0 Simple mixer control 'Front Mic Boost',0 Simple mixer control 'Line Out',0 # Switch only Simple mixer control 'Mic',0 Simple mixer control 'Mic Boost',0 Simple mixer control 'IEC958',0 Simple mixer control 'IEC958',1 Simple mixer control 'IEC958',2 Simple mixer control 'Capture',0 Simple mixer control 'Auto-Mute Mode',0 Simple mixer control 'Dmic0',0 Simple mixer control 'Dmic1 2nd',0 Simple mixer control 'Loopback Mixing',0 Simple mixer control 'PGA1.0 1 Master',0 Simple mixer control 'PGA2.0 2 Master',0 Simple mixer control 'PGA3.0 3 Master',0 Simple mixer control 'PGA4.0 4 Master',0 Simple mixer control 'PGA7.0 7 Master',0 Simple mixer control 'PGA8.0 8 Master',0 Simple mixer control 'PGA9.0 9 Master',0 ``` Jacks: ``` name 'Mic Jack' name 'Front Mic Jack' name 'Line Out Jack' name 'Front Headphone Jack' name 'Speaker Phantom Jack' name 'HDMI/DP,pcm=3 Jack' name 'HDMI/DP,pcm=4 Jack' name 'HDMI/DP,pcm=5 Jack' ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/114 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Intel SOF HDA: Lenovo m920sff support (Realtek ALC 233)
by GitHub issues - edited 04 Nov '21

04 Nov '21

alsa-project/alsa-ucm-conf issue #114 was edited from perexg: Description: An issue with the headphone detection (Speaker/Headphone auto-mute). More info: https://github.com/thesofproject/linux/issues/3088 and https://bugzilla.suse.com/show_bug.cgi?id=1188685 ``` Simple mixer control 'Master',0 Simple mixer control 'Headphone',0 # Switch only Simple mixer control 'Headphone+LO',0 # Volume only Simple mixer control 'Speaker',0 Simple mixer control 'Front Mic',0 Simple mixer control 'Front Mic Boost',0 Simple mixer control 'Line Out',0 # Switch only Simple mixer control 'Mic',0 Simple mixer control 'Mic Boost',0 Simple mixer control 'IEC958',0 Simple mixer control 'IEC958',1 Simple mixer control 'IEC958',2 Simple mixer control 'Capture',0 Simple mixer control 'Auto-Mute Mode',0 Simple mixer control 'Dmic0',0 Simple mixer control 'Dmic1 2nd',0 Simple mixer control 'Loopback Mixing',0 Simple mixer control 'PGA1.0 1 Master',0 Simple mixer control 'PGA2.0 2 Master',0 Simple mixer control 'PGA3.0 3 Master',0 Simple mixer control 'PGA4.0 4 Master',0 Simple mixer control 'PGA7.0 7 Master',0 Simple mixer control 'PGA8.0 8 Master',0 Simple mixer control 'PGA9.0 9 Master',0 ``` ``` name 'Mic Jack' name 'Front Mic Jack' name 'Line Out Jack' name 'Front Headphone Jack' name 'Speaker Phantom Jack' name 'HDMI/DP,pcm=3 Jack' name 'HDMI/DP,pcm=4 Jack' name 'HDMI/DP,pcm=5 Jack' ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/114 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/realtek: fix flexible_array.cocci warning
by Guo Zhengkui 04 Nov '21

04 Nov '21

Fix following coccicheck warnings: ./sound/pci/hda/patch_realtek.c:128:4-24: WARNING use flexible-array member instead. There is a regular need in the kernel to provide a way to declare having a dynamically sized set of trailing elements in a structure. Kernel code should always use “flexible array members” for these cases. The older style of one-element or zero-length arrays should no longer be used. Signed-off-by: Guo Zhengkui <guozhengkui(a)vivo.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6322fac9e694..7935c363092e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -125,7 +125,7 @@ struct alc_spec { unsigned int pll_coef_idx, pll_coef_bit; unsigned int coef0; struct input_dev *kb_dev; - u8 alc_mute_keycode_map[1]; + u8 alc_mute_keycode_map[]; }; /* -- 2.20.1

2 1

Intel SOF HDA: Lenovo m920sff support (Realtek ALC 233)
by GitHub issues - edited 04 Nov '21

04 Nov '21

alsa-project/alsa-ucm-conf issue #114 was edited from perexg: More info: https://github.com/thesofproject/linux/issues/3088 ``` Simple mixer control 'Master',0 Simple mixer control 'Headphone',0 Simple mixer control 'Headphone+LO',0 Simple mixer control 'Speaker',0 Simple mixer control 'Front Mic',0 Simple mixer control 'Front Mic Boost',0 Simple mixer control 'Line Out',0 Simple mixer control 'Mic',0 Simple mixer control 'Mic Boost',0 Simple mixer control 'IEC958',0 Simple mixer control 'IEC958',1 Simple mixer control 'IEC958',2 Simple mixer control 'Capture',0 Simple mixer control 'Auto-Mute Mode',0 Simple mixer control 'Dmic0',0 Simple mixer control 'Dmic1 2nd',0 Simple mixer control 'Loopback Mixing',0 Simple mixer control 'PGA1.0 1 Master',0 Simple mixer control 'PGA2.0 2 Master',0 Simple mixer control 'PGA3.0 3 Master',0 Simple mixer control 'PGA4.0 4 Master',0 Simple mixer control 'PGA7.0 7 Master',0 Simple mixer control 'PGA8.0 8 Master',0 Simple mixer control 'PGA9.0 9 Master',0 ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/114 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo m920sff support (Realtek ALC 233)
by GitHub issues - opened 04 Nov '21

04 Nov '21

alsa-project/alsa-ucm-conf issue #114 was opened from perexg: More info: https://github.com/thesofproject/linux/issues/3088 ``` Simple mixer control 'Master',0 Simple mixer control 'Headphone',0 Simple mixer control 'Headphone+LO',0 Simple mixer control 'Speaker',0 Simple mixer control 'Front Mic',0 Simple mixer control 'Front Mic Boost',0 Simple mixer control 'Line Out',0 Simple mixer control 'Mic',0 Simple mixer control 'Mic Boost',0 Simple mixer control 'IEC958',0 Simple mixer control 'IEC958',1 Simple mixer control 'IEC958',2 Simple mixer control 'Capture',0 Simple mixer control 'Auto-Mute Mode',0 Simple mixer control 'Dmic0',0 Simple mixer control 'Dmic1 2nd',0 Simple mixer control 'Loopback Mixing',0 Simple mixer control 'PGA1.0 1 Master',0 Simple mixer control 'PGA2.0 2 Master',0 Simple mixer control 'PGA3.0 3 Master',0 Simple mixer control 'PGA4.0 4 Master',0 Simple mixer control 'PGA7.0 7 Master',0 Simple mixer control 'PGA8.0 8 Master',0 Simple mixer control 'PGA9.0 9 Master',0 ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/114 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB audio on riscv64
by Kip Warner 04 Nov '21

04 Nov '21

Hey list, Apologies for double posting on alsa-user(a)lists.sourceforge.net as well. Someone suggested this might be more appropriate for the developer mailing list. I am doing some development on riscv64 using a SiFive Unmatched development board. It has only one PCI express slot which I intend to use for a VGA, but it has a number of USB ports. Is anyone aware of any USB audio devices that are known to work, or likely to work, with ALSA on riscv64? I would have thought any device using the generic USB audio module would have worked. But when I went to plug in a small USB speaker, I see it show up on the bus and yet `aplay -l` shows no devices. Yours truly, -- Kip Warner -- Senior Software Engineer OpenPGP signed/encrypted mail preferred https://www.thevertigo.com

1 0

[PATCH] ASoC: amd: vangogh: add SPI dependency
by Arnd Bergmann 03 Nov '21

03 Nov '21

From: Arnd Bergmann <arnd(a)arndb.de> Without CONFIG_SPI, this fails to build: WARNING: unmet direct dependencies detected for SND_SOC_CS35L41_SPI Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SPI_MASTER [=n] Selected by [m]: - SND_SOC_AMD_VANGOGH_MACH [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_AMD_ACP5x [=m] && I2C [=y] ERROR: modpost: "spi_setup" [sound/soc/codecs/snd-soc-cs35l41-spi.ko] undefined! ERROR: modpost: "__devm_regmap_init_spi" [sound/soc/codecs/snd-soc-cs35l41-spi.ko] undefined! Fixes: 96792fdd77cd ("ASoC: amd: enable vangogh platform machine driver build") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index d91a9399777c..edee2098c82e 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -68,7 +68,7 @@ config SND_SOC_AMD_VANGOGH_MACH tristate "AMD Vangogh support for NAU8821 CS35L41" select SND_SOC_NAU8821 select SND_SOC_CS35L41_SPI - depends on SND_SOC_AMD_ACP5x && I2C + depends on SND_SOC_AMD_ACP5x && I2C && SPI help This option enables machine driver for Vangogh platform using NAU8821 and CS35L41 codecs. -- 2.29.2

2 1

[PATCH -next v2] ALSA: timer: Fix use-after-free problem
by Wang Wensheng 03 Nov '21

03 Nov '21

When the timer instance was add into ack_list but was not currently in process, the user could stop it via snd_timer_stop1() without delete it from the ack_list. Then the user could free the timer instance and when it was actually processed UAF occurred. This issue could be reproduced via testcase snd_timer01 in ltp - running several instances of that testcase at the same time. What I actually met was that the ack_list of the timer broken and the kernel went into deadloop with irqoff. That could be detected by hardlockup detector on board or when we run it on qemu, we could use gdb to dump the ack_list when the console has no response. To fix this issue, we delete the timer instance from ack_list and active_list unconditionally in snd_timer_stop1(). Signed-off-by: Wang Wensheng <wangwensheng4(a)huawei.com> Suggested-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/timer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 92b7008fcdb8..4f9bab931951 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -624,13 +624,13 @@ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) if (!timer) return -EINVAL; spin_lock_irqsave(&timer->lock, flags); + list_del_init(&timeri->ack_list); + list_del_init(&timeri->active_list); if (!(timeri->flags & (SNDRV_TIMER_IFLG_RUNNING | SNDRV_TIMER_IFLG_START))) { result = -EBUSY; goto unlock; } - list_del_init(&timeri->ack_list); - list_del_init(&timeri->active_list); if (timer->card && timer->card->shutdown) goto unlock; if (stop) { -- 2.17.1

2 2

[PATCH] ALSA: hda/realtek: Headset fixup for Clevo NH77HJQ
by Tim Crawford 03 Nov '21

03 Nov '21

From: Jeremy Soller <jeremy(a)system76.com> On Clevo NH77HJ, NH77HP, and their 15" variants, there is a headset microphone input attached to 0x19 that does not have a jack detect. In order to get it working, the pin configuration needs to be set correctly, and a new ALC256_FIXUP_HEADSET_MODE_NO_HP_MIC fixup is applied. This is similar to the existing System76 quirk for ALC293, but for ALC256. Signed-off-by: Jeremy Soller <jeremy(a)system76.com> Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8a3e2fe42106..e6b266416393 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6739,6 +6739,7 @@ enum { ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, ALC287_FIXUP_13S_GEN2_SPEAKERS, ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS, + ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -8450,6 +8451,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_gpio_led, }, + [ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11120 }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8750,11 +8760,15 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x40a1, "Clevo NL40GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x40c1, "Clevo NL40[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x40d1, "Clevo NL41DU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x5015, "Clevo NH5[58]H[HJK]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x5017, "Clevo NH7[79]H[HJK]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x50a3, "Clevo NJ51GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x50b3, "Clevo NK50S[BEZ]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x50b6, "Clevo NK50S5", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x50b8, "Clevo NK50SZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x50d5, "Clevo NP50D5", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50e1, "Clevo NH5[58]HPQ", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x50e2, "Clevo NH7[79]HPQ", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x50f0, "Clevo NH50A[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x50f2, "Clevo NH50E[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x50f3, "Clevo NH58DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.31.1

2 3

[PATCH 0/4] Fix kcontrol put callback in Tegra drivers
by Sameer Pujar 03 Nov '21

03 Nov '21

This series fixes kcontrol put callback in some of the Tegra drivers which are used on platforms based on Tegra210 and later. The callback is expected to return 1 whenever the HW update is done. ADMAIF, I2S, DMIC and DSPK drivers are updated as part of it. This idea is suggested by Jaroslav. Similar suggestion came from Mark during review of series [0] and drivers under those series were taken care. This series takes care of remaining drivers. I have added 'Suggested-by" tags accordingly. [0] https://lore.kernel.org/linux-arm-kernel/20210913142307.GF4283@sirena.org.u… Sameer Pujar (4): ASoC: tegra: Fix kcontrol put callback in ADMAIF ASoC: tegra: Fix kcontrol put callback in I2S ASoC: tegra: Fix kcontrol put callback in DMIC ASoC: tegra: Fix kcontrol put callback in DSPK sound/soc/tegra/tegra186_dspk.c | 4 +++- sound/soc/tegra/tegra210_admaif.c | 4 +++- sound/soc/tegra/tegra210_dmic.c | 4 +++- sound/soc/tegra/tegra210_i2s.c | 4 +++- 4 files changed, 12 insertions(+), 4 deletions(-) -- 2.7.4

2 6

[PATCH] ASoC: codecs: Fix WCD_MBHC_HPH_PA_EN usage
by Christophe JAILLET 03 Nov '21

03 Nov '21

'hphpa_on' is known to be false, so the if block at the end of the function is dead code. Turn it into a meaningful code by having 'hphpa_on' be static. Use is as a flip-flop variable. Fixes: 0e5c9e7ff899 ("ASoC: codecs: wcd: add multi button Headset detection support") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- The purpose of this patch is not to be correct (!) but to draw attention on several points: - in 'wcd_mbhc_adc_hs_rem_irq()', the "if (hphpa_on)" path is dead code because 'hphpa_on' is known to be false - What is this magic number '3'? All 'wcd_mbhc_read_field()' look for 0 or non-0 - a 'mutex_[un]lock()' in an IRQ handler looks spurious to me Instead of this (likely broken) patch, it is likely that something is missing elsewhere. Maybe in 'wcd_mbhc_adc_hs_ins_irq()'. I also guess that 'hphpa_on' should be read for somewhere else. --- sound/soc/codecs/wcd-mbhc-v2.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index 405128ccb4b0..783d8c35bc1b 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -1176,7 +1176,7 @@ static irqreturn_t wcd_mbhc_adc_hs_rem_irq(int irq, void *data) struct wcd_mbhc *mbhc = data; unsigned long timeout; int adc_threshold, output_mv, retry = 0; - bool hphpa_on = false; + static bool hphpa_on = false; mutex_lock(&mbhc->lock); timeout = jiffies + msecs_to_jiffies(WCD_FAKE_REMOVAL_MIN_PERIOD_MS); @@ -1212,6 +1212,9 @@ static irqreturn_t wcd_mbhc_adc_hs_rem_irq(int irq, void *data) if (hphpa_on) { hphpa_on = false; + wcd_mbhc_write_field(mbhc, WCD_MBHC_HPH_PA_EN, 0); + } else { + hphpa_on = true; wcd_mbhc_write_field(mbhc, WCD_MBHC_HPH_PA_EN, 3); } exit: -- 2.30.2

3 3

[PATCH] ALSA: seq: Fix RCU stall in snd_seq_write()
by Zqiang 03 Nov '21

03 Nov '21

If we have a lot of cell object, this cycle may take a long time, and trigger RCU stall. insert a conditional reschedule point to fix it. rcu: INFO: rcu_preempt self-detected stall on CPU rcu: 1-....: (1 GPs behind) idle=9f5/1/0x4000000000000000 softirq=16474/16475 fqs=4916 (t=10500 jiffies g=19249 q=192515) NMI backtrace for cpu 1 ...... asm_sysvec_apic_timer_interrupt RIP: 0010:_raw_spin_unlock_irqrestore+0x38/0x70 spin_unlock_irqrestore snd_seq_prioq_cell_out+0x1dc/0x360 snd_seq_check_queue+0x1a6/0x3f0 snd_seq_enqueue_event+0x1ed/0x3e0 snd_seq_client_enqueue_event.constprop.0+0x19a/0x3c0 snd_seq_write+0x2db/0x510 vfs_write+0x1c4/0x900 ksys_write+0x171/0x1d0 do_syscall_64+0x35/0xb0 Reported-by: syzbot+bb950e68b400ab4f65f8(a)syzkaller.appspotmail.com Signed-off-by: Zqiang <qiang.zhang1211(a)gmail.com> --- sound/core/seq/seq_queue.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index d6c02dea976c..f5b1e4562a64 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -263,6 +263,7 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + cond_resched(); } /* Process time queue... */ @@ -272,6 +273,7 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + cond_resched(); } /* free lock */ -- 2.17.1

2 6

[PATCH] ASoC: tegra: Add master volume/mute control support
by Sameer Pujar 03 Nov '21

03 Nov '21

The MVC module has a per channel control bit, based on which it decides to apply channel specific volume/mute settings. When per channel control bit is enabled (which is the default HW configuration), all MVC channel volume/mute can be independently controlled. If the control is disabled, channel-0 volume/mute setting is applied by HW to all remaining channels. Thus add support to leverage this HW feature by exposing master controls for volume/mute. With this, now there are per channel and master volume/mute controls. Users need to just use controls which are suitable for their applications. The per channel control enable/disable is mananged in driver and hidden from users, so that they need to just worry about respective volume/mute controls. Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/tegra/tegra210_mvc.c | 95 +++++++++++++++++++++++++++++++++++++----- sound/soc/tegra/tegra210_mvc.h | 2 + 2 files changed, 87 insertions(+), 10 deletions(-) diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index 7b9c700..40cd21a 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -123,7 +123,42 @@ static int tegra210_mvc_get_mute(struct snd_kcontrol *kcontrol, mute_mask = (val >> TEGRA210_MVC_MUTE_SHIFT) & TEGRA210_MUTE_MASK_EN; - ucontrol->value.integer.value[0] = mute_mask; + if (strstr(kcontrol->id.name, "Per Chan Mute Mask")) { + /* + * If per channel control is enabled, then return + * exact mute/unmute setting of all channels. + * + * Else report setting based on CH0 bit to reflect + * the correct HW state. + */ + if (val & TEGRA210_MVC_PER_CHAN_CTRL_EN) { + ucontrol->value.integer.value[0] = mute_mask; + } else { + if (mute_mask & TEGRA210_MVC_CH0_MUTE_EN) + ucontrol->value.integer.value[0] = + TEGRA210_MUTE_MASK_EN; + else + ucontrol->value.integer.value[0] = 0; + } + } else { + /* + * If per channel control is disabled, then return + * master mute/unmute setting based on CH0 bit. + * + * Else report settings based on state of all + * channels. + */ + if (!(val & TEGRA210_MVC_PER_CHAN_CTRL_EN)) { + ucontrol->value.integer.value[0] = + mute_mask & TEGRA210_MVC_CH0_MUTE_EN; + } else { + if (mute_mask == TEGRA210_MUTE_MASK_EN) + ucontrol->value.integer.value[0] = + TEGRA210_MVC_CH0_MUTE_EN; + else + ucontrol->value.integer.value[0] = 0; + } + } return 0; } @@ -136,6 +171,7 @@ static int tegra210_mvc_put_mute(struct snd_kcontrol *kcontrol, struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); struct tegra210_mvc *mvc = snd_soc_component_get_drvdata(cmpnt); unsigned int value; + u32 reg_mask; u8 mute_mask; int err; @@ -150,11 +186,22 @@ static int tegra210_mvc_put_mute(struct snd_kcontrol *kcontrol, mute_mask = ucontrol->value.integer.value[0]; - err = regmap_update_bits(mvc->regmap, mc->reg, - TEGRA210_MVC_MUTE_MASK, - mute_mask << TEGRA210_MVC_MUTE_SHIFT); - if (err < 0) - goto end; + if (strstr(kcontrol->id.name, "Per Chan Mute Mask")) { + regmap_update_bits(mvc->regmap, TEGRA210_MVC_CTRL, + TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK, + TEGRA210_MVC_PER_CHAN_CTRL_EN); + + reg_mask = TEGRA210_MVC_MUTE_MASK; + } else { + regmap_update_bits(mvc->regmap, TEGRA210_MVC_CTRL, + TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK, + 0); + + reg_mask = TEGRA210_MVC_CH0_MUTE_MASK; + } + + regmap_update_bits(mvc->regmap, mc->reg, reg_mask, + mute_mask << TEGRA210_MVC_MUTE_SHIFT); return 1; @@ -212,11 +259,31 @@ static int tegra210_mvc_put_vol(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0]); /* Configure init volume same as target volume */ - regmap_write(mvc->regmap, - TEGRA210_MVC_REG_OFFSET(TEGRA210_MVC_INIT_VOL, chan), - mvc->volume[chan]); + if (strstr(kcontrol->id.name, "Channel")) { + regmap_update_bits(mvc->regmap, TEGRA210_MVC_CTRL, + TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK, + TEGRA210_MVC_PER_CHAN_CTRL_EN); + + regmap_write(mvc->regmap, + TEGRA210_MVC_REG_OFFSET(TEGRA210_MVC_INIT_VOL, chan), + mvc->volume[chan]); + + regmap_write(mvc->regmap, reg, mvc->volume[chan]); + } else { + int i; + + regmap_update_bits(mvc->regmap, TEGRA210_MVC_CTRL, + TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK, + 0); + + for (i = 1; i < TEGRA210_MVC_MAX_CHAN_COUNT; i++) + mvc->volume[i] = mvc->volume[0]; - regmap_write(mvc->regmap, reg, mvc->volume[chan]); + regmap_write(mvc->regmap, TEGRA210_MVC_INIT_VOL, + mvc->volume[0]); + + regmap_write(mvc->regmap, reg, mvc->volume[0]); + } regmap_update_bits(mvc->regmap, TEGRA210_MVC_SWITCH, TEGRA210_MVC_VOLUME_SWITCH_MASK, @@ -422,6 +489,14 @@ static const struct snd_kcontrol_new tegra210_mvc_vol_ctrl[] = { TEGRA210_MVC_CTRL, 0, TEGRA210_MUTE_MASK_EN, 0, tegra210_mvc_get_mute, tegra210_mvc_put_mute), + /* Master volume */ + SOC_SINGLE_EXT("Volume", TEGRA210_MVC_TARGET_VOL, 0, 16000, 0, + tegra210_mvc_get_vol, tegra210_mvc_put_vol), + + /* Master mute */ + SOC_SINGLE_EXT("Mute", TEGRA210_MVC_CTRL, 0, 1, 0, + tegra210_mvc_get_mute, tegra210_mvc_put_mute), + SOC_ENUM_EXT("Curve Type", tegra210_mvc_curve_type_ctrl, tegra210_mvc_get_curve_type, tegra210_mvc_put_curve_type), }; diff --git a/sound/soc/tegra/tegra210_mvc.h b/sound/soc/tegra/tegra210_mvc.h index def29c4..7f2567e 100644 --- a/sound/soc/tegra/tegra210_mvc.h +++ b/sound/soc/tegra/tegra210_mvc.h @@ -59,6 +59,8 @@ #define TEGRA210_MUTE_MASK_EN 0xff #define TEGRA210_MVC_MUTE_MASK (TEGRA210_MUTE_MASK_EN << TEGRA210_MVC_MUTE_SHIFT) #define TEGRA210_MVC_MUTE_EN (TEGRA210_MUTE_MASK_EN << TEGRA210_MVC_MUTE_SHIFT) +#define TEGRA210_MVC_CH0_MUTE_EN 1 +#define TEGRA210_MVC_CH0_MUTE_MASK (TEGRA210_MVC_CH0_MUTE_EN << TEGRA210_MVC_MUTE_SHIFT) #define TEGRA210_MVC_PER_CHAN_CTRL_EN_SHIFT 30 #define TEGRA210_MVC_PER_CHAN_CTRL_EN_MASK (1 << TEGRA210_MVC_PER_CHAN_CTRL_EN_SHIFT) -- 2.7.4

3 6

[PATCH -next] ALSA: timer: Fix use-after-free problem
by Wang Wensheng 03 Nov '21

03 Nov '21

When the timer instance was add into ack_list but was not currently in process, the user could stop it via snd_timer_stop1() without delete it from the ack_list. Then the user could free the timer instance and when it was actually processed UAF occurred. This issue could be reproduced via testcase snd_timer01 in ltp - running several instances of that testcase at the same time. What I actually met was that the ack_list of the timer broken and the kernel went into deadloop with irqoff. That could be detected by hardlockup detector on board or when we run it on qemu, we could use gdb to dump the ack_list when the console has no response. To fix this issue, we introduce a new flag SNDRV_TIMER_IFLG_ACKING to indicate the state where the timer instance is in ack_list but not currently processed and check against the new flag in snd_timer_stop1() and delete it from ack_list if the flag is set. Signed-off-by: Wang Wensheng <wangwensheng4(a)huawei.com> --- include/sound/timer.h | 1 + sound/core/timer.c | 12 +++++++++--- 2 files changed, 10 insertions(+), 3 deletions(-) diff --git a/include/sound/timer.h b/include/sound/timer.h index 760e132cc0cd..549288e94a39 100644 --- a/include/sound/timer.h +++ b/include/sound/timer.h @@ -31,6 +31,7 @@ #define SNDRV_TIMER_IFLG_CALLBACK 0x00000020 /* timer callback is active */ #define SNDRV_TIMER_IFLG_EXCLUSIVE 0x00000040 /* exclusive owner - no more instances */ #define SNDRV_TIMER_IFLG_EARLY_EVENT 0x00000080 /* write early event to the poll queue */ +#define SNDRV_TIMER_IFLG_ACKING 0x00000100 /* the timeri was added to ack_list */ #define SNDRV_TIMER_FLG_CHANGE 0x00000001 #define SNDRV_TIMER_FLG_RESCHED 0x00000002 /* need reschedule */ diff --git a/sound/core/timer.c b/sound/core/timer.c index 92b7008fcdb8..1d1e4274919c 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -625,10 +625,12 @@ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) return -EINVAL; spin_lock_irqsave(&timer->lock, flags); if (!(timeri->flags & (SNDRV_TIMER_IFLG_RUNNING | - SNDRV_TIMER_IFLG_START))) { + SNDRV_TIMER_IFLG_START | + SNDRV_TIMER_IFLG_ACKING))) { result = -EBUSY; goto unlock; } + list_del_init(&timeri->ack_list); list_del_init(&timeri->active_list); if (timer->card && timer->card->shutdown) @@ -649,7 +651,8 @@ static int snd_timer_stop1(struct snd_timer_instance *timeri, bool stop) } } } - timeri->flags &= ~(SNDRV_TIMER_IFLG_RUNNING | SNDRV_TIMER_IFLG_START); + timeri->flags &= ~(SNDRV_TIMER_IFLG_RUNNING | SNDRV_TIMER_IFLG_START | + SNDRV_TIMER_IFLG_ACKING); if (stop) timeri->flags &= ~SNDRV_TIMER_IFLG_PAUSED; else @@ -786,6 +789,7 @@ static void snd_timer_process_callbacks(struct snd_timer *timer, /* remove from ack_list and make empty */ list_del_init(&ti->ack_list); + ti->flags &= ~SNDRV_TIMER_IFLG_ACKING; if (!(ti->flags & SNDRV_TIMER_IFLG_DEAD)) { ticks = ti->pticks; @@ -890,8 +894,10 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left) ack_list_head = &timer->ack_list_head; else ack_list_head = &timer->sack_list_head; - if (list_empty(&ti->ack_list)) + if (list_empty(&ti->ack_list)) { list_add_tail(&ti->ack_list, ack_list_head); + ti->flags |= SNDRV_TIMER_IFLG_ACKING; + } list_for_each_entry(ts, &ti->slave_active_head, active_list) { ts->pticks = ti->pticks; ts->resolution = resolution; -- 2.17.1

3 2

[PATCH v5 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 02 Nov '21

02 Nov '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver Changes Since V4: -- Add COMPILE_TEST flag in sc7280 configuration. -- Remove redundant startup and shutdown callbacks of snd_soc_ops. -- Fix typo errors. Changes Since V3: -- Change audio jack playpause key value. Changes Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Changes Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 170 ++++++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7280.c | 299 +++++++++++++++++++++ 4 files changed, 483 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 3

[PATCH] ASoC: rsnd: Fix an error handling path in 'rsnd_node_count()'
by Christophe JAILLET 02 Nov '21

02 Nov '21

If we return before the end of the 'for_each_child_of_node()' iterator, the reference taken on 'np' must be released. Add the missing 'of_node_put()' call. Fixes: c413983eb66a ("ASoC: rsnd: adjust disabled module") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/sh/rcar/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 978bd0406729..6a8fe0da7670 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1225,6 +1225,7 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name if (i < 0) { dev_err(dev, "strange node numbering (%s)", of_node_full_name(node)); + of_node_put(np); return 0; } i++; -- 2.30.2

3 2

[PATCH] sound: soc: tlv320aic3x: Make aic3x_remove() return void
by Uwe Kleine-König 02 Nov '21

02 Nov '21

Up to now aic3x_remove() returns zero unconditionally. Make it return void instead which makes it easier to see in the callers that there is no error to handle. Also the return value of i2c and spi remove callbacks is ignored anyway. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/codecs/tlv320aic3x-i2c.c | 4 +++- sound/soc/codecs/tlv320aic3x-spi.c | 4 +++- sound/soc/codecs/tlv320aic3x.c | 3 +-- sound/soc/codecs/tlv320aic3x.h | 2 +- 4 files changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x-i2c.c b/sound/soc/codecs/tlv320aic3x-i2c.c index cd0558ed4dd4..2f272bc3f5da 100644 --- a/sound/soc/codecs/tlv320aic3x-i2c.c +++ b/sound/soc/codecs/tlv320aic3x-i2c.c @@ -32,7 +32,9 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *i static int aic3x_i2c_remove(struct i2c_client *i2c) { - return aic3x_remove(&i2c->dev); + aic3x_remove(&i2c->dev); + + return 0; } static const struct i2c_device_id aic3x_i2c_id[] = { diff --git a/sound/soc/codecs/tlv320aic3x-spi.c b/sound/soc/codecs/tlv320aic3x-spi.c index 8c7b6bb9223f..494e84402232 100644 --- a/sound/soc/codecs/tlv320aic3x-spi.c +++ b/sound/soc/codecs/tlv320aic3x-spi.c @@ -37,7 +37,9 @@ static int aic3x_spi_probe(struct spi_device *spi) static int aic3x_spi_remove(struct spi_device *spi) { - return aic3x_remove(&spi->dev); + aic3x_remove(&spi->dev); + + return 0; } static const struct spi_device_id aic3x_spi_id[] = { diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 7731593a5509..d53037b1509d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1870,7 +1870,7 @@ int aic3x_probe(struct device *dev, struct regmap *regmap, kernel_ulong_t driver } EXPORT_SYMBOL(aic3x_probe); -int aic3x_remove(struct device *dev) +void aic3x_remove(struct device *dev) { struct aic3x_priv *aic3x = dev_get_drvdata(dev); @@ -1881,7 +1881,6 @@ int aic3x_remove(struct device *dev) gpio_set_value(aic3x->gpio_reset, 0); gpio_free(aic3x->gpio_reset); } - return 0; } EXPORT_SYMBOL(aic3x_remove); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 7e0063913017..14298f9e6d9b 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -14,7 +14,7 @@ struct regmap_config; extern const struct regmap_config aic3x_regmap; int aic3x_probe(struct device *dev, struct regmap *regmap, kernel_ulong_t driver_data); -int aic3x_remove(struct device *dev); +void aic3x_remove(struct device *dev); #define AIC3X_MODEL_3X 0 #define AIC3X_MODEL_33 1 -- 2.30.2

2 2

[PATCH v4 0/2] Add code to manage DSP clocks and provide dts-binding document
by YC Hung 02 Nov '21

02 Nov '21

From: "yc.hung" <yc.hung(a)mediatek.com> This code is based on top of SOF topic/sof-dev branch and we want to have a review with ALSA and device Tree communities the it will be merged to SOF tree and then merged into ALSA tree. It provides two patches, one is for mt8195 dsp clocks related. Another is for mt8195 dsp dts binding decription. YC Hung (2): ASoC: SOF: mediatek: Add mt8195 dsp clock support dt-bindings: dsp: mediatek: Add mt8195 DSP binding support .../bindings/dsp/mtk,mt8195-dsp.yaml | 139 +++++++++++++++ sound/soc/sof/mediatek/adsp_helper.h | 2 +- sound/soc/sof/mediatek/mt8195/Makefile | 2 +- sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 158 ++++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195-clk.h | 28 ++++ sound/soc/sof/mediatek/mt8195/mt8195.c | 22 ++- 6 files changed, 347 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/mtk,mt8195-dsp.yaml create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-clk.h -- 2.18.0

4 6

[PATCH] ALSA: usb-audio: Add quirk for Audient iD14
by Takashi Iwai 02 Nov '21

02 Nov '21

Audient iD14 (2708:0002) may get a control message error that interferes the operation e.g. with alsactl. Add the quirk to ignore such errors like other devices. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1191247 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 1d418c3fc598..64e1c20311ed 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1888,6 +1888,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x21b4, 0x0081, /* AudioQuest DragonFly */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */ + QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x2912, 0x30c8, /* Audioengine D1 */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x30be, 0x0101, /* Schiit Hel */ -- 2.31.1

1 0

[PATCH] Make top/rear speaker, mute and micmute leds work on HP x360 14-ea000 laptops that use Realtek 245 codec
by Jonathan Clarke 02 Nov '21

02 Nov '21

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210633 Signed-off-by: Jonathan Clarke <jonathan.a.clarke(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 22d27b12c..e3c6d17ea 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4409,6 +4409,13 @@ static void alc245_fixup_hp_x360_amp(struct hda_codec *codec, case HDA_FIXUP_ACT_PRE_PROBE: spec->gpio_mask |= 0x01; spec->gpio_dir |= 0x01; + + /* use only amp at 0x02 for bottom(front) speaker, + * otherwise it is set to use 0x02,0x03,0x06 and when used in conjunction + * with top(rear) speaker 0x14, gets locked at full volume */ + static const hda_nid_t conn1[] = { 0x02 }; + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn1), conn1); + break; case HDA_FIXUP_ACT_INIT: /* need to toggle GPIO to enable the amp */ @@ -4503,6 +4510,26 @@ static void alc236_fixup_hp_mute_led_coefbit(struct hda_codec *codec, } } +static void alc245_fixup_hp_x360_mute_leds(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + /* mic mute is set via gpio 0x04 */ + spec->micmute_led_polarity = 1; + codec->power_filter = led_power_filter; + alc_fixup_hp_gpio_led(codec, action, 0x00, 0x04); + + /* output mute is set via SET_COEF_INDEX,SET_PROC_COEF */ + spec->mute_led_polarity = 0; + spec->mute_led_coef.idx = 0x0b; + spec->mute_led_coef.mask = 0xffff; + spec->mute_led_coef.on = 0xa02f; + spec->mute_led_coef.off = 0x7774; + snd_hda_gen_add_mute_led_cdev(codec, coef_mute_led_set); + } +} + /* turn on/off mic-mute LED per capture hook by coef bit */ static int coef_micmute_led_set(struct led_classdev *led_cdev, enum led_brightness brightness) @@ -6557,6 +6584,8 @@ enum { ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, ALC245_FIXUP_HP_X360_AMP, + ALC245_FIXUP_HP_X360_MUTE_LEDS, + ALC245_FIXUP_HP_X360_GPIO_TOP_SPEAKER, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, @@ -7293,6 +7322,21 @@ static const struct hda_fixup alc269_fixups[] = { [ALC245_FIXUP_HP_X360_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_x360_amp, + .chained = true, + .chain_id = ALC245_FIXUP_HP_X360_MUTE_LEDS + }, + [ALC245_FIXUP_HP_X360_MUTE_LEDS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc245_fixup_hp_x360_mute_leds, + .chained = true, + .chain_id = ALC245_FIXUP_HP_X360_GPIO_TOP_SPEAKER + }, + [ALC245_FIXUP_HP_X360_GPIO_TOP_SPEAKER] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170110 }, /* enable top(back) speaker in addition to bottom(front) speaker at 0x17 */ + { }, + } }, [ALC288_FIXUP_DELL_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, @@ -9003,6 +9047,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, + {.id = ALC245_FIXUP_HP_X360_MUTE_LEDS, .name = "alc245-hp-x360-mute-leds"}, + {.id = ALC245_FIXUP_HP_X360_GPIO_TOP_SPEAKER, .name = "alc245-hp-x360-gpio-top-speaker"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, -- 2.25.1

3 6

[PATCH v2] ALSA: seq: Fix RCU stall in snd_seq_write()
by Zqiang 02 Nov '21

02 Nov '21

If we have a lot of cell object, this cycle may take a long time, and trigger RCU stall. insert a conditional reschedule point to fix it. rcu: INFO: rcu_preempt self-detected stall on CPU rcu: 1-....: (1 GPs behind) idle=9f5/1/0x4000000000000000 softirq=16474/16475 fqs=4916 (t=10500 jiffies g=19249 q=192515) NMI backtrace for cpu 1 ...... asm_sysvec_apic_timer_interrupt RIP: 0010:_raw_spin_unlock_irqrestore+0x38/0x70 spin_unlock_irqrestore snd_seq_prioq_cell_out+0x1dc/0x360 snd_seq_check_queue+0x1a6/0x3f0 snd_seq_enqueue_event+0x1ed/0x3e0 snd_seq_client_enqueue_event.constprop.0+0x19a/0x3c0 snd_seq_write+0x2db/0x510 vfs_write+0x1c4/0x900 ksys_write+0x171/0x1d0 do_syscall_64+0x35/0xb0 Reported-by: syzbot+bb950e68b400ab4f65f8(a)syzkaller.appspotmail.com Signed-off-by: Zqiang <qiang.zhang1211(a)gmail.com> --- v1->v2: Add atomic context detection. sound/core/seq/seq_queue.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index d6c02dea976c..3a05b5e0d555 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -263,6 +263,8 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (!atomic) + cond_resched(); } /* Process time queue... */ @@ -272,6 +274,8 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) if (!cell) break; snd_seq_dispatch_event(cell, atomic, hop); + if (!atomic) + cond_resched(); } /* free lock */ -- 2.17.1

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Config file for a52 plugin (60-a52-encoder.conf) produces no/garbled sound via optical digital out (S/PDIF)
by GitHub issues - edited 02 Nov '21

02 Nov '21

alsa-project/alsa-plugins issue #38 was edited from jensgw: Dear Maintainer, this bug was originally reported on the debian tracker. They advised me to post it here. https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=998125 * What led up to the situation? Fresh Debian 11 install. I want to output 5.1 sound via toslink. Used KDE settings to switch from analog output to digital out 5.1. This produced no or garbled sound and Youtube videos would play back way to fast without sound. Then I switched to digital out stereo, this worked as expected but only stereo. I could trace back the error to the shipped config file for alsa's a52 plugin: 60-a52-encoder.conf. When I replaced that file with a much simpler one (see below) 5.1 digital out worked as expected. I tried with onboard sound (PCH9) and dedicated card (Audigy Z) with the same result, the inlcuded a52 config file does not work. == Start working a52.conf ``` pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 448 channels 6 card $CARD } rate 48000 } } ``` == End working a52.conf Therefore I assume the shipped config file with debian has an error somewhere. Please fix this. Best reagards Jens -- System Information: Debian Release: 11.1 Architecture: amd64 (x86_64) Kernel: Linux 5.14.0-0.bpo.2-amd64 (SMP w/4 CPU threads) Kernel taint flags: TAINT_UNSIGNED_MODULE Locale: LANG=de_DE.UTF-8, LC_CTYPE=de_DE.UTF-8 (charmap=UTF-8), LANGUAGE=en_US:en Shell: /bin/sh linked to /usr/bin/dash Init: systemd (via /run/systemd/system) Versions of packages libasound2-plugins depends on: ii libasound2 1.2.4-1.1 ii libavcodec-extra58 [libavcodec58] 7:4.3.2-0+deb11u2 ii libavresample4 7:4.3.2-0+deb11u2 ii libavutil56 7:4.3.2-0+deb11u2 ii libc6 2.31-13+deb11u2 ii libjack0 [libjack-0.125] 1:0.125.0-3+b1 ii libpulse0 14.2-2 ii libsamplerate0 0.2.1+ds0-1 Issue URL : https://github.com/alsa-project/alsa-plugins/issues/38 Repository URL: https://github.com/alsa-project/alsa-plugins

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Config file for a52 plugin (60-a52-encoder.conf) produces no/garbled sound via optical digital out (S/PDIF)
by GitHub issues - edited 02 Nov '21

02 Nov '21

alsa-project/alsa-plugins issue #38 was edited from jensgw: Dear Maintainer, this bug was originally reported on the debian tracker. They advised me to post it here. https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=998125 * What led up to the situation? Fresh Debian 11 install. I want to output 5.1 sound via toslink. Used KDE settings to switch from analog output to digital out 5.1. This produced no or garbled sound and Youtube videos would play back way to fast without sound. Then I switched to digital out stereo, this worked as expected but only stereo. I could trace back the error to the shipped config file for alsa's a52 plugin: 60-a52-encoder.conf. When I replaced that file with a much simpler one (see below) 5.1 digital out worked as expected. I tried with onboard sound (PCH9) and dedicated card (Audigy Z) with the same result, the inlcuded a52 config file does not work. == Start a52.conf ``` pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 448 channels 6 card $CARD } rate 48000 } } ``` == End a52.conf Therefore I assume the shipped config file with debian has an error somewhere. Please fix this. Best reagards Jens -- System Information: Debian Release: 11.1 Architecture: amd64 (x86_64) Kernel: Linux 5.14.0-0.bpo.2-amd64 (SMP w/4 CPU threads) Kernel taint flags: TAINT_UNSIGNED_MODULE Locale: LANG=de_DE.UTF-8, LC_CTYPE=de_DE.UTF-8 (charmap=UTF-8), LANGUAGE=en_US:en Shell: /bin/sh linked to /usr/bin/dash Init: systemd (via /run/systemd/system) Versions of packages libasound2-plugins depends on: ii libasound2 1.2.4-1.1 ii libavcodec-extra58 [libavcodec58] 7:4.3.2-0+deb11u2 ii libavresample4 7:4.3.2-0+deb11u2 ii libavutil56 7:4.3.2-0+deb11u2 ii libc6 2.31-13+deb11u2 ii libjack0 [libjack-0.125] 1:0.125.0-3+b1 ii libpulse0 14.2-2 ii libsamplerate0 0.2.1+ds0-1 Issue URL : https://github.com/alsa-project/alsa-plugins/issues/38 Repository URL: https://github.com/alsa-project/alsa-plugins

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Config file for a52 plugin (60-a52-encoder.conf) produces no/garbled sound via optical digital out (S/PDIF)
by GitHub issues - edited 02 Nov '21

02 Nov '21

alsa-project/alsa-plugins issue #38 was edited from jensgw: Dear Maintainer, this bug was originally reported on the debian tracker. They advised me to post it here. https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=998125 * What led up to the situation? Fresh Debian 11 install. I want to output 5.1 sound via toslink. Used KDE settings to switch from analog output to digital out 5.1. This produced no or garbled sound and Youtube videos would play back way to fast without sound. Then I switched to digital out stereo, this worked as expected but only stereo. I could trace back the error to the shipped config file for alsa's a52 plugin: 60-a52-encoder.conf. When I replaced that file with a much simpler one (see below) 5.1 digital out worked as expected. I tried with onboard sound (PCH9) and dedicated card (Audigy Z) with the same result, the inlcuded a52 config file does not work. == Start a52.conf `pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 448 channels 6 card $CARD } rate 48000 } }` == End a52.conf Therefore I assume the shipped config file with debian has an error somewhere. Please fix this. Best reagards Jens -- System Information: Debian Release: 11.1 Architecture: amd64 (x86_64) Kernel: Linux 5.14.0-0.bpo.2-amd64 (SMP w/4 CPU threads) Kernel taint flags: TAINT_UNSIGNED_MODULE Locale: LANG=de_DE.UTF-8, LC_CTYPE=de_DE.UTF-8 (charmap=UTF-8), LANGUAGE=en_US:en Shell: /bin/sh linked to /usr/bin/dash Init: systemd (via /run/systemd/system) Versions of packages libasound2-plugins depends on: ii libasound2 1.2.4-1.1 ii libavcodec-extra58 [libavcodec58] 7:4.3.2-0+deb11u2 ii libavresample4 7:4.3.2-0+deb11u2 ii libavutil56 7:4.3.2-0+deb11u2 ii libc6 2.31-13+deb11u2 ii libjack0 [libjack-0.125] 1:0.125.0-3+b1 ii libpulse0 14.2-2 ii libsamplerate0 0.2.1+ds0-1 Issue URL : https://github.com/alsa-project/alsa-plugins/issues/38 Repository URL: https://github.com/alsa-project/alsa-plugins

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Config file for a52 plugin (60-a52-encoder.conf) produces no/garbled sound via optical digital out (S/PDIF)
by GitHub issues - edited 02 Nov '21

02 Nov '21

alsa-project/alsa-plugins issue #38 was edited from jensgw: Dear Maintainer, this bug was originally reported on the debian tracker. They advised me to post it here. https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=998125 * What led up to the situation? Fresh Debian 11 install. I want to output 5.1 sound via toslink. Used KDE settings to switch from analog output to digital out 5.1. This produced no or garbled sound and Youtube videos would play back way to fast without sound. Then I switched to digital out stereo, this worked as expected but only stereo. I could trace back the error to the shipped config file for alsa's a52 plugin: 60-a52-encoder.conf. When I replaced that file with a much simpler one (see below) 5.1 digital out worked as expected. I tried with onboard sound (PCH9) and dedicated card (Audigy Z) with the same result, the inlcuded a52 config file does not work. == Start a52.conf pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 448 channels 6 card $CARD } rate 48000 } } == End a52.conf Therefore I assume the shipped config file with debian has an error somewhere. Please fix this. Best reagards Jens -- System Information: Debian Release: 11.1 Architecture: amd64 (x86_64) Kernel: Linux 5.14.0-0.bpo.2-amd64 (SMP w/4 CPU threads) Kernel taint flags: TAINT_UNSIGNED_MODULE Locale: LANG=de_DE.UTF-8, LC_CTYPE=de_DE.UTF-8 (charmap=UTF-8), LANGUAGE=en_US:en Shell: /bin/sh linked to /usr/bin/dash Init: systemd (via /run/systemd/system) Versions of packages libasound2-plugins depends on: ii libasound2 1.2.4-1.1 ii libavcodec-extra58 [libavcodec58] 7:4.3.2-0+deb11u2 ii libavresample4 7:4.3.2-0+deb11u2 ii libavutil56 7:4.3.2-0+deb11u2 ii libc6 2.31-13+deb11u2 ii libjack0 [libjack-0.125] 1:0.125.0-3+b1 ii libpulse0 14.2-2 ii libsamplerate0 0.2.1+ds0-1 Issue URL : https://github.com/alsa-project/alsa-plugins/issues/38 Repository URL: https://github.com/alsa-project/alsa-plugins

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Config file for a52 plugin (60-a52-encoder.conf) produces no/garbled sound via optical digital out (S/PDIF)
by GitHub issues - opened 02 Nov '21

02 Nov '21

alsa-project/alsa-plugins issue #38 was opened from jensgw: Dear Maintainer, this bug was originally reported on the debian tracker. They advised me to post it here. https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=998125 * What led up to the situation? Fresh Debian 11 install. I want to output 5.1 sound via toslink. Used KDE settings to switch from analog output to digital out 5.1. This produced no or garbled sound and Youtube videos would play back way to fast without sound. Then I switched to digital out stereo, this worked as expected but only stereo. I could trace back the error to the shipped config file for alsa's a52 plugin: 60-a52-encoder.conf. When I replaced that file with a much simpler one (see below) 5.1 digital out worked as expedted. I tried with onboard sound (PCH9) and dedicated card (Audigy Z) with the same result, the inlcuded a52 config file does not work. == Start a52.conf pcm.a52 { @args [CARD] @args.CARD { type string } type rate slave { pcm { type a52 bitrate 448 channels 6 card $CARD } rate 48000 } } == End a52.conf Therefore I assume the shipped config file with debian has an error somewhere. Please fix this. Best reagards Jens -- System Information: Debian Release: 11.1 Architecture: amd64 (x86_64) Kernel: Linux 5.14.0-0.bpo.2-amd64 (SMP w/4 CPU threads) Kernel taint flags: TAINT_UNSIGNED_MODULE Locale: LANG=de_DE.UTF-8, LC_CTYPE=de_DE.UTF-8 (charmap=UTF-8), LANGUAGE=en_US:en Shell: /bin/sh linked to /usr/bin/dash Init: systemd (via /run/systemd/system) Versions of packages libasound2-plugins depends on: ii libasound2 1.2.4-1.1 ii libavcodec-extra58 [libavcodec58] 7:4.3.2-0+deb11u2 ii libavresample4 7:4.3.2-0+deb11u2 ii libavutil56 7:4.3.2-0+deb11u2 ii libc6 2.31-13+deb11u2 ii libjack0 [libjack-0.125] 1:0.125.0-3+b1 ii libpulse0 14.2-2 ii libsamplerate0 0.2.1+ds0-1 Issue URL : https://github.com/alsa-project/alsa-plugins/issues/38 Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH RESEND] ALSA: hda/realtek: Add quirk for Clevo PC70HS
by Tim Crawford 02 Nov '21

02 Nov '21

Apply the PB51ED PCI quirk to the Clevo PC70HS. Fixes audio output from the internal speakers. Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6322fac9e694..8a3e2fe42106 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2539,6 +2539,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x67f1, "Clevo PC70H[PRS]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170SM", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x7715, "Clevo X170KM-G", ALC1220_FIXUP_CLEVO_PB51ED), -- 2.31.1

2 1

alsatplg broken with musl-libc
by GitHub issues - opened 01 Nov '21

01 Nov '21

alsa-project/alsa-lib issue #187 was opened from Johnnynator: alsatplg feeds empty and non Integer strings to `strtol(3)` and friends. Glibc just silently accepts such inputs and does not set errno and returns 0. Musl sets `errno` to `EINVAL` with such inputs. This can easily be seen by most of the `alsatplg` invocation when compiling https://github.com/thesofproject/sof/ `STRTOL(3)` manpage explains that this behavior is allowed. > The implementation may also set errno to EINVAL in case no conversion was performed (no digits seen, and 0 returned). There are like two ways I see this can be fixed a) ignore EINVAL (and check if base is correct ourself) b) make sure that strtol is only getting called with a valid input (I'm not sure if the current behavior is intended to just return 0 if e,g, `tplg_get_unsigned` did not even parse anything valid Issue URL : https://github.com/alsa-project/alsa-lib/issues/187 Repository URL: https://github.com/alsa-project/alsa-lib

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[GIT PULL] ASoC updates for v5.16
by Mark Brown 01 Nov '21

01 Nov '21

The following changes since commit 519d81956ee277b4419c723adfb154603c2565ba: Linux 5.15-rc6 (2021-10-17 20:00:13 -1000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.16 for you to fetch changes up to 318a54c0ee4aaa3bfd69fdf505588510c7672c0c: Merge remote-tracking branch 'asoc/for-5.16' into asoc-next (2021-10-29 22:00:00 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.16 This is an unusually large set of updates, mostly a large crop of unusually big drivers coupled with extensive overhauls of existing code. There's a SH change here for the DAI format terminology, the change is straightforward and the SH maintainers don't seem very active. - A new version of the audio graph card which supports a wider range of systems. - Move of the Cirrus DSP framework into drivers/firmware to allow for future use by non-audio DSPs. - Several conversions to YAML DT bindings. - Continuing cleanups to the SOF and Intel code. - A very big overhaul of the cs42l42 driver, correcting many problems. - Support for AMD Vangogh and Yelow Cap, Cirrus CS35L41, Maxim MAX98520 and MAX98360A, Mediatek MT8195, Nuvoton NAU8821, nVidia Tegra210, NXP i.MX8ULP, Qualcomm AudioReach, Realtek ALC5682I-VS, RT5682S, and RT9120 and Rockchip RV1126 and RK3568 ---------------------------------------------------------------- Ajit Kumar Pandey (10): ASoC: amd: Kconfig: Select fch clock support with machine driver ASoC: amd: Add common framework to support I2S on ACP SOC ASoC: amd: acp: Add I2S support on Renoir platform ASoC: amd: acp: Add callback for machine driver on ACP ASoC: amd: acp: Add generic machine driver support for ACP cards ASoC: amd: acp: Add legacy sound card support for Chrome audio ASoC: amd: acp: Add SOF audio support on Chrome board ASoC: amd: acp: Add support for Maxim amplifier codec ASoC: amd: acp: Add support for RT5682-VS codec ASoC: amd: acp: Add acp_machine struct for renoir platform. Alejandro Tafalla (2): ASoC: max98927: Handle reset gpio when probing i2c ASoC: max98927: Add reset-gpios optional property Alexander Stein (1): ASoC: meson: t9015: Add missing AVDD-supply property Amadeusz Sławiński (5): ASoC: core: Remove invalid snd_soc_component_set_jack call ASoC: topology: Check for dapm widget completeness ASoC: topology: Use correct device for prints ASoC: topology: Change topology device to card device ASoC: Stop dummy from overriding hwparams Andy Shevchenko (12): ASoC: Intel: bytcht_es8316: Get platform data via dev_get_platdata() ASoC: Intel: bytcht_es8316: Use temporary variable for struct device ASoC: Intel: bytcht_es8316: Switch to use gpiod_get_optional() ASoC: Intel: bytcht_es8316: Utilize dev_err_probe() to avoid log saturation ASoC: Intel: bytcr_rt5640: Get platform data via dev_get_platdata() ASoC: Intel: bytcr_rt5640: Use temporary variable for struct device ASoC: Intel: bytcr_rt5640: use devm_clk_get_optional() for mclk ASoC: Intel: bytcr_rt5640: Utilize dev_err_probe() to avoid log saturation ASoC: Intel: bytcr_rt5651: Get platform data via dev_get_platdata() ASoC: Intel: bytcr_rt5651: Use temporary variable for struct device ASoC: Intel: bytcr_rt5651: use devm_clk_get_optional() for mclk ASoC: Intel: bytcr_rt5651: Utilize dev_err_probe() to avoid log saturation AngeloGioacchino Del Regno (1): ASoC: rt5682-i2c: Use devm_clk_get_optional for optional clock Arnd Bergmann (3): ASoC: samsung: add missing "fallthrough;" ASoC: amd: acp: fix Kconfig dependencies ASoC: amd: acp: select CONFIG_SND_SOC_ACPI Bard Liao (1): ASoC: SOF: dai-intel: add SOF_DAI_INTEL_SSP_CLKCTRL_MCLK/BCLK_ES bits Bixuan Cui (1): ASoC: mediatek: mt8195: Add missing of_node_put() Brent Lu (9): ASoC: Intel: sof_rt5682: support ALC5682I-VS codec ASoC: intel: sof_rt5682: support jsl_rt5682s_rt1015p board ASoC: intel: sof_rt5682: support jsl_rt5682s_rt1015 board ASoC: intel: sof_rt5682: support jsl_rt5682s_mx98360a board ASoC: intel: sof_rt5682: update platform device name for Maxim amplifier ASoC: Intel: glk_rt5682_max98357a: support ALC5682I-VS codec ASoC: soc-acpi: add comp_ids field for machine driver matching ASoC: Intel: sof_rt5682: detect codec variant in probe function ASoC: Intel: sof_rt5682: use comp_ids to enumerate rt5682s Bud Liviu-Alexandru (1): ASoC: SOF: Make Intel IPC stream ops generic Cai Huoqing (1): ASoC: mediatek: mt8195: Make use of the helper function devm_platform_ioremap_resource() Cezary Rojewski (1): ASoC: topology: Add header payload_size verification Charles Keepax (10): ASoC: cs35l41: Fix use of an uninitialised variable ASoC: cs35l41: Use regmap_read_poll_timeout to wait for OTP boot ASoC: cs35l41: Combine adjacent register writes ASoC: cs35l41: Don't overwrite returned error code ASoC: cs35l41: Fixup the error messages ASoC: cs35l41: Fix a bunch of trivial code formating/style issues misc: cs35l41: Remove unused pdn variable ASoC: wm_adsp: Move check for control existence ASoC: wm_adsp: Switch to using wm_coeff_read_ctrl for compressed buffers ASoC: wm_adsp: Move sys_config_size to wm_adsp ChiYuan Huang (4): ASoC: dt-bindings: rt9120: Add initial bindings ASoC: rt9120: Add rt9210 audio amplifier support ASoC: dt-bindings: rt9120: Add initial bindings ASoC: rt9120: Add rt9210 audio amplifier support Christian Hewitt (1): ASoC: meson: implement driver_name for snd_soc_card in meson-card-utils Christophe JAILLET (1): ASoC: rsnd: Fix an error handling path in 'rsnd_node_count()' Colin Ian King (7): ASoC: mediatek: mt8195: Fix unused initialization of pointer etdm_data ASoC: mediatek: mt8195: make array adda_dai_list static const ASoC: qdsp6: q6afe-dai: Fix spelling mistake "Fronend" -> "Frontend" ASoC: codecs: Fix spelling mistake "Unsupport" -> "Unsupported" ASoC: meson: aiu: Fix spelling mistake "Unsupport" -> "Unsupported" ASoC: rockchip: i2s-tdm: Remove call to rockchip_i2s_ch_to_io ASoC: rockchip: i2s-tdm: Fix error handling on i2s_tdm_prepare_enable_mclk failure Daniel Baluta (4): ASoC: SOF: OF: Add fw_path and tplg_path parameters ASoC: SOF: Introduce snd_sof_mailbox_read / snd_sof_mailbox_write callbacks ASoC: SOF: imx: Use newly introduced generic IPC stream ops ASoC: SOF: Introduce fragment elapsed notification API David Lin (2): ASoC: nau8825: add set_jack coponment support ASoC: nau8825: add clock management for power saving David Rhodes (3): ASoC: cs35l41: CS35L41 Boosted Smart Amplifier ASoC: cs35l41: Add bindings for CS35L41 ASoC: cs35l41: Binding fixes Derek Fang (11): ASoC: rt5682s: Add driver for ALC5682I-VS codec ASoC: dt-bindings: rt5682s: add bindings for rt5682s ASoC: rt5682s: Remove the volatile SW reset register from reg_default ASoC: rt5682s: Use dev_dbg instead of pr_debug ASoC: rt5682s: Revise the macro RT5682S_PLLB_SRC_MASK ASoC: dt-bindings: rt5682s: fix the device-tree schema errors ASoC: rt5682s: Enable ASRC auto-disable to fix pop during jack plug-in while playback ASoC: rt5682s: Fix HP noise caused by SAR mode switch when the system resumes ASoC: rt5682s: Fix hp pop produced immediately after resuming ASoC: rt5682: fix a little pop while playback ASoC: rt5682s: Downsizing the DAC volume scale Dmitry Osipenko (2): ASoC: tegra: Restore AC97 support ASoC: tegra: Set default card name for Trimslice Geert Uytterhoeven (7): ASoC: dt-bindings: wlf,wm8978: Fix I2C address in example ASoC: dt-bindings: rockchip: i2s-tdm: Fix rockchip,i2s-[rt]x-route ASoC: wm8962: Convert to devm_clk_get_optional() ASoC: dt-bindings: wlf,wm8962: Convert to json-schema ASoC: amd: acp: Wrap AMD Audio ACP components in SND_SOC_AMD_ACP_COMMON ASoC: amd: acp: SND_SOC_AMD_{LEGACY_MACH,SOF_MACH} should depend on X86 && PCI && I2C ASoC: amd: acp: SND_SOC_AMD_ACP_COMMON should depend on X86 && PCI George Song (2): ASoC: dt-bindings: max98520: add initial bindings ASoC: max98520: add max98520 audio amplifier driver Guennadi Liakhovetski (1): ASoC: SOF: add error handling to snd_sof_ipc_msg_data() Guenter Roeck (1): spi: tegra20-slink: Declare runtime suspend and resume functions conditionally Guo Zhengkui (1): ASoC: wm_adsp: remove a repeated including Hans de Goede (6): ASoC: es8316: Use IRQF_NO_AUTOEN when requesting the IRQ ASoC: rt5651: Use IRQF_NO_AUTOEN when requesting the IRQ ASoC: nau8824: Fix NAU8824_JACK_LOGIC define ASoC: nau8824: Add DMI quirk mechanism for active-high jack-detect ASoC: nau8824: Add a nau8824_components() helper ASoC: Intel: cht_bsw_nau8824: Set card.components string Jack Yu (2): ASoC: rt1011: add i2s reference control for rt1011 ASoC: rt5682: move clk related code to rt5682_i2c_probe Jerome Brunet (2): ASoC: meson: axg-card: make links nonatomic ASoC: meson: axg-tdm-interface: manage formatters in trigger Jiapeng Chong (1): ASoC: rt5682s: make rt5682s_aif2_dai_ops and rt5682s_soc_component_dev John Keeping (1): ASoC: doc: update codec example code Julian Braha (4): ASoC: fix unmet dependency on GPIOLIB ASoC: fix unmet dependency on GPIOLIB for SND_SOC_MAX98357A ASoC: fix unmet dependencies on GPIOLIB for SND_SOC_DMIC ASoC: fix unmet dependencies on GPIOLIB for SND_SOC_RT1015P Krzysztof Kozlowski (1): ASoC: dt-bindings: rt5682s: correct several errors Kunihiko Hayashi (1): ASoC: dt-bindings: uniphier: Add description of each port number Kuninori Morimoto (21): ASoC: test-component: add Test Component YAML bindings ASoC: test-component: add Test Component for Sound debug/test ASoC: simple-card-utils: add asoc_graph_is_ports0() ASoC: simple-card-utils: add codec2codec support ASoC: add Audio Graph Card2 driver ASoC: audio-graph-card2: add Multi CPU/Codec support ASoC: audio-graph-card2: add DPCM support ASoC: audio-graph-card2: add Codec2Codec support ASoC: add Audio Graph Card2 Yaml Document ASoC: add Audio Graph Card2 Custom Sample ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add Sample DT for Normal (Nulti) ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add DPCM sample (Multi) ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Single) ASoC: audio-graph-card2-custom-sample.dtsi: add Codec2Codec sample (Multi) ASoC: soc-pcm: tidyup soc_pcm_hw_clean() - step1 ASoC: soc-pcm: tidyup soc_pcm_hw_clean() - step2 ASoC: soc-component: add snd_soc_component_is_codec() ASoC: soc-core: tidyup empty function ASoC: soc-core: accept zero format at snd_soc_runtime_set_dai_fmt() Malik_Hsu (1): ASoC: Intel: sof_rt5682: Add support for max98360a speaker amp Marc Herbert (1): ASoC: SOF: prefix some terse and cryptic dev_dbg() with __func__ Mark Brown (93): Merge existing fixes from asoc/for-5.15 Merge series "Convert name-prefix doc to json-schema" from Sameer Pujar <spujar(a)nvidia.com>: Merge series "ARM: dts: Last round of DT schema fixes" from Maxime Ripard <maxime(a)cerno.tech>: Merge series "Cirrus Logic CS35L41 Amplifier" from David Rhodes <drhodes(a)opensource.cirrus.com>: Merge series "Patches to update for rockchip pdm" from Sugar Zhang <sugar.zhang(a)rock-chips.com>: Merge series "Support ALC5682I-VS codec" from Brent Lu <brent.lu(a)intel.com>: Merge series "ASoC: SOF: Intel: hda: Cleanups for local function uses" from Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com>: Merge series "ASoC: SOF: Remove unused members from struct sof_dev_desc" from Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com>: ASoC: atmel: Convert to new style DAI format definitions ASoC: au1x: Convert to modern terminology for DAI clocking Merge series "ASoC: SOF: Clean up the probe support" from Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com>: Merge series "ASoC: cs42l42: Implement Manual Type detection as fallback" from Vitaly Rodionov <vitalyr(a)opensource.cirrus.com>: ASoC: 88pm860x: Update to modern clocking terminology Merge series "ASoC: SOF: ipc: Small cleanups for message handler functions" from Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com>: ASoC: ab8500: Update to modern clocking terminology Merge series "Extend AHUB audio support for Tegra210 and later" from Sameer Pujar <spujar(a)nvidia.com>: Merge series "ASoC: compress: Support module_get on stream open" from Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com>: ASoC: Drop mistakenly applied SPI patch ASoC: cros_ec_codec: Use modern ASoC DAI format terminology ASoC: eureka-tlv320: Update to modern clocking terminology ASoC: fsl-asoc-card: Update to modern clocking terminology ASoC: fsl-audmix: Update to modern clocking terminology ASoC: fsl-esai: Update to modern clocking terminology ASoC: fsl-mqs: Update to modern clocking terminology ASoC: fsl_sai: Update to modern clocking terminology ASoC: fsl_ssi: Update to modern clocking terminology ASoC: imx-audmix: Update to modern clocking terminology ASoC: imx-card: Update to modern clocking terminology ASoC: imx-es8328: Update to modern clocking terminology ASoC: imx-hdmi: Update to modern clocking terminology ASoC: imx-rpmsg: Update to modern clocking terminology ASoC: imx-sgtl5000: Update to modern clocking terminology ASoC: mpc8610_hpcd: Update to modern clocking terminology ASoC: pl1022_ds: Update to modern clocking terminology ASoC: pl1022_rdk: Update to modern clocking terminology ASoC: zl38060: Update to modern clocking terminology ASoC: ak4118: Update to modern clocking terminology ASoC: ak4104: Update to modern clocking terminology ASoC: alc5632: Use modern ASoC DAI format terminology ASoC: ak4642: Use modern ASoC DAI format terminology ASoC: ep93xx: Convert to modern clocking terminology ASoC: ad1836: Update to modern clocking terminology ASoC: adau1372: Update to modern clocking terminology ASoC: adau1373: Update to modern clocking terminology ASoC: adau1701: Update to modern clocking terminology ASoC: adau17x1: Update to modern clocking terminology ASoC: adau1977: Update to modern clocking terminology ASoC: adav80x: Update to modern clocking terminology ASoC: ad193x: Update to modern clocking terminology ASoC: cpcap: Use modern ASoC DAI format terminology ASoC: bcm: Convert to modern clocking terminology ASoC: alc5623: Use modern ASoC DAI format terminology ASoC: ak4671: Use modern ASoC DAI format terminology ASoC: dwc-i2s: Update to modern clocking terminology ASoC: ak5558: Use modern ASoC DAI format terminology ASoC: ak4458: Use modern ASoC DAI format terminology Merge series "add driver to support firmware loading on Cirrus Logic DSPs" from Simon Trimmer <simont(a)opensource.cirrus.com>: Merge series "Add support for on demand pipeline setup/destroy" from Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com>: Merge series "ASoC: SOF: topology: minor updates" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: SOF: Intel: add flags to turn on SSP clocks early" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "ASoC: Intel: machine driver updates for 5.16" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Rockchip I2S/TDM controller" from Nicolas Frattaroli <frattaroli.nicolas(a)gmail.com>: Merge series "ASoC: SOF: Improvements for debugging" from Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com>: Merge series "Introduce new SOF helpers" from Daniel Baluta <daniel.baluta(a)oss.nxp.com> Daniel Baluta <daniel.baluta(a)nxp.com>: Merge series "Add reset-gpios handling for max98927" from Alejandro Tafalla <atafalla(a)dnyon.com>: Merge series "ASoC: Intel: bytcht_es8316: few cleanups" from Andy Shevchenko <andriy.shevchenko(a)linux.intel.com>: Merge series "ASoC: rt9120: Add Richtek RT9120 supprot" from cy_huang <u0084500(a)gmail.com> ChiYuan Huang <cy_huang(a)richtek.com>: sh: Use modern ASoC DAI format terminology Merge series "ASoC: Intel: bytcr_rt5651: few cleanups" from Andy Shevchenko <andriy.shevchenko(a)linux.intel.com>: ASoC: rt9120: Drop rt9210 audio amplifier support Merge series "ASoC: rt9120: Add Richtek RT9120 supprot" from cy_huang <u0084500(a)gmail.com> ChiYuan Huang <cy_huang(a)richtek.com>: Merge series "ASoC: Add Audio Graph Card2 support" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge tag 'v5.15-rc6' into asoc-5.16 Merge series "ASoC: cleanup / tidyup soc-pcm/core/component" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Merge branch 'asoc-5.15' into asoc-5.16 Merge series "ASoC: qcom: sm8250: add support for TX and RX Macro dais" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge branch 'asoc-5.15' into asoc-5.16 Merge series "ASoC: meson: axg: fix TDM channel order sync" from Jerome Brunet <jbrunet(a)baylibre.com>: Merge series "Add Yellow Carp platform ASoC driver" from Vijendar Mukunda <Vijendar.Mukunda(a)amd.com>: Merge series "ASoC: Add common modules support for ACP hw block" from Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com>: Merge series "ASoC: wm8962: Conversion to json-schema and fix" from Geert Uytterhoeven <geert+renesas(a)glider.be>: ASoC: topology: Fix stub for snd_soc_tplg_component_remove() Merge tag '20210927135559.738-6-srinivas.kandagatla(a)linaro.org' of https://git.kernel.org/pub/scm/linux/kernel/git/qcom/linux into v11_20211026_srinivas_kandagatla_asoc_qcom_add_audioreach_support for audioreach support Merge series "ASoC: qcom: Add AudioReach support" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "Update Lpass digital codec macro drivers" from Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org>: Merge series "ASoC: cs42l42: Fixes to power-down" from Richard Fitzgerald <rf(a)opensource.cirrus.com>: Merge series "ASoC: minor cleanup of warnings" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: Merge series "Make genaral and simple for new sof machine driver" from David Lin <CTLIN0(a)nuvoton.com>: Merge series "ASoC: cs42l42: Fix definition and handling of jack switch invert" from Richard Fitzgerald <rf(a)opensource.cirrus.com>: Merge series "ASoC: Sanity checks and soc-topology updates" from Cezary Rojewski <cezary.rojewski(a)intel.com>: Merge series "Multiple headphone codec driver support" from Brent Lu <brent.lu(a)intel.com>: Merge remote-tracking branch 'asoc/for-5.15' into asoc-linus Merge remote-tracking branch 'asoc/for-5.16' into asoc-next Masahiro Yamada (1): ASoC: ti: rename CONFIG_SND_SOC_DM365_VOICE_CODEC_MODULE Masanari Iida (1): ASoC: q6afe: q6asm: Fix typos in qcom,q6afe.txt and qcom,q6asm.txt Maxime Ripard (4): ASoC: dt-bindings: Add WM8978 Binding ASoC: dt-bindings: Convert Bluetooth SCO Link binding to a schema ASoC: dt-bindings: Convert SPDIF Transmitter binding to a schema ASoC: dt-bindings: Convert Simple Amplifier binding to a schema Nathan Chancellor (1): ASoC: qdsp6: audioreach: Fix clang -Wimplicit-fallthrough Nicolas Frattaroli (5): ASoC: rockchip: add support for i2s-tdm controller ASoC: dt-bindings: rockchip: add i2s-tdm bindings ASoC: rockchip: i2s-tdm: Strip out direct CRU use ASoC: dt-bindings: rockchip: i2s-tdm: Drop rockchip,cru property ASoC: rockchip: i2s-tdm: Fix refcount test Peter Ujfalusi (56): ASoC: soc-topology: Move template info print soc_tplg_dapm_widget_create() ASoC: SOF: Handle control change notification from firmware ASoC: SOF: intel: Do no initialize resindex_dma_base ASoC: SOF: Drop resindex_dma_base, dma_engine, dma_size from sof_dev_desc ASoC: SOF: Intel: hda-dsp: Declare locally used functions as static ASoC: SOF: Intel: hda: Remove boot_firmware skl and iccmax_icl declarations ASoC: SOF: Intel: hda: Relocate inline definitions from hda.h to hda.c for sdw ASoC: SOF: loader: load_firmware callback is mandatory, treat it like that ASoC: SOF: Intel: bdw: Set the mailbox offset directly in bdw_probe ASoC: SOF: ipc: Remove snd_sof_dsp_mailbox_init() ASoC: SOF: imx: Do not initialize the snd_sof_dsp_ops.read64 ASoC: SOF: loader: No need to export snd_sof_fw_parse_ext_data() ASoC: SOF: core: Do not use 'bar' as parameter for block_read/write ASoC: SOF: debug: Add generic API and ops for DSP regions ASoC: SOF: imx: Provide debugfs_add_region_item ops for core ASoC: SOF: Intel: Provide debugfs_add_region_item ops for core ASoC: SOF: loader: Use the generic ops for region debugfs handling ASoC: SOF: debug: No need to export the snd_sof_debugfs_io_item() ASoC: SOF: ipc: Add probe message logging to ipc_log_header() ASoC: SOF: pcm: Remove non existent CONFIG_SND_SOC_SOF_COMPRESS reference ASoC: SOF: probe: Merge and clean up the probe and compress files ASoC: SOF: Intel: Rename hda-compress.c to hda-probes.c ASoC: SOF: sof-probes: Correct the function names used for snd_soc_cdai_ops ASoC: SOF: core: Move probe work related code under a single if () branch ASoC: SOF: ipc: Clarify the parameter name for ipc_trace_message() ASoC: SOF: ipc: Print 0x prefix for errors in ipc_trace/stream_message() ASoC: SOF: ipc: Remove redundant error check from sof_ipc_tx_message_unlocked ASoC: SOF: Rename sof_arch_ops to dsp_arch_ops ASoC: soc-component: Convert the mark_module to void* ASoC: compress/component: Use module_get_when_open/put_when_close for cstream ASoC: SOF: Remove struct sof_ops_table and sof_get_ops() macro ASoC: Intel: boards: Update to modern clocking terminology ASoC: SOF: Intel: hda-stream: Print stream name on STREAM_SD_OFFSET timeout ASoC: SOF: ipc: Make the error prints consistent in tx_wait_done() ASoC: SOF: Change SND_SOC_SOF_TOPLEVEL from config to menuconfig ASoC: SOF: debug: Swap the dsp_dump and ipc_dump sequence for fw_exception ASoC: SOF: ipc and dsp dump: Add markers for better visibility ASoC: SOF: Print the dbg_dump and ipc_dump once to reduce kernel log noise ASoC: SOF: loader: Print the DSP dump if boot fails ASoC: SOF: intel: atom: No need to do a DSP dump in atom_run() ASoC: SOF: debug/ops: Move the IPC and DSP dump functions out from the header ASoC: SOF: debug: Add SOF_DBG_DUMP_OPTIONAL flag for DSP dumping ASoC: SOF: intel: hda-loader: Use snd_sof_dsp_dbg_dump() for DSP dump ASoC: SOF: Drop SOF_DBG_DUMP_FORCE_ERR_LEVEL and sof_dev_dbg_or_err ASoC: SOF: debug: Print out the fw_state along with the DSP dump ASoC: SOF: ipc: Re-enable dumps after successful IPC tx ASoC: SOF: ops: Force DSP panic dumps to be printed ASoC: SOF: Introduce macro to set the firmware state ASoC: SOF: intel: hda: Drop 'error' prefix from error dump functions ASoC: SOF: core: Clean up snd_sof_get_status() prints ASoC: SOF: loader: Drop SOF_DBG_DUMP_REGS flag when firmware start fails ASoC: SOF: Intel: hda-loader: Drop SOF_DBG_DUMP_REGS flag from dbg_dump calls ASoC: SOF: Intel: hda: Dump registers and stack when SOF_DBG_DUMP_REGS is set ASoC: SOF: pipelines: Harmonize all functions to use struct snd_sof_dev ASoC: rt1011: Fix 'I2S Reference' enum control caused error ASoC: rt1011: Fix 'I2S Reference' enum control Pierre-Louis Bossart (29): ASoC: amd: acp: declare and add prefix to 'bt_uart_enable' symbol ASoC: SOF: core: allow module parameter to override dma trace Kconfig ASoC: SOF: imx: add header file for ops ASoC: Intel: soc-acpi: apl/glk/tgl: add entry for devices based on ES8336 codec ALSA: intel-dsp-config: add quirk for APL/GLK/TGL devices based on ES8336 codec ASoC: Intel: add machine driver for SOF+ES8336 ASoC: Intel: soc-acpi: add missing quirk for TGL SDCA single amp ASoC: Intel: sof_sdw: add missing quirk for Dell SKU 0A45 ASoC: SOF: dai: mirror group_id definition added in firmware ASoC: SOF: dai: include new flags for DAI_CONFIG ASoC: SOF: Intel: hda: add new flags for DAI_CONFIG ASoC: SOF: Intel: hda-dai: improve SSP DAI handling for dynamic pipelines ASoC: SOF: topology: show clks_control value in dynamic debug ASoC: SOF: topology: allow for dynamic pipelines override for debug ASoC: SOF: core: debug: force all processing on primary core ASoC: soc-pcm: restore mixer functionality ASoC: topology: handle endianness warning ASoC: rt5682s: use 'static' qualifier ASoC: nau8821: fix kernel-doc ASoC: nau8821: clarify out-of-bounds check ASoC: mediatek: remove unnecessary initialization ASoC: mediatek: mt8195: rename shadowed array ASoC: mediatek: mt8195: fix return value ASoC: rockchip: i2s_tdm: improve return value handling ASoC: Intel: soc-acpi: add entry for ESSX8336 on JSL ASoC: es8316: add support for ESSX8336 ACPI _HID ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs ASoC: Intel: soc-acpi: use const for all uses of snd_soc_acpi_codecs Rander Wang (2): ASoC: SOF: control: fix a typo in put operations for kcontrol ASoC: SOF: prepare code to allocate IPC messages in fw_ready Ranjani Sridharan (16): ASoC: SOF: compress: move and export sof_probe_compr_ops ASoC: SOF: pm: fix a stale comment ASoC: topology: change the complete op in snd_soc_tplg_ops to return int ASoC: SOF: control: Add access field in struct snd_sof_control ASoC: SOF: topology: Add new token for dynamic pipeline ASoC: SOF: sof-audio: add helpers for widgets, kcontrols and dai config set up AsoC: dapm: export a couple of functions ASoC: SOF: Add new fields to snd_sof_route ASoC: SOF: restore kcontrols for widget during set up ASoC: SOF: Don't set up widgets during topology parsing ASoC: SOF: Introduce widget use_count ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC ASoC: SOF: Add support for dynamic pipelines ASoC: SOF: topology: Add kernel parameter for topology verification ASoC: SOF: topology: return error if sof_connect_dai_widget() fails ASoC: SOF: topology: do not power down primary core during topology removal Richard Fitzgerald (18): ASoC: cs42l42: Don't reconfigure the PLL while it is running ASoC: cs42l42: Always configure both ASP TX channels ASoC: cs42l42: Correct some register default values ASoC: cs42l42: Don't set defaults for volatile registers ASoC: cs42l42: Defer probe if request_threaded_irq() returns EPROBE_DEFER ASoC: cs42l42: Don't claim to support 192k ASoC: cs42l42: Use PLL for SCLK > 12.288MHz ASoC: cs42l42: Allow time for HP/ADC to power-up after enable ASoC: cs42l42: Set correct SRC MCLK ASoC: cs42l42: Mark OSC_SWITCH_STATUS register volatile ASoC: cs42l42: Fix WARN in remove() if running without an interrupt ASoC: cs42l42: Always enable TS_PLUG and TS_UNPLUG interrupts ASoC: cs42l42: Remove unused runtime_suspend/runtime_resume callbacks ASoC: cs42l42: Prevent NULL pointer deref in interrupt handler ASoC: cs42l42: Reset and power-down on remove() and failed probe() ASoC: cs42l42: free_irq() before powering-down on probe() fail ASoC: dt-bindings: cs42l42: Correct description of ts-inv ASoC: cs42l42: Correct configuring of switch inversion from ts-inv Rikard Falkeborn (9): ASoC: fsl: Constify static snd_soc_ops ASoC: tegra: Constify static snd_soc_dai_ops structs ASoC: ti: Constify static snd_soc_ops ASoC: ux500: mop500: Constify static snd_soc_ops ASoC: tegra: Constify static snd_soc_ops ASoC: qcom: apq8096: Constify static snd_soc_ops ASoC: amd: acp-rt5645: Constify static snd_soc_ops ASoC: mediatek: Constify static snd_soc_ops ASoC: amd: vangogh: constify static struct snd_soc_dai_ops Sameer Pujar (14): ASoC: Add json-schema documentation for sound-name-prefix ASoC: Use schema reference for sound-name-prefix ASoC: Remove name-prefix.txt ASoC: soc-pcm: Don't reconnect an already active BE ASoC: simple-card-utils: Increase maximum DAI links limit to 512 ASoC: audio-graph: Fixup CPU endpoint hw_params in a BE<->BE link ASoC: dt-bindings: tegra: Few more Tegra210 AHUB modules ASoC: tegra: Add routes for few AHUB modules ASoC: tegra: Add Tegra210 based MVC driver ASoC: tegra: Add Tegra210 based SFC driver ASoC: tegra: Add Tegra210 based AMX driver ASoC: tegra: Add Tegra210 based ADX driver ASoC: tegra: Add Tegra210 based Mixer driver ASoC: Fix warning related to 'sound-name-prefix' binding Sebastian Andrzej Siewior (1): ASoC: mediatek: mt8195: Remove unsued irqs_lock. Seven Lee (1): ASoC: nau8821: new driver Shengjiu Wang (3): ASoC: fsl_rpmsg: add soc specific data structure ASoC: fsl_spdif: Add support for i.MX8ULP ASoC: fsl_rpmsg: Add rpmsg audio support for i.MX8ULP Simon Trimmer (14): ASoC: wm_adsp: Remove use of snd_ctl_elem_type_t ASoC: wm_adsp: Cancel ongoing work when removing controls ASoC: wm_adsp: Rename generic DSP support ASoC: wm_adsp: Introduce cs_dsp logging macros ASoC: wm_adsp: Separate some ASoC and generic functions ASoC: wm_adsp: Split DSP power operations into helper functions ASoC: wm_adsp: Separate generic cs_dsp_coeff_ctl handling ASoC: wm_adsp: Move check of dsp->running to better place ASoC: wm_adsp: Pass firmware names as parameters when starting DSP core ASoC: wm_adsp: move firmware loading to client ASoC: wm_adsp: Split out struct cs_dsp from struct wm_adsp ASoC: wm_adsp: Separate wm_adsp specifics in cs_dsp_client_ops firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs ASoC: soc-component: Remove conditional definition of debugfs data members Srinivas Kandagatla (25): soc: dt-bindings: qcom: apr: convert to yaml soc: dt-bindings: qcom: apr: deprecate qcom,apr-domain property soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support ASoC: soc-component: improve error reporting for register access ASoC: qcom: sm8250: add support for TX and RX Macro dais ASoC: qcom: sm8250: Add Jack support ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks Srinivasa Rao Mandadapu (7): ASoC: dt-bindings: lpass: add binding headers for digital codecs ASoC: dt-bindings: lpass: add binding headers for digital codecs ASoC: qcom: Add compatible names in va,wsa,rx,tx codec drivers for sc7280 ASoC: qcom: dt-bindings: Add compatible names for lpass sc7280 digital codecs ASoC: codecs: tx-macro: Enable tx top soundwire mic clock ASoC: codecs: tx-macro: Update tx default values ASoC: codecs: Change bulk clock voting to optional voting in digital codecs Stefan Binding (2): ASoC: cs42l42: Implement Manual Type detection as fallback ASoC: cs42l42: Use two thresholds and increased wait time for manual type detection Stephan Gerhold (1): ASoC: qcom: common: Respect status = "disabled" on DAI link nodes Sugar Zhang (9): ASoC: rockchip: Add support for rv1126 pdm ASoC: dt-bindings: rockchip: Add binding for rv1126 pdm ASoC: rockchip: pdm: Add support for rk3568 pdm ASoC: dt-bindings: rockchip: Add binding for rk3568 pdm ASoC: rockchip: pdm: Add support for path map ASoC: dt-bindings: rockchip: pdm: Document property 'rockchip,path-map' ASoC: dt-bindings: rockchip: Convert pdm bindings to yaml ASoC: dmaengine: Introduce module option prealloc_buffer_size_kbytes ASoC: rockchip: Use generic dmaengine code Trevor Wu (7): ASoC: mt8195: remove unnecessary CONFIG_PM ASoC: mediatek: mt6359: Fix unexpected error in bind/unbind flow ASoC: mediatek: mt8195: add missing of_node_put in probe ASoC: mediatek: mt8195: move of_node_put to remove function ASoC: mediatek: mt8195: update audsys clock parent name ASoC: mediatek: mt8195: add machine driver with mt6359, rt1011 and rt5682 ASoC: mediatek: mt8195: add mt8195-mt6359-rt1011-rt5682 bindings document Tzung-Bi Shih (2): ASoC: dt-bindings: mediatek: mt8192: re-add audio afe document ASoC: dt-bindings: mediatek: rename reset controller headers in DT example Uwe Kleine-König (5): ASoC: tlv320aic32x4: Make aic32x4_remove() return void ASoc: wm8731: Drop empty spi_driver remove callback ASoc: wm8900: Drop empty spi_driver remove callback ASoC: cs35l41: Make cs35l41_remove() return void ASoC: tlv320aic3x: Make aic3x_remove() return void Vijendar Mukunda (16): ASoc: amd: create platform device for VG machine driver ASoC: amd: add vangogh machine driver ASoC: amd: enable vangogh platform machine driver build ASoC: amd: add Yellow Carp ACP6x IP register header ASoC: amd: add Yellow Carp ACP PCI driver ASoC: amd: add acp6x init/de-init functions ASoC: amd: add platform devices for acp6x pdm driver and dmic driver ASoC: amd: add acp6x pdm platform driver ASoC: amd: add acp6x irq handler ASoC: amd: add acp6x pdm driver dma ops ASoC: amd: add acp6x pci driver pm ops ASoC: amd: add acp6x pdm driver pm ops ASoC: amd: enable Yellow carp acp6x drivers build ASoC: amd: create platform device for acp6x machine driver ASoC: amd: add YC machine driver using dmic ASoC: amd: enable Yellow Carp platform machine driver build Vincent Knecht (1): ASoC: codecs: tfa989x: Add support for tfa9897 RCV bit Viorel Suman (2): ASoC: SOF: imx8m: add SAI1 info ASoC: fsl_spdif: implement bypass mode from in to out Vitaly Rodionov (1): ASoC: cs42l42: Minor fix all errors reported by checkpatch.pl script Wolfram Sang (1): ASoC: codecs: max98390: simplify getting the adapter of a client Yang Yingliang (1): ASoC: amd: acp: Fix return value check in acp_machine_select() Yassine Oudjana (1): ASoC: wcd9335: Use correct version to initialize Class H gearhead (1): ASoC: pcm5102a: increase rate from 192k to 384k .../devicetree/bindings/soc/qcom/qcom,apr.txt | 134 - .../devicetree/bindings/soc/qcom/qcom,apr.yaml | 177 + .../devicetree/bindings/sound/amlogic,t9015.yaml | 6 + .../bindings/sound/audio-graph-card2.yaml | 57 + Documentation/devicetree/bindings/sound/bt-sco.txt | 13 - .../devicetree/bindings/sound/cirrus,cs35l41.yaml | 157 + .../devicetree/bindings/sound/cs42l42.txt | 13 +- .../devicetree/bindings/sound/linux,bt-sco.yaml | 38 + .../devicetree/bindings/sound/linux,spdif-dit.yaml | 32 + .../devicetree/bindings/sound/max9892x.txt | 3 + .../devicetree/bindings/sound/maxim,max98520.yaml | 36 + .../devicetree/bindings/sound/mt8192-afe-pcm.yaml | 100 + .../sound/mt8195-mt6359-rt1011-rt5682.yaml | 47 + .../devicetree/bindings/sound/name-prefix.txt | 24 - .../devicetree/bindings/sound/name-prefix.yaml | 21 + .../devicetree/bindings/sound/nau8821.txt | 55 + .../bindings/sound/nvidia,tegra186-dspk.yaml | 9 +- .../bindings/sound/nvidia,tegra210-adx.yaml | 76 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 20 + .../bindings/sound/nvidia,tegra210-amx.yaml | 76 + .../bindings/sound/nvidia,tegra210-dmic.yaml | 9 +- .../bindings/sound/nvidia,tegra210-i2s.yaml | 9 +- .../bindings/sound/nvidia,tegra210-mixer.yaml | 74 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 76 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 73 + .../devicetree/bindings/sound/nxp,tfa989x.yaml | 9 +- .../bindings/sound/qcom,lpass-rx-macro.yaml | 4 +- .../bindings/sound/qcom,lpass-tx-macro.yaml | 4 +- .../bindings/sound/qcom,lpass-va-macro.yaml | 4 +- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 4 +- .../devicetree/bindings/sound/qcom,q6afe.txt | 181 - .../devicetree/bindings/sound/qcom,q6apm-dai.yaml | 53 + .../devicetree/bindings/sound/qcom,q6asm.txt | 2 +- .../bindings/sound/qcom,q6dsp-lpass-clocks.yaml | 77 + .../bindings/sound/qcom,q6dsp-lpass-ports.yaml | 205 ++ .../devicetree/bindings/sound/realtek,rt5682s.yaml | 117 + .../devicetree/bindings/sound/richtek,rt9120.yaml | 59 + .../bindings/sound/rockchip,i2s-tdm.yaml | 182 + .../devicetree/bindings/sound/rockchip,pdm.txt | 46 - .../devicetree/bindings/sound/rockchip,pdm.yaml | 120 + Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- .../devicetree/bindings/sound/simple-amplifier.txt | 17 - .../bindings/sound/simple-audio-amplifier.yaml | 45 + .../bindings/sound/simple-audio-mux.yaml | 9 +- .../bindings/sound/socionext,uniphier-aio.yaml | 22 +- .../bindings/sound/socionext,uniphier-evea.yaml | 6 +- .../bindings/sound/spdif-transmitter.txt | 10 - .../devicetree/bindings/sound/test-component.yaml | 33 + .../devicetree/bindings/sound/wlf,wm8962.yaml | 118 + .../devicetree/bindings/sound/wlf,wm8978.yaml | 58 + Documentation/devicetree/bindings/sound/wm8962.txt | 43 - Documentation/sound/alsa-configuration.rst | 9 + Documentation/sound/soc/codec.rst | 8 +- MAINTAINERS | 18 + arch/sh/boards/mach-ecovec24/setup.c | 2 +- arch/sh/boards/mach-se/7724/setup.c | 2 +- drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/cirrus/Kconfig | 5 + drivers/firmware/cirrus/Makefile | 3 + drivers/firmware/cirrus/cs_dsp.c | 3109 +++++++++++++++++ drivers/soc/qcom/Kconfig | 2 +- drivers/soc/qcom/apr.c | 287 +- include/dt-bindings/soc/qcom,gpr.h | 19 + include/dt-bindings/sound/qcom,lpass.h | 31 + include/dt-bindings/sound/qcom,q6afe.h | 203 +- include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h | 208 ++ include/linux/firmware/cirrus/cs_dsp.h | 242 ++ .../linux/firmware/cirrus}/wmfw.h | 8 +- include/linux/soc/qcom/apr.h | 70 +- include/sound/cs35l41.h | 34 + include/sound/graph_card.h | 21 + include/sound/rt5682s.h | 48 + include/sound/simple_card_utils.h | 6 +- include/sound/soc-acpi.h | 3 + include/sound/soc-component.h | 21 +- include/sound/soc-dpcm.h | 1 + include/sound/soc-topology.h | 5 +- include/sound/sof.h | 5 - include/sound/sof/dai-intel.h | 4 + include/sound/sof/dai.h | 10 +- include/uapi/sound/snd_ar_tokens.h | 208 ++ include/uapi/sound/sof/tokens.h | 1 + sound/hda/intel-dsp-config.c | 22 +- sound/soc/amd/Kconfig | 40 +- sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-da7219-max98357a.c | 6 +- sound/soc/amd/acp-pcm-dma.c | 6 +- sound/soc/amd/acp-rt5645.c | 2 +- sound/soc/amd/acp.h | 2 + sound/soc/amd/acp/Kconfig | 61 + sound/soc/amd/acp/Makefile | 26 + sound/soc/amd/acp/acp-i2s.c | 340 ++ sound/soc/amd/acp/acp-legacy-mach.c | 104 + sound/soc/amd/acp/acp-mach-common.c | 600 ++++ sound/soc/amd/acp/acp-mach.h | 57 + sound/soc/amd/acp/acp-platform.c | 315 ++ sound/soc/amd/acp/acp-renoir.c | 163 + sound/soc/amd/acp/acp-sof-mach.c | 131 + sound/soc/amd/acp/amd.h | 145 + sound/soc/amd/acp/chip_offset_byte.h | 76 + sound/soc/amd/vangogh/Makefile | 2 + sound/soc/amd/vangogh/acp5x-i2s.c | 2 +- sound/soc/amd/vangogh/acp5x-mach.c | 386 +++ sound/soc/amd/vangogh/acp5x.h | 2 +- sound/soc/amd/vangogh/pci-acp5x.c | 3 + sound/soc/amd/yc/Makefile | 9 + sound/soc/amd/yc/acp6x-mach.c | 194 ++ sound/soc/amd/yc/acp6x-pdm-dma.c | 448 +++ sound/soc/amd/yc/acp6x.h | 107 + sound/soc/amd/yc/acp6x_chip_offset_byte.h | 444 +++ sound/soc/amd/yc/pci-acp6x.c | 338 ++ sound/soc/atmel/atmel-i2s.c | 6 +- sound/soc/atmel/atmel_ssc_dai.c | 26 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 10 +- sound/soc/atmel/mikroe-proto.c | 4 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- sound/soc/au1x/db1200.c | 6 +- sound/soc/au1x/i2sc.c | 6 +- sound/soc/au1x/psc-i2s.c | 10 +- sound/soc/bcm/bcm2835-i2s.c | 56 +- sound/soc/bcm/cygnus-ssp.c | 6 +- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 10 +- sound/soc/cirrus/snappercl15.c | 2 +- sound/soc/codecs/88pm860x-codec.c | 18 +- sound/soc/codecs/Kconfig | 46 + sound/soc/codecs/Makefile | 12 + sound/soc/codecs/ab8500-codec.c | 20 +- sound/soc/codecs/ad1836.c | 6 +- sound/soc/codecs/ad193x.c | 10 +- sound/soc/codecs/adau1372.c | 14 +- sound/soc/codecs/adau1373.c | 14 +- sound/soc/codecs/adau1701.c | 6 +- sound/soc/codecs/adau17x1.c | 6 +- sound/soc/codecs/adau1977.c | 31 +- sound/soc/codecs/adav80x.c | 6 +- sound/soc/codecs/ak4104.c | 4 +- sound/soc/codecs/ak4118.c | 20 +- sound/soc/codecs/ak4458.c | 12 +- sound/soc/codecs/ak4642.c | 8 +- sound/soc/codecs/ak4671.c | 6 +- sound/soc/codecs/ak5558.c | 10 +- sound/soc/codecs/alc5623.c | 8 +- sound/soc/codecs/alc5632.c | 8 +- sound/soc/codecs/cpcap.c | 18 +- sound/soc/codecs/cros_ec_codec.c | 4 +- sound/soc/codecs/cs35l41-i2c.c | 115 + sound/soc/codecs/cs35l41-spi.c | 140 + sound/soc/codecs/cs35l41-tables.c | 594 ++++ sound/soc/codecs/cs35l41.c | 1445 ++++++++ sound/soc/codecs/cs35l41.h | 775 +++++ sound/soc/codecs/cs42l42.c | 481 +-- sound/soc/codecs/cs42l42.h | 64 +- sound/soc/codecs/cs4341.c | 7 + sound/soc/codecs/cs47l15.c | 26 +- sound/soc/codecs/cs47l24.c | 20 +- sound/soc/codecs/cs47l35.c | 26 +- sound/soc/codecs/cs47l85.c | 34 +- sound/soc/codecs/cs47l90.c | 36 +- sound/soc/codecs/cs47l92.c | 20 +- sound/soc/codecs/es8316.c | 8 +- sound/soc/codecs/lpass-rx-macro.c | 3 +- sound/soc/codecs/lpass-tx-macro.c | 25 +- sound/soc/codecs/lpass-va-macro.c | 3 +- sound/soc/codecs/lpass-wsa-macro.c | 1 + sound/soc/codecs/madera.c | 18 +- sound/soc/codecs/max98390.c | 2 +- sound/soc/codecs/max98520.c | 769 +++++ sound/soc/codecs/max98520.h | 159 + sound/soc/codecs/max98927.c | 25 + sound/soc/codecs/max98927.h | 1 + sound/soc/codecs/mt6359.c | 2 +- sound/soc/codecs/nau8821.c | 1714 ++++++++++ sound/soc/codecs/nau8821.h | 533 +++ sound/soc/codecs/nau8824.c | 74 +- sound/soc/codecs/nau8824.h | 3 +- sound/soc/codecs/nau8825.c | 48 +- sound/soc/codecs/pcm179x-spi.c | 1 + sound/soc/codecs/pcm5102a.c | 2 +- sound/soc/codecs/pcm512x.c | 2 + sound/soc/codecs/rt1011.c | 10 + sound/soc/codecs/rt1015.c | 2 +- sound/soc/codecs/rt1016.c | 2 +- sound/soc/codecs/rt1019.c | 2 +- sound/soc/codecs/rt1305.c | 2 +- sound/soc/codecs/rt1308.c | 2 +- sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 9 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5660.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/rt5682-i2c.c | 17 + sound/soc/codecs/rt5682.c | 132 +- sound/soc/codecs/rt5682.h | 23 + sound/soc/codecs/rt5682s.c | 3197 ++++++++++++++++++ sound/soc/codecs/rt5682s.h | 1474 ++++++++ sound/soc/codecs/rt9120.c | 495 +++ sound/soc/codecs/tfa989x.c | 21 + sound/soc/codecs/tlv320aic32x4-i2c.c | 4 +- sound/soc/codecs/tlv320aic32x4-spi.c | 4 +- sound/soc/codecs/tlv320aic32x4.c | 4 +- sound/soc/codecs/tlv320aic32x4.h | 2 +- sound/soc/codecs/tlv320aic3x-i2c.c | 4 +- sound/soc/codecs/tlv320aic3x-spi.c | 4 +- sound/soc/codecs/tlv320aic3x.c | 3 +- sound/soc/codecs/tlv320aic3x.h | 2 +- sound/soc/codecs/wcd9335.c | 2 +- sound/soc/codecs/wcd938x.c | 6 +- sound/soc/codecs/wm2200.c | 30 +- sound/soc/codecs/wm5102.c | 16 +- sound/soc/codecs/wm5110.c | 24 +- sound/soc/codecs/wm8731.c | 6 - sound/soc/codecs/wm8900.c | 6 - sound/soc/codecs/wm8960.c | 13 +- sound/soc/codecs/wm8962.c | 13 +- sound/soc/codecs/wm_adsp.c | 3303 ++---------------- sound/soc/codecs/wm_adsp.h | 105 +- sound/soc/codecs/zl38060.c | 4 +- sound/soc/dwc/dwc-i2s.c | 12 +- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 54 +- sound/soc/fsl/fsl_audmix.c | 8 +- sound/soc/fsl/fsl_esai.c | 28 +- sound/soc/fsl/fsl_mqs.c | 4 +- sound/soc/fsl/fsl_rpmsg.c | 47 +- sound/soc/fsl/fsl_rpmsg.h | 12 + sound/soc/fsl/fsl_sai.c | 34 +- sound/soc/fsl/fsl_sai.h | 2 +- sound/soc/fsl/fsl_spdif.c | 85 + sound/soc/fsl/fsl_ssi.c | 38 +- sound/soc/fsl/fsl_xcvr.c | 17 +- sound/soc/fsl/imx-audmix.c | 12 +- sound/soc/fsl/imx-card.c | 6 +- sound/soc/fsl/imx-es8328.c | 2 +- sound/soc/fsl/imx-hdmi.c | 6 +- sound/soc/fsl/imx-rpmsg.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 16 +- sound/soc/fsl/p1022_ds.c | 16 +- sound/soc/fsl/p1022_rdk.c | 2 +- sound/soc/generic/Kconfig | 20 + sound/soc/generic/Makefile | 6 + sound/soc/generic/audio-graph-card.c | 4 +- .../soc/generic/audio-graph-card2-custom-sample.c | 183 + .../generic/audio-graph-card2-custom-sample.dtsi | 227 ++ sound/soc/generic/audio-graph-card2.c | 1281 +++++++ sound/soc/generic/simple-card-utils.c | 50 +- sound/soc/generic/test-component.c | 659 ++++ sound/soc/intel/boards/Kconfig | 18 +- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 4 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 4 +- sound/soc/intel/boards/bytcht_da7213.c | 4 +- sound/soc/intel/boards/bytcht_es8316.c | 41 +- sound/soc/intel/boards/bytcht_nocodec.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 120 +- sound/soc/intel/boards/bytcr_rt5651.c | 122 +- sound/soc/intel/boards/bytcr_wm5102.c | 4 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 56 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 4 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 6 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 4 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 4 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_es8336.c | 569 ++++ sound/soc/intel/boards/sof_rt5682.c | 96 +- sound/soc/intel/boards/sof_sdw.c | 10 + sound/soc/intel/common/soc-acpi-intel-adl-match.c | 24 +- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 8 +- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 68 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 69 +- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 8 +- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 17 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 27 +- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 12 +- sound/soc/intel/common/soc-acpi-intel-skl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 60 +- sound/soc/intel/skylake/skl-topology.c | 6 +- sound/soc/mediatek/Kconfig | 23 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 3 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 2 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- sound/soc/mediatek/mt8195/Makefile | 1 + sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 10 +- sound/soc/mediatek/mt8195/mt8195-audsys-clk.c | 152 +- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 8 +- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 4 +- .../mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c | 1155 +++++++ .../mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 54 +- sound/soc/meson/aiu-encoder-spdif.c | 2 +- sound/soc/meson/axg-card.c | 1 + sound/soc/meson/axg-tdm-interface.c | 26 +- sound/soc/meson/meson-card-utils.c | 1 + sound/soc/meson/meson-codec-glue.c | 3 - sound/soc/qcom/Kconfig | 25 +- sound/soc/qcom/apq8096.c | 2 +- sound/soc/qcom/common.c | 4 +- sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1130 +++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 ++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 689 +--- sound/soc/qcom/qdsp6/q6apm-dai.c | 416 +++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++ sound/soc/qcom/qdsp6/q6apm.c | 822 +++++ sound/soc/qcom/qdsp6/q6apm.h | 152 + sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 + sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 ++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 + sound/soc/qcom/qdsp6/q6prm.c | 202 ++ sound/soc/qcom/qdsp6/q6prm.h | 78 + sound/soc/qcom/qdsp6/topology.c | 1113 ++++++ sound/soc/qcom/sm8250.c | 79 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 5 +- sound/soc/rockchip/rockchip_i2s.c | 3 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 1762 ++++++++++ sound/soc/rockchip/rockchip_i2s_tdm.h | 398 +++ sound/soc/rockchip/rockchip_pcm.c | 44 - sound/soc/rockchip/rockchip_pcm.h | 11 - sound/soc/rockchip/rockchip_pdm.c | 112 +- sound/soc/rockchip/rockchip_pdm.h | 6 + sound/soc/samsung/s3c-i2s-v2.c | 2 + sound/soc/sh/rcar/core.c | 1 + sound/soc/soc-acpi.c | 24 +- sound/soc/soc-component.c | 87 +- sound/soc/soc-compress.c | 43 +- sound/soc/soc-core.c | 55 +- sound/soc/soc-dapm.c | 15 +- sound/soc/soc-generic-dmaengine-pcm.c | 6 +- sound/soc/soc-pcm.c | 27 +- sound/soc/soc-topology.c | 52 +- sound/soc/soc-utils.c | 13 + sound/soc/sof/Kconfig | 6 +- sound/soc/sof/Makefile | 6 +- sound/soc/sof/compress.c | 158 +- sound/soc/sof/compress.h | 32 - sound/soc/sof/control.c | 192 +- sound/soc/sof/core.c | 35 +- sound/soc/sof/debug.c | 87 +- sound/soc/sof/imx/Kconfig | 2 + sound/soc/sof/imx/imx-ops.h | 10 + sound/soc/sof/imx/imx8.c | 47 +- sound/soc/sof/imx/imx8m.c | 41 +- sound/soc/sof/intel/Makefile | 5 +- sound/soc/sof/intel/apl.c | 7 +- sound/soc/sof/intel/atom.c | 5 +- sound/soc/sof/intel/bdw.c | 19 +- sound/soc/sof/intel/byt.c | 30 +- sound/soc/sof/intel/cnl.c | 7 +- sound/soc/sof/intel/hda-dai.c | 262 +- sound/soc/sof/intel/hda-dsp.c | 61 +- sound/soc/sof/intel/hda-ipc.c | 15 +- sound/soc/sof/intel/hda-loader.c | 11 +- .../soc/sof/intel/{hda-compress.c => hda-probes.c} | 0 sound/soc/sof/intel/hda-stream.c | 92 +- sound/soc/sof/intel/hda.c | 232 +- sound/soc/sof/intel/hda.h | 52 +- sound/soc/sof/intel/icl.c | 7 +- sound/soc/sof/intel/pci-apl.c | 2 - sound/soc/sof/intel/pci-cnl.c | 3 - sound/soc/sof/intel/pci-icl.c | 2 - sound/soc/sof/intel/pci-tgl.c | 5 - sound/soc/sof/intel/pci-tng.c | 16 +- sound/soc/sof/intel/tgl.c | 7 +- sound/soc/sof/ipc.c | 217 +- sound/soc/sof/loader.c | 161 +- sound/soc/sof/ops.c | 3 + sound/soc/sof/ops.h | 77 +- sound/soc/sof/pcm.c | 71 +- sound/soc/sof/pm.c | 12 +- sound/soc/sof/probe.h | 85 - sound/soc/sof/sof-audio.c | 715 +++- sound/soc/sof/sof-audio.h | 52 +- sound/soc/sof/sof-of-dev.c | 24 +- sound/soc/sof/sof-priv.h | 154 +- sound/soc/sof/{probe.c => sof-probes.c} | 280 +- sound/soc/sof/sof-probes.h | 38 + sound/soc/sof/{intel/intel-ipc.c => stream-ipc.c} | 56 +- sound/soc/sof/topology.c | 417 +-- sound/soc/sof/trace.c | 5 +- sound/soc/sof/utils.c | 28 +- sound/soc/sof/xtensa/core.c | 2 +- sound/soc/tegra/Kconfig | 48 + sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra210_adx.c | 531 +++ sound/soc/tegra/tegra210_adx.h | 72 + sound/soc/tegra/tegra210_ahub.c | 511 ++- sound/soc/tegra/tegra210_amx.c | 600 ++++ sound/soc/tegra/tegra210_amx.h | 93 + sound/soc/tegra/tegra210_mixer.c | 674 ++++ sound/soc/tegra/tegra210_mixer.h | 100 + sound/soc/tegra/tegra210_mvc.c | 645 ++++ sound/soc/tegra/tegra210_mvc.h | 117 + sound/soc/tegra/tegra210_sfc.c | 3549 ++++++++++++++++++++ sound/soc/tegra/tegra210_sfc.h | 78 + sound/soc/tegra/tegra_asoc_machine.c | 62 +- sound/soc/tegra/tegra_asoc_machine.h | 1 + sound/soc/ti/Kconfig | 2 +- sound/soc/ti/davinci-evm.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 2 +- sound/soc/ux500/mop500_ab8500.c | 2 +- sound/soc/ux500/mop500_ab8500.h | 2 +- 429 files changed, 46511 insertions(+), 7638 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml create mode 100644 Documentation/devicetree/bindings/sound/linux,bt-sco.yaml create mode 100644 Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98520.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1011-rt5682.yaml delete mode 100644 Documentation/devicetree/bindings/sound/name-prefix.txt create mode 100644 Documentation/devicetree/bindings/sound/name-prefix.yaml create mode 100644 Documentation/devicetree/bindings/sound/nau8821.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml create mode 100644 Documentation/devicetree/bindings/sound/richtek,rt9120.yaml create mode 100644 Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip,pdm.yaml delete mode 100644 Documentation/devicetree/bindings/sound/simple-amplifier.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml delete mode 100644 Documentation/devicetree/bindings/sound/spdif-transmitter.txt create mode 100644 Documentation/devicetree/bindings/sound/test-component.yaml create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8962.yaml create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8978.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8962.txt create mode 100644 drivers/firmware/cirrus/Kconfig create mode 100644 drivers/firmware/cirrus/Makefile create mode 100644 drivers/firmware/cirrus/cs_dsp.c create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/linux/firmware/cirrus/cs_dsp.h rename {sound/soc/codecs => include/linux/firmware/cirrus}/wmfw.h (91%) create mode 100644 include/sound/cs35l41.h create mode 100644 include/sound/rt5682s.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/amd/acp/Kconfig create mode 100644 sound/soc/amd/acp/Makefile create mode 100644 sound/soc/amd/acp/acp-i2s.c create mode 100644 sound/soc/amd/acp/acp-legacy-mach.c create mode 100644 sound/soc/amd/acp/acp-mach-common.c create mode 100644 sound/soc/amd/acp/acp-mach.h create mode 100644 sound/soc/amd/acp/acp-platform.c create mode 100644 sound/soc/amd/acp/acp-renoir.c create mode 100644 sound/soc/amd/acp/acp-sof-mach.c create mode 100644 sound/soc/amd/acp/amd.h create mode 100644 sound/soc/amd/acp/chip_offset_byte.h create mode 100644 sound/soc/amd/vangogh/acp5x-mach.c create mode 100644 sound/soc/amd/yc/Makefile create mode 100644 sound/soc/amd/yc/acp6x-mach.c create mode 100644 sound/soc/amd/yc/acp6x-pdm-dma.c create mode 100644 sound/soc/amd/yc/acp6x.h create mode 100644 sound/soc/amd/yc/acp6x_chip_offset_byte.h create mode 100644 sound/soc/amd/yc/pci-acp6x.c create mode 100644 sound/soc/codecs/cs35l41-i2c.c create mode 100644 sound/soc/codecs/cs35l41-spi.c create mode 100644 sound/soc/codecs/cs35l41-tables.c create mode 100644 sound/soc/codecs/cs35l41.c create mode 100644 sound/soc/codecs/cs35l41.h create mode 100644 sound/soc/codecs/max98520.c create mode 100644 sound/soc/codecs/max98520.h create mode 100644 sound/soc/codecs/nau8821.c create mode 100644 sound/soc/codecs/nau8821.h create mode 100644 sound/soc/codecs/rt5682s.c create mode 100644 sound/soc/codecs/rt5682s.h create mode 100644 sound/soc/codecs/rt9120.c create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.c create mode 100644 sound/soc/generic/audio-graph-card2-custom-sample.dtsi create mode 100644 sound/soc/generic/audio-graph-card2.c create mode 100644 sound/soc/generic/test-component.c create mode 100644 sound/soc/intel/boards/sof_es8336.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.c create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.h delete mode 100644 sound/soc/rockchip/rockchip_pcm.c delete mode 100644 sound/soc/rockchip/rockchip_pcm.h delete mode 100644 sound/soc/sof/compress.h create mode 100644 sound/soc/sof/imx/imx-ops.h rename sound/soc/sof/intel/{hda-compress.c => hda-probes.c} (100%) delete mode 100644 sound/soc/sof/probe.h rename sound/soc/sof/{probe.c => sof-probes.c} (52%) create mode 100644 sound/soc/sof/sof-probes.h rename sound/soc/sof/{intel/intel-ipc.c => stream-ipc.c} (56%) create mode 100644 sound/soc/tegra/tegra210_adx.c create mode 100644 sound/soc/tegra/tegra210_adx.h create mode 100644 sound/soc/tegra/tegra210_amx.c create mode 100644 sound/soc/tegra/tegra210_amx.h create mode 100644 sound/soc/tegra/tegra210_mixer.c create mode 100644 sound/soc/tegra/tegra210_mixer.h create mode 100644 sound/soc/tegra/tegra210_mvc.c create mode 100644 sound/soc/tegra/tegra210_mvc.h create mode 100644 sound/soc/tegra/tegra210_sfc.c create mode 100644 sound/soc/tegra/tegra210_sfc.h

2 1

[PATCH v2 2/2] ASoC: codec: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 01 Nov '21

01 Nov '21

New codec driver for Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1071 ++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 379 +++++++++++ 4 files changed, 1459 insertions(+) create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 326f2d611ad4..b81c4721c0d5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -220,6 +220,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TDA7419 imply SND_SOC_TFA9879 imply SND_SOC_TFA989X + imply SND_SOC_TLV320ADC3XXX imply SND_SOC_TLV320ADCX140 imply SND_SOC_TLV320AIC23_I2C imply SND_SOC_TLV320AIC23_SPI @@ -1486,6 +1487,12 @@ config SND_SOC_TFA989X Note that the driver currently bypasses the built-in "CoolFlux DSP" and does not support (hardware) volume control. +config SND_SOC_TLV320ADC3XXX + tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" + help + Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 + ADCs. + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9acfbcbfc46d..5112c12e6d76 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -238,6 +238,7 @@ snd-soc-tda7419-objs := tda7419.o snd-soc-tas2770-objs := tas2770.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o +snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o @@ -572,6 +573,7 @@ obj-$(CONFIG_SND_SOC_TDA7419) += snd-soc-tda7419.o obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TFA989X) += snd-soc-tfa989x.o +obj-$(CONFIG_SND_SOC_TLV320ADC3XXX) += snd-soc-tlv320adc3xxx.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c new file mode 100644 index 000000000000..584cdfbd92b4 --- /dev/null +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -0,0 +1,1071 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov +// +// History: +// +// Author: "Shahina Shaik" < shahina.s(a)mistralsolutions.com > +// Copyright: (C) 2010 Mistral Solutions Pvt Ltd. +// +// Author: Dongge wu <dgwu(a)ambarella.com> +// 2015/10/28 - [Dongge wu] Created file +// Copyright (C) 2014-2018, Ambarella, Inc. +// +// Author: Ricard Wanderlof <ricardw(a)axis.com> +// 2020/11/05: Fixing driver for Linux 4.14: more clocking modes, etc. +// 2021/09/03: Porting driver to Linux 5.4 and later. Added GPIO pin control, +// DAPM controlled clock chain, general cleanup. +// Copyright (C) 2021 Axis Communications AB +// + +#include <dt-bindings/sound/tlv320adc3xxx.h> +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio/driver.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/cdev.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "tlv320adc3xxx.h" + +enum adc3xxx_type { + ADC3001 = 0, + ADC3101 +}; + +struct adc3xxx { + struct device *dev; + enum adc3xxx_type type; + struct clk *mclk; + struct regmap *regmap; + unsigned int pll_mode; + unsigned int rst_pin; + unsigned int rst_active; + unsigned int sysclk; + unsigned int gpio_cfg[ADC3XXX_GPIOS_MAX]; /* value+1 (0 => not set) */ + unsigned int micbias_vg[ADC3XXX_MICBIAS_PINS]; + int master; + u8 page_no; + int use_pll; +#ifdef CONFIG_GPIOLIB + struct gpio_chip gpio_chip; +#endif +}; + +static const unsigned int adc3xxx_gpio_ctrl_reg[ADC3XXX_GPIOS_MAX] = { + ADC3XXX_GPIO1_CTRL, + ADC3XXX_GPIO2_CTRL +}; + +static const unsigned int adc3xxx_micbias_shift[ADC3XXX_MICBIAS_PINS] = { + ADC3XXX_MICBIAS1_SHIFT, + ADC3XXX_MICBIAS2_SHIFT +}; + +static const struct reg_default adc3xxx_defaults[] = { + /* Page 0 */ + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, { 3, 0x00 }, + { 4, 0x00 }, { 5, 0x11 }, { 6, 0x04 }, { 7, 0x00 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x00 }, { 11, 0x00 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x00 }, { 15, 0x00 }, + { 16, 0x00 }, { 17, 0x00 }, { 18, 0x01 }, { 19, 0x01 }, + { 20, 0x80 }, { 21, 0x80 }, { 22, 0x04 }, { 23, 0x00 }, + { 24, 0x00 }, { 25, 0x00 }, { 26, 0x01 }, { 27, 0x00 }, + { 28, 0x00 }, { 29, 0x02 }, { 30, 0x01 }, { 31, 0x00 }, + { 32, 0x00 }, { 33, 0x10 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x00 }, { 38, 0x02 }, { 39, 0x00 }, + { 40, 0x00 }, { 41, 0x00 }, { 42, 0x00 }, { 43, 0x00 }, + { 44, 0x00 }, { 45, 0x00 }, { 46, 0x00 }, { 47, 0x00 }, + { 48, 0x00 }, { 49, 0x00 }, { 50, 0x00 }, { 51, 0x00 }, + { 52, 0x00 }, { 53, 0x12 }, { 54, 0x00 }, { 55, 0x00 }, + { 56, 0x00 }, { 57, 0x00 }, { 58, 0x00 }, { 59, 0x44 }, + { 60, 0x00 }, { 61, 0x01 }, { 62, 0x00 }, { 63, 0x00 }, + { 64, 0x00 }, { 65, 0x00 }, { 66, 0x00 }, { 67, 0x00 }, + { 68, 0x00 }, { 69, 0x00 }, { 70, 0x00 }, { 71, 0x00 }, + { 72, 0x00 }, { 73, 0x00 }, { 74, 0x00 }, { 75, 0x00 }, + { 76, 0x00 }, { 77, 0x00 }, { 78, 0x00 }, { 79, 0x00 }, + { 80, 0x00 }, { 81, 0x00 }, { 82, 0x88 }, { 83, 0x00 }, + { 84, 0x00 }, { 85, 0x00 }, { 86, 0x00 }, { 87, 0x00 }, + { 88, 0x7f }, { 89, 0x00 }, { 90, 0x00 }, { 91, 0x00 }, + { 92, 0x00 }, { 93, 0x00 }, { 94, 0x00 }, { 95, 0x00 }, + { 96, 0x7f }, { 97, 0x00 }, { 98, 0x00 }, { 99, 0x00 }, + { 100, 0x00 }, { 101, 0x00 }, { 102, 0x00 }, { 103, 0x00 }, + { 104, 0x00 }, { 105, 0x00 }, { 106, 0x00 }, { 107, 0x00 }, + { 108, 0x00 }, { 109, 0x00 }, { 110, 0x00 }, { 111, 0x00 }, + { 112, 0x00 }, { 113, 0x00 }, { 114, 0x00 }, { 115, 0x00 }, + { 116, 0x00 }, { 117, 0x00 }, { 118, 0x00 }, { 119, 0x00 }, + { 120, 0x00 }, { 121, 0x00 }, { 122, 0x00 }, { 123, 0x00 }, + { 124, 0x00 }, { 125, 0x00 }, { 126, 0x00 }, { 127, 0x00 }, + + /* Page 1 */ + { 128, 0x00 }, { 129, 0x00 }, { 130, 0x00 }, { 131, 0x00 }, + { 132, 0x00 }, { 133, 0x00 }, { 134, 0x00 }, { 135, 0x00 }, + { 136, 0x00 }, { 137, 0x00 }, { 138, 0x00 }, { 139, 0x00 }, + { 140, 0x00 }, { 141, 0x00 }, { 142, 0x00 }, { 143, 0x00 }, + { 144, 0x00 }, { 145, 0x00 }, { 146, 0x00 }, { 147, 0x00 }, + { 148, 0x00 }, { 149, 0x00 }, { 150, 0x00 }, { 151, 0x00 }, + { 152, 0x00 }, { 153, 0x00 }, { 154, 0x00 }, { 155, 0x00 }, + { 156, 0x00 }, { 157, 0x00 }, { 158, 0x00 }, { 159, 0x00 }, + { 160, 0x00 }, { 161, 0x00 }, { 162, 0x00 }, { 163, 0x00 }, + { 164, 0x00 }, { 165, 0x00 }, { 166, 0x00 }, { 167, 0x00 }, + { 168, 0x00 }, { 169, 0x00 }, { 170, 0x00 }, { 171, 0x00 }, + { 172, 0x00 }, { 173, 0x00 }, { 174, 0x00 }, { 175, 0x00 }, + { 176, 0x00 }, { 177, 0x00 }, { 178, 0x00 }, { 179, 0x00 }, + { 180, 0xff }, { 181, 0x00 }, { 182, 0x3f }, { 183, 0xff }, + { 184, 0x00 }, { 185, 0x3f }, { 186, 0x00 }, { 187, 0x80 }, + { 188, 0x80 }, { 189, 0x00 }, { 190, 0x00 }, { 191, 0x00 }, +}; + +static bool adc3xxx_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADC3XXX_RESET: + return true; + default: + return false; + } +} + +static const struct regmap_range_cfg adc3xxx_ranges[] = { + { + .range_min = 0, + .range_max = 2 * ADC3XXX_PAGE_SIZE, + .selector_reg = ADC3XXX_PAGE_SELECT, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = ADC3XXX_PAGE_SIZE, + } +}; + +static const struct regmap_config adc3xxx_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = adc3xxx_defaults, + .num_reg_defaults = ARRAY_SIZE(adc3xxx_defaults), + + .volatile_reg = adc3xxx_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + + .ranges = adc3xxx_ranges, + .num_ranges = ARRAY_SIZE(adc3xxx_ranges), + .max_register = 2 * ADC3XXX_PAGE_SIZE, +}; + +struct adc3xxx_rate_divs { + u32 mclk; + u32 rate; + u8 pll_p; + u8 pll_r; + u8 pll_j; + u16 pll_d; + u8 nadc; + u8 madc; + u8 aosr; +}; + +/* + * PLL and Clock settings. + * If p member is 0, PLL is not used. + * The order of the entries in this table have the PLL entries before + * the non-PLL entries, so that the PLL modes are preferred unless + * the PLL mode setting says otherwise. + */ +static const struct adc3xxx_rate_divs adc3xxx_divs[] = { + /* mclk, rate, p, r, j, d, nadc, madc, aosr */ + /* 8k rate */ + { 12000000, 8000, 1, 1, 7, 1680, 42, 2, 128 }, + { 12288000, 8000, 1, 1, 7, 0000, 42, 2, 128 }, + /* 11.025k rate */ + { 12000000, 11025, 1, 1, 6, 8208, 29, 2, 128 }, + /* 16k rate */ + { 12000000, 16000, 1, 1, 7, 1680, 21, 2, 128 }, + { 12288000, 16000, 1, 1, 7, 0000, 21, 2, 128 }, + /* 22.05k rate */ + { 12000000, 22050, 1, 1, 7, 560, 15, 2, 128 }, + /* 32k rate */ + { 12000000, 32000, 1, 1, 8, 1920, 12, 2, 128 }, + { 12288000, 32000, 1, 1, 8, 0000, 12, 2, 128 }, + /* 44.1k rate */ + { 12000000, 44100, 1, 1, 7, 5264, 8, 2, 128 }, + /* 48k rate */ + { 12000000, 48000, 1, 1, 7, 1680, 7, 2, 128 }, + { 12288000, 48000, 1, 1, 7, 0000, 7, 2, 128 }, + { 24576000, 48000, 1, 1, 3, 5000, 7, 2, 128 }, /* With PLL */ + { 24576000, 48000, 0, 0, 0, 0000, 2, 2, 128 }, /* Without PLL */ + /* 88.2k rate */ + { 12000000, 88200, 1, 1, 7, 5264, 4, 4, 64 }, + /* 96k rate */ + { 12000000, 96000, 1, 1, 8, 1920, 4, 4, 64 }, +}; + +static int adc3xxx_get_divs(struct device *dev, int mclk, int rate, int pll_mode) +{ + int i; + + dev_dbg(dev, "mclk = %d, rate = %d, clock mode %u\n", + mclk, rate, pll_mode); + for (i = 0; i < ARRAY_SIZE(adc3xxx_divs); i++) { + const struct adc3xxx_rate_divs *mode = &adc3xxx_divs[i]; + + /* Skip this entry if it doesn't fulfill the intended clock + * mode requirement. We consider anything besides the two + * modes below to be the same as ADC3XXX_PLL_AUTO. + */ + if ((pll_mode == ADC3XXX_PLL_BYPASS && mode->pll_p) || + (pll_mode == ADC3XXX_PLL_ENABLE && !mode->pll_p)) + continue; + + if (mode->rate == rate && mode->mclk == mclk) + return i; + } + + dev_info(dev, "Master clock rate %d and sample rate %d is not supported\n", + mclk, rate); + return -EINVAL; +} + +int adc3xxx_pll_delay(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* 10msec delay needed after PLL power-up to allow + * PLL and dividers to stabilize (datasheet p13). + */ + usleep_range(10000, 20000); + + return 0; +} + +static const char * const adc_softstepping_text[] = { "1 step", "2 step", "off" }; +static const SOC_ENUM_SINGLE_DECL(adc_softstepping_enum, ADC3XXX_ADC_DIGITAL, 0, + adc_softstepping_text); + +static const char * const multiplier_text[] = { "1", "2", "4", "8", "16", "32", "64", "128" }; +static const SOC_ENUM_SINGLE_DECL(left_agc_attack_mult_enum, + ADC3XXX_LEFT_CHN_AGC_4, 0, multiplier_text); +static const SOC_ENUM_SINGLE_DECL(right_agc_attack_mult_enum, + ADC3XXX_RIGHT_CHN_AGC_4, 0, multiplier_text); +static const SOC_ENUM_SINGLE_DECL(left_agc_decay_mult_enum, + ADC3XXX_LEFT_CHN_AGC_5, 0, multiplier_text); +static const SOC_ENUM_SINGLE_DECL(right_agc_decay_mult_enum, + ADC3XXX_RIGHT_CHN_AGC_5, 0, multiplier_text); + +static const char * const dither_dc_offset_text[] = { + "0mV", "15mV", "30mV", "45mV", "60mV", "75mV", "90mV", "105mV", + "-15mV", "-30mV", "-45mV", "-60mV", "-75mV", "-90mV", "-105mV" +}; +static const unsigned int dither_dc_offset_values[] = { + 0, 1, 2, 3, 4, 5, 6, 7, 9, 10, 11, 12, 13, 14, 15 +}; +static const SOC_VALUE_ENUM_DOUBLE_DECL(dither_dc_offset_enum, + ADC3XXX_DITHER_CTRL, + 4, 0, 0xf, dither_dc_offset_text, + dither_dc_offset_values); + +static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fine_tlv, -40, 10, 0); +/* AGC target: 8 values: -5.5, -8, -10, -12, -14, -17, -20, -24 dB */ +/* It would be nice to declare these in the order above, but empirically + * TLV_DB_SCALE_ITEM doesn't take lightly to the increment (second) parameter + * being negative, despite there being examples to the contrary in other + * drivers. So declare these in the order from lowest to highest, and + * set the invert flag in the SOC_DOUBLE_R_TLV declaration instead. + */ +static const DECLARE_TLV_DB_RANGE(agc_target_tlv, + 0, 0, TLV_DB_SCALE_ITEM(-2400, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(-2000, 300, 0), + 4, 6, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 7, 7, TLV_DB_SCALE_ITEM(-550, 0, 0)); +/* Since the 'disabled' value (mute) is at the highest value in the dB + * range (i.e. just before -32 dB) rather than the lowest, we need to resort + * to using a TLV_DB_RANGE in order to get the mute value in the right place. + */ +static const DECLARE_TLV_DB_RANGE(agc_thresh_tlv, + 0, 30, TLV_DB_SCALE_ITEM(-9000, 200, 0), + 31, 31, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +/* AGC hysteresis: 4 values: 1, 2, 4 dB, disabled (= mute) */ +static const DECLARE_TLV_DB_RANGE(agc_hysteresis_tlv, + 0, 1, TLV_DB_SCALE_ITEM(100, 100, 0), + 2, 2, TLV_DB_SCALE_ITEM(400, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(0, 0, 1)); /* disabled = mute */ +static const DECLARE_TLV_DB_SCALE(agc_max_tlv, 0, 50, 0); +/* Input attenuation: -6 dB or 0 dB */ +static const DECLARE_TLV_DB_SCALE(input_attenuation_tlv, -600, 600, 0); + +static const struct snd_kcontrol_new adc3xxx_snd_controls[] = { + SOC_DOUBLE_R_TLV("PGA Capture Volume", ADC3XXX_LEFT_APGA_CTRL, + ADC3XXX_RIGHT_APGA_CTRL, 0, 80, 0, pga_tlv), + SOC_DOUBLE("PGA Capture Switch", ADC3XXX_ADC_FGA, 7, 3, 1, 1), + SOC_DOUBLE_R("AGC Capture Switch", ADC3XXX_LEFT_CHN_AGC_1, + ADC3XXX_RIGHT_CHN_AGC_1, 7, 1, 0), + SOC_DOUBLE_R_TLV("AGC Target Level Capture Volume", ADC3XXX_LEFT_CHN_AGC_1, + ADC3XXX_RIGHT_CHN_AGC_2, 4, 0x07, 1, agc_target_tlv), + SOC_DOUBLE_R_TLV("AGC Noise Threshold Capture Volume", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 1, 0x1f, 1, agc_thresh_tlv), + SOC_DOUBLE_R_TLV("AGC Hysteresis Capture Volume", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 6, 3, 0, agc_hysteresis_tlv), + SOC_DOUBLE_R("AGC Clip Stepping Capture Switch", ADC3XXX_LEFT_CHN_AGC_2, + ADC3XXX_RIGHT_CHN_AGC_2, 0, 1, 0), + /* + * Oddly enough, the data sheet says the default value + * for the left/right AGC maximum gain register field + * (ADC3XXX_LEFT/RIGHT_CHN_AGC_3 bits 0..6) is 0x7f = 127 + * (verified empirically) even though this value (indeed, above + * 0x50) is specified as 'Reserved. Do not use.' in the accompanying + * table in the data sheet. + */ + SOC_DOUBLE_R_TLV("AGC Maximum Capture Volume", ADC3XXX_LEFT_CHN_AGC_3, + ADC3XXX_RIGHT_CHN_AGC_3, 0, 0x50, 0, agc_max_tlv), + SOC_DOUBLE_R("AGC Attack Time", ADC3XXX_LEFT_CHN_AGC_4, + ADC3XXX_RIGHT_CHN_AGC_4, 3, 0x1f, 0), + /* Would like to have the multipliers as LR pairs, but there is + * no SOC_ENUM_foo which accepts two values in separate registers. + */ + SOC_ENUM("AGC Left Attack Time Multiplier", left_agc_attack_mult_enum), + SOC_ENUM("AGC Right Attack Time Multiplier", right_agc_attack_mult_enum), + SOC_DOUBLE_R("AGC Decay Time", ADC3XXX_LEFT_CHN_AGC_5, + ADC3XXX_RIGHT_CHN_AGC_5, 3, 0x1f, 0), + SOC_ENUM("AGC Left Decay Time Multiplier", left_agc_decay_mult_enum), + SOC_ENUM("AGC Right Decay Time Multiplier", right_agc_decay_mult_enum), + SOC_DOUBLE_R("AGC Noise Debounce", ADC3XXX_LEFT_CHN_AGC_6, + ADC3XXX_RIGHT_CHN_AGC_6, 0, 0x1f, 0), + SOC_DOUBLE_R("AGC Signal Debounce", ADC3XXX_LEFT_CHN_AGC_7, + ADC3XXX_RIGHT_CHN_AGC_7, 0, 0x0f, 0), + /* Read only register */ + SOC_DOUBLE_R_S_TLV("AGC Applied Capture Volume", ADC3XXX_LEFT_AGC_GAIN, + ADC3XXX_RIGHT_AGC_GAIN, 0, -24, 40, 6, 0, adc_tlv), + /* ADC soft stepping */ + SOC_ENUM("ADC Soft Stepping", adc_softstepping_enum), + /* Left/Right Input attenuation */ + SOC_SINGLE_TLV("Left Input IN_1L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_2L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_3L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input IN_1R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_2L_3L Capture Volume", + ADC3XXX_LEFT_PGA_SEL_1, 6, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_1L_1R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Left Input DIF_2R_3R Capture Volume", + ADC3XXX_LEFT_PGA_SEL_2, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_1R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_2R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 2, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_3R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input IN_1L Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 0, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_2R_3R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_1, 6, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_1L_1R Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 4, 1, 1, input_attenuation_tlv), + SOC_SINGLE_TLV("Right Input DIF_2L_3L Capture Volume", + ADC3XXX_RIGHT_PGA_SEL_2, 2, 1, 1, input_attenuation_tlv), + SOC_DOUBLE_R_S_TLV("ADC Volume Control Capture Volume", ADC3XXX_LADC_VOL, + ADC3XXX_RADC_VOL, 0, -24, 40, 6, 0, adc_tlv), + /* Empirically, the following doesn't work the way it's supposed + * to. Values 0, -0.1, -0.2 and -0.3 dB result in the same level, and + * -0.4 dB drops about 0.12 dB on a specific chip. + */ + SOC_DOUBLE_TLV("ADC Fine Volume Control Capture Volume", ADC3XXX_ADC_FGA, + 4, 0, 4, 1, adc_fine_tlv), + SOC_SINGLE("Left ADC Unselected CM Bias Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 6, 1, 0), + SOC_SINGLE("Right ADC Unselected CM Bias Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 6, 1, 0), + SOC_ENUM("Dither Control DC Offset", dither_dc_offset_enum), +}; + +/* Left input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L Capture Switch", + ADC3XXX_LEFT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1R Capture Switch", + ADC3XXX_LEFT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Right input selection, Single Ended inputs and Differential inputs */ +static const struct snd_kcontrol_new right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN_1R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 1, 0x1, 1), + SOC_DAPM_SINGLE("IN_2R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_3R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2R_3R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_1, 7, 0x1, 1), + SOC_DAPM_SINGLE("DIF_1L_1R Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 5, 0x1, 1), + SOC_DAPM_SINGLE("DIF_2L_3L Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 3, 0x1, 1), + SOC_DAPM_SINGLE("IN_1L Capture Switch", + ADC3XXX_RIGHT_PGA_SEL_2, 1, 0x1, 1), +}; + +/* Left Digital Mic input for left ADC */ +static const struct snd_kcontrol_new left_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Left ADC Capture Switch", + ADC3XXX_ADC_DIGITAL, 3, 0x1, 0), +}; + +/* Right Digital Mic input for Right ADC */ +static const struct snd_kcontrol_new right_input_dmic_controls[] = { + SOC_DAPM_SINGLE("Right ADC Capture Switch", + ADC3XXX_ADC_DIGITAL, 2, 0x1, 0), +}; + +/* DAPM widgets */ +static const struct snd_soc_dapm_widget adc3xxx_dapm_widgets[] = { + + /* Left Input Selection */ + SND_SOC_DAPM_MIXER("Left Input", SND_SOC_NOPM, 0, 0, + &left_input_mixer_controls[0], + ARRAY_SIZE(left_input_mixer_controls)), + /* Right Input Selection */ + SND_SOC_DAPM_MIXER("Right Input", SND_SOC_NOPM, 0, 0, + &right_input_mixer_controls[0], + ARRAY_SIZE(right_input_mixer_controls)), + /* PGA selection */ + SND_SOC_DAPM_PGA("Left PGA", ADC3XXX_LEFT_APGA_CTRL, 7, 1, NULL, 0), + SND_SOC_DAPM_PGA("Right PGA", ADC3XXX_RIGHT_APGA_CTRL, 7, 1, NULL, 0), + + /* Digital Microphone Input Control for Left/Right ADC */ + SND_SOC_DAPM_MIXER("Left DMic Input", SND_SOC_NOPM, 0, 0, + &left_input_dmic_controls[0], + ARRAY_SIZE(left_input_dmic_controls)), + SND_SOC_DAPM_MIXER("Right DMic Input", SND_SOC_NOPM, 0, 0, + &right_input_dmic_controls[0], + ARRAY_SIZE(right_input_dmic_controls)), + + /* Left/Right ADC */ + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ADC3XXX_ADC_DIGITAL, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ADC3XXX_ADC_DIGITAL, 6, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("IN_1L"), + SND_SOC_DAPM_INPUT("IN_1R"), + SND_SOC_DAPM_INPUT("IN_2L"), + SND_SOC_DAPM_INPUT("IN_2R"), + SND_SOC_DAPM_INPUT("IN_3L"), + SND_SOC_DAPM_INPUT("IN_3R"), + SND_SOC_DAPM_INPUT("DIFL_1L_1R"), + SND_SOC_DAPM_INPUT("DIFL_2L_3L"), + SND_SOC_DAPM_INPUT("DIFL_2R_3R"), + SND_SOC_DAPM_INPUT("DIFR_1L_1R"), + SND_SOC_DAPM_INPUT("DIFR_2L_3L"), + SND_SOC_DAPM_INPUT("DIFR_2R_3R"), + SND_SOC_DAPM_INPUT("DMic_L"), + SND_SOC_DAPM_INPUT("DMic_R"), + + /* Digital audio interface output */ + SND_SOC_DAPM_AIF_OUT("AIF_OUT", "Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Clocks */ + SND_SOC_DAPM_SUPPLY("PLL_CLK", ADC3XXX_PLL_PROG_PR, ADC3XXX_ENABLE_PLL_SHIFT, + 0, adc3xxx_pll_delay, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_SUPPLY("ADC_CLK", ADC3XXX_ADC_NADC, ADC3XXX_ENABLE_NADC_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC_MOD_CLK", ADC3XXX_ADC_MADC, ADC3XXX_ENABLE_MADC_SHIFT, + 0, NULL, 0), + + /* This refers to the generated BCLK in master mode. */ + SND_SOC_DAPM_SUPPLY("BCLK", ADC3XXX_BCLK_N_DIV, ADC3XXX_ENABLE_BCLK_SHIFT, + 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route adc3xxx_intercon[] = { + /* Left input selection from switches */ + { "Left Input", "IN_1L Capture Switch", "IN_1L" }, + { "Left Input", "IN_2L Capture Switch", "IN_2L" }, + { "Left Input", "IN_3L Capture Switch", "IN_3L" }, + { "Left Input", "DIF_2L_3L Capture Switch", "DIFL_2L_3L" }, + { "Left Input", "DIF_1L_1R Capture Switch", "DIFL_1L_1R" }, + { "Left Input", "DIF_2R_3R Capture Switch", "DIFL_2R_3R" }, + { "Left Input", "IN_1R Capture Switch", "IN_1R" }, + + /* Left input selection to left PGA */ + { "Left PGA", NULL, "Left Input" }, + + /* Left PGA to left ADC */ + { "Left ADC", NULL, "Left PGA" }, + + /* Right input selection from switches */ + { "Right Input", "IN_1R Capture Switch", "IN_1R" }, + { "Right Input", "IN_2R Capture Switch", "IN_2R" }, + { "Right Input", "IN_3R Capture Switch", "IN_3R" }, + { "Right Input", "DIF_2R_3R Capture Switch", "DIFR_2R_3R" }, + { "Right Input", "DIF_1L_1R Capture Switch", "DIFR_1L_1R" }, + { "Right Input", "DIF_2L_3L Capture Switch", "DIFR_2L_3L" }, + { "Right Input", "IN_1L Capture Switch", "IN_1L" }, + + /* Right input selection to right PGA */ + { "Right PGA", NULL, "Right Input" }, + + /* Right PGA to right ADC */ + { "Right ADC", NULL, "Right PGA" }, + + /* Left DMic Input selection from switch */ + { "Left DMic Input", "Left ADC Capture Switch", "DMic_L" }, + + /* Left DMic to left ADC */ + { "Left ADC", NULL, "Left DMic Input" }, + + /* Right DMic Input selection from switch */ + { "Right DMic Input", "Right ADC Capture Switch", "DMic_R" }, + + /* Right DMic to right ADC */ + { "Right ADC", NULL, "Right DMic Input" }, + + /* ADC to AIF output */ + { "AIF_OUT", NULL, "Left ADC" }, + { "AIF_OUT", NULL, "Right ADC" }, + + /* Clocking */ + { "ADC_MOD_CLK", NULL, "ADC_CLK" }, + { "Left ADC", NULL, "ADC_MOD_CLK" }, + { "Right ADC", NULL, "ADC_MOD_CLK" }, + + { "BCLK", NULL, "ADC_CLK" }, +}; + +static const struct snd_soc_dapm_route adc3xxx_pll_intercon[] = { + { "ADC_CLK", NULL, "PLL_CLK" }, +}; + +static const struct snd_soc_dapm_route adc3xxx_bclk_out_intercon[] = { + { "AIF_OUT", NULL, "BCLK" } +}; + +#ifdef CONFIG_GPIOLIB +static int adc3xxx_gpio_request(struct gpio_chip *chip, unsigned int offset) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + + if (offset >= ADC3XXX_GPIOS_MAX) + return -EINVAL; + + /* GPIO1 is offset 0, GPIO2 is offset 1 */ + /* We check here that the GPIO pins are either not configured in the + * DT, or that they purposely are set as outputs. + * (Input mode not yet implemented). + */ + if (adc3xxx->gpio_cfg[offset] != 0 && + adc3xxx->gpio_cfg[offset] != ADC3XXX_GPIO_GPO + 1) + return -EINVAL; + + return 0; +} + +static int adc3xxx_gpio_direction_out(struct gpio_chip *chip, + unsigned int offset, int value) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + + /* Set GPIO output function. */ + return regmap_update_bits(adc3xxx->regmap, + adc3xxx_gpio_ctrl_reg[offset], + ADC3XXX_GPIO_CTRL_CFG_MASK | + ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK, + ADC3XXX_GPIO_GPO << ADC3XXX_GPIO_CTRL_CFG_SHIFT | + !!value << ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_SHIFT); +} + +/* With only GPIO outputs configured, we never get the .direction_out call, + * so we set the output mode and output value in the same call. Hence + * .set in practice does the same thing as .direction_out . + */ +static void adc3xxx_gpio_set(struct gpio_chip *chip, unsigned int offset, + int value) +{ + (void) adc3xxx_gpio_direction_out(chip, offset, value); +} + +/* Even though we only support GPIO output for now, some GPIO clients + * want to read the current pin state using the .get callback. + */ +static int adc3xxx_gpio_get(struct gpio_chip *chip, unsigned int offset) +{ + struct adc3xxx *adc3xxx = gpiochip_get_data(chip); + unsigned int regval; + int ret; + + /* We only allow output pins, so just read the value set in the output + * pin register field. + */ + ret = regmap_read(adc3xxx->regmap, adc3xxx_gpio_ctrl_reg[offset], &regval); + if (ret) + return ret; + return !!(regval & ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK); +} + +static const struct gpio_chip adc3xxx_gpio_chip = { + .label = "adc3xxx", + .owner = THIS_MODULE, + .request = adc3xxx_gpio_request, + .direction_output = adc3xxx_gpio_direction_out, + .set = adc3xxx_gpio_set, + .get = adc3xxx_gpio_get, + .can_sleep = 1, +}; + +static void adc3xxx_free_gpio(struct adc3xxx *adc3xxx) +{ + gpiochip_remove(&adc3xxx->gpio_chip); +} + +#endif + +static void adc3xxx_init_gpio(struct adc3xxx *adc3xxx) +{ + int gpio, micbias; +#ifdef CONFIG_GPIOLIB + int ret; + + adc3xxx->gpio_chip = adc3xxx_gpio_chip; + adc3xxx->gpio_chip.ngpio = ADC3XXX_GPIOS_MAX; + adc3xxx->gpio_chip.parent = adc3xxx->dev; + adc3xxx->gpio_chip.base = -1; + + ret = gpiochip_add_data(&adc3xxx->gpio_chip, adc3xxx); + if (ret) + dev_err(adc3xxx->dev, "Failed to add gpios: %d\n", ret); +#endif + + /* Set up potential GPIO configuration from the devicetree. + * This allows us to set up things which are not software + * controllable GPIOs, such as PDM microphone I/O, + */ + for (gpio = 0; gpio < ADC3XXX_GPIOS_MAX; gpio++) { + unsigned int cfg = adc3xxx->gpio_cfg[gpio]; + + if (cfg) { + cfg--; /* actual value to use is stored +1 */ + regmap_update_bits(adc3xxx->regmap, + adc3xxx_gpio_ctrl_reg[gpio], + ADC3XXX_GPIO_CTRL_CFG_MASK, + cfg << ADC3XXX_GPIO_CTRL_CFG_SHIFT); + } + } + + /* Set up micbias voltage */ + for (micbias = 0; micbias < ADC3XXX_MICBIAS_PINS; micbias++) { + unsigned int vg = adc3xxx->micbias_vg[micbias]; + + regmap_update_bits(adc3xxx->regmap, + ADC3XXX_MICBIAS_CTRL, + ADC3XXX_MICBIAS_MASK << adc3xxx_micbias_shift[micbias], + vg << adc3xxx_micbias_shift[micbias]); + } +} + +static int adc3xxx_parse_dt_gpio(struct adc3xxx *adc3xxx, + const char *propname, unsigned int *cfg) +{ + struct device *dev = adc3xxx->dev; + struct device_node *np = dev->of_node; + unsigned int val; + + if (!of_property_read_u32(np, propname, &val)) { + if (val & ~15 || val == 7 || val >= 11) { + dev_err(dev, "Invalid property value for '%s'\n", propname); + return -EINVAL; + } + if (val == ADC3XXX_GPIO_GPI) + dev_warn(dev, "GPIO Input read not yet implemented\n"); + *cfg = val + 1; /* 0 => not set up, all others shifted +1 */ + } + return 0; +} + +static int adc3xxx_parse_dt_micbias(struct adc3xxx *adc3xxx, + const char *propname, unsigned int *vg) +{ + struct device *dev = adc3xxx->dev; + struct device_node *np = dev->of_node; + unsigned int val; + + if (!of_property_read_u32(np, propname, &val)) { + if (val >= ADC3XXX_MICBIAS_AVDD) { + dev_err(dev, "Invalid property value for '%s'\n", propname); + return -EINVAL; + } + *vg = val; + } + return 0; +} + +static int adc3xxx_parse_pll_mode(uint32_t val, unsigned int *pll_mode) +{ + if (val != ADC3XXX_PLL_ENABLE && val != ADC3XXX_PLL_BYPASS && + val != ADC3XXX_PLL_AUTO) + return -EINVAL; + + *pll_mode = val; + + return 0; +} + +static void adc3xxx_setup_pll(struct snd_soc_component *component, + int div_entry) +{ + int i = div_entry; + + /* P & R values */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_PR, + (adc3xxx_divs[i].pll_p << ADC3XXX_PLLP_SHIFT) | + (adc3xxx_divs[i].pll_r << ADC3XXX_PLLR_SHIFT)); + /* J value */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_J, + adc3xxx_divs[i].pll_j & ADC3XXX_PLLJ_MASK); + /* D value */ + snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_LSB, + adc3xxx_divs[i].pll_d & ADC3XXX_PLLD_LSB_MASK); + snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_MSB, + (adc3xxx_divs[i].pll_d >> 8) & ADC3XXX_PLLD_MSB_MASK); +} + +static int adc3xxx_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 snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(dai->component); + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + int i, width = 16; + u8 iface_len, bdiv; + + i = adc3xxx_get_divs(component->dev, adc3xxx->sysclk, + params_rate(params), adc3xxx->pll_mode); + + if (i < 0) + return i; + + /* select data word length */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface_len = ADC3XXX_IFACE_16BITS; + width = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface_len = ADC3XXX_IFACE_20BITS; + width = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface_len = ADC3XXX_IFACE_24BITS; + width = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface_len = ADC3XXX_IFACE_32BITS; + width = 32; + break; + default: + dev_err(component->dev, "Unsupported serial data format\n"); + return -EINVAL; + } + snd_soc_component_update_bits(component, ADC3XXX_INTERFACE_CTRL_1, + ADC3XXX_WLENGTH_MASK, iface_len); + if (adc3xxx_divs[i].pll_p) { /* If PLL used for this mode */ + adc3xxx_setup_pll(component, i); + snd_soc_component_write(component, ADC3XXX_CLKGEN_MUX, ADC3XXX_USE_PLL); + if (!adc3xxx->use_pll) { + snd_soc_dapm_add_routes(dapm, adc3xxx_pll_intercon, + ARRAY_SIZE(adc3xxx_pll_intercon)); + adc3xxx->use_pll = 1; + } + } else { + snd_soc_component_write(component, ADC3XXX_CLKGEN_MUX, ADC3XXX_NO_PLL); + if (adc3xxx->use_pll) { + snd_soc_dapm_del_routes(dapm, adc3xxx_pll_intercon, + ARRAY_SIZE(adc3xxx_pll_intercon)); + adc3xxx->use_pll = 0; + } + } + + /* NADC */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_NADC, + ADC3XXX_NADC_MASK, adc3xxx_divs[i].nadc); + /* MADC */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_MADC, + ADC3XXX_MADC_MASK, adc3xxx_divs[i].madc); + /* AOSR */ + snd_soc_component_update_bits(component, ADC3XXX_ADC_AOSR, + ADC3XXX_AOSR_MASK, adc3xxx_divs[i].aosr); + /* BDIV N Value */ + /* BCLK is (by default) set up to be derived from ADC_CLK */ + bdiv = (adc3xxx_divs[i].aosr * adc3xxx_divs[i].madc) / (2 * width); + snd_soc_component_update_bits(component, ADC3XXX_BCLK_N_DIV, + ADC3XXX_BDIV_MASK, bdiv); + + return 0; +} + +static const char * const adc3xxx_pll_mode_text(int pll_mode) +{ + switch (pll_mode) { + case ADC3XXX_PLL_AUTO: + return "PLL auto"; + case ADC3XXX_PLL_ENABLE: + return "PLL enable"; + case ADC3XXX_PLL_BYPASS: + return "PLL bypass"; + default: + break; + } + + return "PLL unknown"; +} + +static int adc3xxx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + int ret; + + ret = adc3xxx_parse_pll_mode(clk_id, &adc3xxx->pll_mode); + if (ret < 0) + return ret; + + adc3xxx->sysclk = freq; + dev_dbg(component->dev, "Set sysclk to %u Hz, %s\n", + freq, adc3xxx_pll_mode_text(adc3xxx->pll_mode)); + return 0; +} + +static int adc3xxx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); + u8 clkdir = 0, format = 0; + int master = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + master = 1; + clkdir = ADC3XXX_BCLK_MASTER | ADC3XXX_WCLK_MASTER; + break; + case SND_SOC_DAIFMT_CBC_CFC: + master = 0; + break; + default: + dev_err(component->dev, "Invalid DAI clock setup\n"); + return -EINVAL; + } + + /* + * match both interface format and signal polarities since they + * are fixed + */ + switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK)) { + case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF: + format = ADC3XXX_FORMAT_DSP; + break; + case SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF: + format = ADC3XXX_FORMAT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_RJF; + break; + case SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF: + format = ADC3XXX_FORMAT_LJF; + break; + default: + dev_err(component->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* Add/del route enabling BCLK output as applicable */ + if (master && !adc3xxx->master) + snd_soc_dapm_add_routes(dapm, adc3xxx_bclk_out_intercon, + ARRAY_SIZE(adc3xxx_bclk_out_intercon)); + else if (!master && adc3xxx->master) + snd_soc_dapm_del_routes(dapm, adc3xxx_bclk_out_intercon, + ARRAY_SIZE(adc3xxx_bclk_out_intercon)); + adc3xxx->master = master; + + /* set clock direction and format */ + return snd_soc_component_update_bits(component, + ADC3XXX_INTERFACE_CTRL_1, + ADC3XXX_CLKDIR_MASK | ADC3XXX_FORMAT_MASK, + clkdir | format); +} + +static struct snd_soc_dai_ops adc3xxx_dai_ops = { + .hw_params = adc3xxx_hw_params, + .set_sysclk = adc3xxx_set_dai_sysclk, + .set_fmt = adc3xxx_set_dai_fmt, +}; + +struct snd_soc_dai_driver adc3xxx_dai = { + .name = "tlv320adc3xxx-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ADC3XXX_RATES, + .formats = ADC3XXX_FORMATS, + }, + .ops = &adc3xxx_dai_ops, +}; + +static const struct snd_soc_component_driver soc_component_dev_adc3xxx = { + .controls = adc3xxx_snd_controls, + .num_controls = ARRAY_SIZE(adc3xxx_snd_controls), + .dapm_widgets = adc3xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adc3xxx_dapm_widgets), + .dapm_routes = adc3xxx_intercon, + .num_dapm_routes = ARRAY_SIZE(adc3xxx_intercon), +}; + +static int adc3xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; + struct adc3xxx *adc3xxx = NULL; + enum of_gpio_flags flags; + int ret; + + adc3xxx = devm_kzalloc(dev, sizeof(struct adc3xxx), GFP_KERNEL); + if (!adc3xxx) + return -ENOMEM; + adc3xxx->dev = dev; + + adc3xxx->mclk = devm_clk_get(dev, NULL); + if (IS_ERR(adc3xxx->mclk)) { + /* + * The chip itself supports running off the BCLK either + * directly or via the PLL, but the driver does not (yet), so + * having a specified mclk is required. Otherwise, we could + * use the lack of a clocks property to indicate when BCLK is + * intended as the clock source. + */ + return PTR_ERR(adc3xxx->mclk); + } else if (adc3xxx->mclk) { + ret = clk_prepare_enable(adc3xxx->mclk); + if (ret < 0) + return ret; + dev_dbg(dev, "Enabled MCLK, freq %lu Hz\n", clk_get_rate(adc3xxx->mclk)); + } + + adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmdin-gpio1", &adc3xxx->gpio_cfg[0]); + adc3xxx_parse_dt_gpio(adc3xxx, "ti,dmclk-gpio2", &adc3xxx->gpio_cfg[1]); + adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias1-vg", &adc3xxx->micbias_vg[0]); + adc3xxx_parse_dt_micbias(adc3xxx, "ti,micbias2-vg", &adc3xxx->micbias_vg[1]); + + adc3xxx->regmap = devm_regmap_init_i2c(i2c, &adc3xxx_regmap); + if (IS_ERR(adc3xxx->regmap)) { + ret = PTR_ERR(adc3xxx->regmap); + return ret; + } + + i2c_set_clientdata(i2c, adc3xxx); + + adc3xxx->rst_pin = of_get_named_gpio_flags(np, "reset-gpios", 0, &flags); + if (adc3xxx->rst_pin < 0 || !gpio_is_valid(adc3xxx->rst_pin)) + return -ENXIO; + + adc3xxx->rst_active = !(flags & OF_GPIO_ACTIVE_LOW); + adc3xxx->type = id->driver_data; + + ret = devm_gpio_request(dev, adc3xxx->rst_pin, "adc3xxx reset"); + if (ret < 0) { + dev_err(dev, "Failed to request rst_pin: %d\n", ret); + return ret; + } + + /* Reset codec chip */ + gpio_direction_output(adc3xxx->rst_pin, adc3xxx->rst_active); + usleep_range(2000, 100000); /* Requirement: > 10 ns (datasheet p13) */ + gpio_direction_output(adc3xxx->rst_pin, !adc3xxx->rst_active); + + /* Potentially set up pins used as GPIOs */ + adc3xxx_init_gpio(adc3xxx); + + ret = snd_soc_register_component(dev, + &soc_component_dev_adc3xxx, &adc3xxx_dai, 1); + if (ret < 0) + dev_err(dev, "Failed to register codec: %d\n", ret); + + return ret; +} + +static int __exit adc3xxx_i2c_remove(struct i2c_client *client) +{ + struct adc3xxx *adc3xxx = i2c_get_clientdata(client); + + if (adc3xxx->mclk) + clk_disable_unprepare(adc3xxx->mclk); +#ifdef CONFIG_GPIOLIB + adc3xxx_free_gpio(adc3xxx); +#endif + snd_soc_unregister_component(&client->dev); + return 0; +} + +static const struct of_device_id tlv320adc3xxx_of_match[] = { + { .compatible = "ti,tlv320adc3001", }, + { .compatible = "ti,tlv320adc3101", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320adc3xxx_of_match); + +static const struct i2c_device_id adc3xxx_i2c_id[] = { + { "tlv320adc3001", ADC3001 }, + { "tlv320adc3101", ADC3101 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, adc3xxx_i2c_id); + +static struct i2c_driver adc3xxx_i2c_driver = { + .driver = { + .name = "tlv320adc3xxx-codec", + .of_match_table = tlv320adc3xxx_of_match, + }, + .probe = adc3xxx_i2c_probe, + .remove = adc3xxx_i2c_remove, + .id_table = adc3xxx_i2c_id, +}; + +module_i2c_driver(adc3xxx_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320ADC3xxx codec driver"); +MODULE_AUTHOR(" shahina.s(a)mistralsolutions.com "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/tlv320adc3xxx.h b/sound/soc/codecs/tlv320adc3xxx.h new file mode 100644 index 000000000000..82f82c3e7bc5 --- /dev/null +++ b/sound/soc/codecs/tlv320adc3xxx.h @@ -0,0 +1,379 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Based on sound/soc/codecs/tlv320aic3x.c by Vladimir Barinov + * + * History: + * + * Author: "Shahina Shaik" < shahina.s(a)mistralsolutions.com > + * Copyright: (C) 2010 Mistral Solutions Pvt Ltd. + * + * Author: Dongge wu <dgwu(a)ambarella.com> + * 2015/10/28 - [Dongge wu] Created file + * Copyright (C) 2014-2018, Ambarella, Inc. + * + * Author: Ricard Wanderlof <ricardw(a)axis.com> + * 2020/11/05: Fixing driver for Linux 4.14: more clocking modes, etc. + * 2021/09/03: Porting to Linux 5.4, with enhancements. + * Copyright (C) 2021 Axis Communications AB + * + */ + +#ifndef _ADC3XXX_H +#define _ADC3XXX_H + +/* 8 bit mask value */ +#define ADC3XXX_8BITS_MASK 0xFF + +#define ADC3XXX_PAGE_SIZE 128 +#define ADC3XXX_REG(page, reg) ((page * ADC3XXX_PAGE_SIZE) + reg) + +#define ADC3XXX_MICBIAS_PINS 2 + +/* Number of GPIO pins exposed via the gpiolib interface */ +#define ADC3XXX_GPIOS_MAX 2 + +/* + * PLL modes, to be used for clk_id for set_sysclk callback. + * + * The default behavior (AUTO) is to take the first matching entry in the clock + * table, which is intended to be the PLL based one if there is more than one. + * + * Setting the clock source using simple-card (clocks or + * system-clock-frequency property) sets clk_id = 0 = ADC3XXX_PLL_AUTO. + */ +#define ADC3XXX_PLL_AUTO 0 /* Use first available mode */ +#define ADC3XXX_PLL_ENABLE 1 /* Use PLL for clock generation */ +#define ADC3XXX_PLL_BYPASS 2 /* Don't use PLL for clock generation */ + + +/****************************************************************************/ +/* Page 0 Registers */ +/****************************************************************************/ + +/* Page select register */ +#define ADC3XXX_PAGE_SELECT ADC3XXX_REG(0, 0) +/* Software reset register */ +#define ADC3XXX_RESET ADC3XXX_REG(0, 1) + +/* 2-3 Reserved */ + +/* PLL programming register B */ +#define ADC3XXX_CLKGEN_MUX ADC3XXX_REG(0, 4) +/* PLL P and R-Val */ +#define ADC3XXX_PLL_PROG_PR ADC3XXX_REG(0, 5) +/* PLL J-Val */ +#define ADC3XXX_PLL_PROG_J ADC3XXX_REG(0, 6) +/* PLL D-Val MSB */ +#define ADC3XXX_PLL_PROG_D_MSB ADC3XXX_REG(0, 7) +/* PLL D-Val LSB */ +#define ADC3XXX_PLL_PROG_D_LSB ADC3XXX_REG(0, 8) + +/* 9-17 Reserved */ + +/* ADC NADC */ +#define ADC3XXX_ADC_NADC ADC3XXX_REG(0, 18) +/* ADC MADC */ +#define ADC3XXX_ADC_MADC ADC3XXX_REG(0, 19) +/* ADC AOSR */ +#define ADC3XXX_ADC_AOSR ADC3XXX_REG(0, 20) +/* ADC IADC */ +#define ADC3XXX_ADC_IADC ADC3XXX_REG(0, 21) + +/* 23-24 Reserved */ + +/* CLKOUT MUX */ +#define ADC3XXX_CLKOUT_MUX ADC3XXX_REG(0, 25) +/* CLOCKOUT M divider value */ +#define ADC3XXX_CLKOUT_M_DIV ADC3XXX_REG(0, 26) +/*Audio Interface Setting Register 1*/ +#define ADC3XXX_INTERFACE_CTRL_1 ADC3XXX_REG(0, 27) +/* Data Slot Offset (Ch_Offset_1) */ +#define ADC3XXX_CH_OFFSET_1 ADC3XXX_REG(0, 28) +/* ADC interface control 2 */ +#define ADC3XXX_INTERFACE_CTRL_2 ADC3XXX_REG(0, 29) +/* BCLK N Divider */ +#define ADC3XXX_BCLK_N_DIV ADC3XXX_REG(0, 30) +/* Secondary audio interface control 1 */ +#define ADC3XXX_INTERFACE_CTRL_3 ADC3XXX_REG(0, 31) +/* Secondary audio interface control 2 */ +#define ADC3XXX_INTERFACE_CTRL_4 ADC3XXX_REG(0, 32) +/* Secondary audio interface control 3 */ +#define ADC3XXX_INTERFACE_CTRL_5 ADC3XXX_REG(0, 33) +/* I2S sync */ +#define ADC3XXX_I2S_SYNC ADC3XXX_REG(0, 34) + +/* 35 Reserved */ + +/* ADC flag register */ +#define ADC3XXX_ADC_FLAG ADC3XXX_REG(0, 36) +/* Data slot offset 2 (Ch_Offset_2) */ +#define ADC3XXX_CH_OFFSET_2 ADC3XXX_REG(0, 37) +/* I2S TDM control register */ +#define ADC3XXX_I2S_TDM_CTRL ADC3XXX_REG(0, 38) + +/* 39-41 Reserved */ + +/* Interrupt flags (overflow) */ +#define ADC3XXX_INTR_FLAG_1 ADC3XXX_REG(0, 42) +/* Interrupt flags (overflow) */ +#define ADC3XXX_INTR_FLAG_2 ADC3XXX_REG(0, 43) + +/* 44 Reserved */ + +/* Interrupt flags ADC */ +#define ADC3XXX_INTR_FLAG_ADC1 ADC3XXX_REG(0, 45) + +/* 46 Reserved */ + +/* Interrupt flags ADC */ +#define ADC3XXX_INTR_FLAG_ADC2 ADC3XXX_REG(0, 47) +/* INT1 interrupt control */ +#define ADC3XXX_INT1_CTRL ADC3XXX_REG(0, 48) +/* INT2 interrupt control */ +#define ADC3XXX_INT2_CTRL ADC3XXX_REG(0, 49) + +/* 50 Reserved */ + +/* DMCLK/GPIO2 control */ +#define ADC3XXX_GPIO2_CTRL ADC3XXX_REG(0, 51) +/* DMDIN/GPIO1 control */ +#define ADC3XXX_GPIO1_CTRL ADC3XXX_REG(0, 52) +/* DOUT Control */ +#define ADC3XXX_DOUT_CTRL ADC3XXX_REG(0, 53) + +/* 54-56 Reserved */ + +/* ADC sync control 1 */ +#define ADC3XXX_SYNC_CTRL_1 ADC3XXX_REG(0, 57) +/* ADC sync control 2 */ +#define ADC3XXX_SYNC_CTRL_2 ADC3XXX_REG(0, 58) +/* ADC CIC filter gain control */ +#define ADC3XXX_CIC_GAIN_CTRL ADC3XXX_REG(0, 59) + +/* 60 Reserved */ + +/* ADC processing block selection */ +#define ADC3XXX_PRB_SELECT ADC3XXX_REG(0, 61) +/* Programmable instruction mode control bits */ +#define ADC3XXX_INST_MODE_CTRL ADC3XXX_REG(0, 62) + +/* 63-79 Reserved */ + +/* Digital microphone polarity control */ +#define ADC3XXX_MIC_POLARITY_CTRL ADC3XXX_REG(0, 80) +/* ADC Digital */ +#define ADC3XXX_ADC_DIGITAL ADC3XXX_REG(0, 81) +/* ADC Fine Gain Adjust */ +#define ADC3XXX_ADC_FGA ADC3XXX_REG(0, 82) +/* Left ADC Channel Volume Control */ +#define ADC3XXX_LADC_VOL ADC3XXX_REG(0, 83) +/* Right ADC Channel Volume Control */ +#define ADC3XXX_RADC_VOL ADC3XXX_REG(0, 84) +/* ADC phase compensation */ +#define ADC3XXX_ADC_PHASE_COMP ADC3XXX_REG(0, 85) +/* Left Channel AGC Control Register 1 */ +#define ADC3XXX_LEFT_CHN_AGC_1 ADC3XXX_REG(0, 86) +/* Left Channel AGC Control Register 2 */ +#define ADC3XXX_LEFT_CHN_AGC_2 ADC3XXX_REG(0, 87) +/* Left Channel AGC Control Register 3 */ +#define ADC3XXX_LEFT_CHN_AGC_3 ADC3XXX_REG(0, 88) +/* Left Channel AGC Control Register 4 */ +#define ADC3XXX_LEFT_CHN_AGC_4 ADC3XXX_REG(0, 89) +/* Left Channel AGC Control Register 5 */ +#define ADC3XXX_LEFT_CHN_AGC_5 ADC3XXX_REG(0, 90) +/* Left Channel AGC Control Register 6 */ +#define ADC3XXX_LEFT_CHN_AGC_6 ADC3XXX_REG(0, 91) +/* Left Channel AGC Control Register 7 */ +#define ADC3XXX_LEFT_CHN_AGC_7 ADC3XXX_REG(0, 92) +/* Left AGC gain */ +#define ADC3XXX_LEFT_AGC_GAIN ADC3XXX_REG(0, 93) +/* Right Channel AGC Control Register 1 */ +#define ADC3XXX_RIGHT_CHN_AGC_1 ADC3XXX_REG(0, 94) +/* Right Channel AGC Control Register 2 */ +#define ADC3XXX_RIGHT_CHN_AGC_2 ADC3XXX_REG(0, 95) +/* Right Channel AGC Control Register 3 */ +#define ADC3XXX_RIGHT_CHN_AGC_3 ADC3XXX_REG(0, 96) +/* Right Channel AGC Control Register 4 */ +#define ADC3XXX_RIGHT_CHN_AGC_4 ADC3XXX_REG(0, 97) +/* Right Channel AGC Control Register 5 */ +#define ADC3XXX_RIGHT_CHN_AGC_5 ADC3XXX_REG(0, 98) +/* Right Channel AGC Control Register 6 */ +#define ADC3XXX_RIGHT_CHN_AGC_6 ADC3XXX_REG(0, 99) +/* Right Channel AGC Control Register 7 */ +#define ADC3XXX_RIGHT_CHN_AGC_7 ADC3XXX_REG(0, 100) +/* Right AGC gain */ +#define ADC3XXX_RIGHT_AGC_GAIN ADC3XXX_REG(0, 101) + +/* 102-127 Reserved */ + +/****************************************************************************/ +/* Page 1 Registers */ +/****************************************************************************/ +/* 1-25 Reserved */ + +/* Dither control */ +#define ADC3XXX_DITHER_CTRL ADC3XXX_REG(1, 26) + +/* 27-50 Reserved */ + +/* MICBIAS Configuration Register */ +#define ADC3XXX_MICBIAS_CTRL ADC3XXX_REG(1, 51) +/* Left ADC input selection for Left PGA */ +#define ADC3XXX_LEFT_PGA_SEL_1 ADC3XXX_REG(1, 52) + +/* 53 Reserved */ + +/* Right ADC input selection for Left PGA */ +#define ADC3XXX_LEFT_PGA_SEL_2 ADC3XXX_REG(1, 54) +/* Right ADC input selection for right PGA */ +#define ADC3XXX_RIGHT_PGA_SEL_1 ADC3XXX_REG(1, 55) + +/* 56 Reserved */ + +/* Right ADC input selection for right PGA */ +#define ADC3XXX_RIGHT_PGA_SEL_2 ADC3XXX_REG(1, 57) + +/* 58 Reserved */ + +/* Left analog PGA settings */ +#define ADC3XXX_LEFT_APGA_CTRL ADC3XXX_REG(1, 59) +/* Right analog PGA settings */ +#define ADC3XXX_RIGHT_APGA_CTRL ADC3XXX_REG(1, 60) +/* ADC Low current Modes */ +#define ADC3XXX_LOW_CURRENT_MODES ADC3XXX_REG(1, 61) +/* ADC analog PGA flags */ +#define ADC3XXX_ANALOG_PGA_FLAGS ADC3XXX_REG(1, 62) + +/* 63-127 Reserved */ + +/****************************************************************************/ +/* Macros and definitions */ +/****************************************************************************/ + +#define ADC3XXX_RATES SNDRV_PCM_RATE_8000_96000 +#define ADC3XXX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/****************************************************************************/ +/* ADC3XXX Register bits */ +/****************************************************************************/ +/* PLL Enable bits */ +#define ADC3XXX_ENABLE_PLL_SHIFT 7 +#define ADC3XXX_ENABLE_PLL (1 << ADC3XXX_ENABLE_PLL_SHIFT) +#define ADC3XXX_ENABLE_NADC_SHIFT 7 +#define ADC3XXX_ENABLE_NADC (1 << ADC3XXX_ENABLE_NADC_SHIFT) +#define ADC3XXX_ENABLE_MADC_SHIFT 7 +#define ADC3XXX_ENABLE_MADC (1 << ADC3XXX_ENABLE_MADC_SHIFT) +#define ADC3XXX_ENABLE_BCLK_SHIFT 7 +#define ADC3XXX_ENABLE_BCLK (1 << ADC3XXX_ENABLE_BCLK_SHIFT) + +/* Power bits */ +#define ADC3XXX_LADC_PWR_ON 0x80 +#define ADC3XXX_RADC_PWR_ON 0x40 + +#define ADC3XXX_SOFT_RESET 0x01 +#define ADC3XXX_BCLK_MASTER 0x08 +#define ADC3XXX_WCLK_MASTER 0x04 + +/* Interface register masks */ +#define ADC3XXX_FORMAT_MASK 0xc0 +#define ADC3XXX_FORMAT_SHIFT 6 +#define ADC3XXX_WLENGTH_MASK 0x30 +#define ADC3XXX_WLENGTH_SHIFT 4 +#define ADC3XXX_CLKDIR_MASK 0x0c +#define ADC3XXX_CLKDIR_SHIFT 2 + +/* Interface register bit patterns */ +#define ADC3XXX_FORMAT_I2S (0 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_DSP (1 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_RJF (2 << ADC3XXX_FORMAT_SHIFT) +#define ADC3XXX_FORMAT_LJF (3 << ADC3XXX_FORMAT_SHIFT) + +#define ADC3XXX_IFACE_16BITS (0 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_20BITS (1 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_24BITS (2 << ADC3XXX_WLENGTH_SHIFT) +#define ADC3XXX_IFACE_32BITS (3 << ADC3XXX_WLENGTH_SHIFT) + +/* PLL P/R bit offsets */ +#define ADC3XXX_PLLP_SHIFT 4 +#define ADC3XXX_PLLR_SHIFT 0 +#define ADC3XXX_PLL_PR_MASK 0x7f +#define ADC3XXX_PLLJ_MASK 0x3f +#define ADC3XXX_PLLD_MSB_MASK 0x3f +#define ADC3XXX_PLLD_LSB_MASK 0xff +#define ADC3XXX_NADC_MASK 0x7f +#define ADC3XXX_MADC_MASK 0x7f +#define ADC3XXX_AOSR_MASK 0xff +#define ADC3XXX_IADC_MASK 0xff +#define ADC3XXX_BDIV_MASK 0x7f + +/* PLL_CLKIN bits */ +#define ADC3XXX_PLL_CLKIN_SHIFT 2 +#define ADC3XXX_PLL_CLKIN_MCLK 0x0 +#define ADC3XXX_PLL_CLKIN_BCLK 0x1 +#define ADC3XXX_PLL_CLKIN_ZERO 0x3 + +/* CODEC_CLKIN bits */ +#define ADC3XXX_CODEC_CLKIN_SHIFT 0 +#define ADC3XXX_CODEC_CLKIN_MCLK 0x0 +#define ADC3XXX_CODEC_CLKIN_BCLK 0x1 +#define ADC3XXX_CODEC_CLKIN_PLL_CLK 0x3 + +#define ADC3XXX_USE_PLL ((ADC3XXX_PLL_CLKIN_MCLK << ADC3XXX_PLL_CLKIN_SHIFT) | \ + (ADC3XXX_CODEC_CLKIN_PLL_CLK << ADC3XXX_CODEC_CLKIN_SHIFT)) +#define ADC3XXX_NO_PLL ((ADC3XXX_PLL_CLKIN_ZERO << ADC3XXX_PLL_CLKIN_SHIFT) | \ + (ADC3XXX_CODEC_CLKIN_MCLK << ADC3XXX_CODEC_CLKIN_SHIFT)) + +/* Analog PGA control bits */ +#define ADC3XXX_LPGA_MUTE 0x80 +#define ADC3XXX_RPGA_MUTE 0x80 + +#define ADC3XXX_LPGA_GAIN_MASK 0x7f +#define ADC3XXX_RPGA_GAIN_MASK 0x7f + +/* ADC current modes */ +#define ADC3XXX_ADC_LOW_CURR_MODE 0x01 + +/* Left ADC Input selection bits */ +#define ADC3XXX_LCH_SEL1_SHIFT 0 +#define ADC3XXX_LCH_SEL2_SHIFT 2 +#define ADC3XXX_LCH_SEL3_SHIFT 4 +#define ADC3XXX_LCH_SEL4_SHIFT 6 + +#define ADC3XXX_LCH_SEL1X_SHIFT 0 +#define ADC3XXX_LCH_SEL2X_SHIFT 2 +#define ADC3XXX_LCH_SEL3X_SHIFT 4 +#define ADC3XXX_LCH_COMMON_MODE 0x40 +#define ADC3XXX_BYPASS_LPGA 0x80 + +/* Right ADC Input selection bits */ +#define ADC3XXX_RCH_SEL1_SHIFT 0 +#define ADC3XXX_RCH_SEL2_SHIFT 2 +#define ADC3XXX_RCH_SEL3_SHIFT 4 +#define ADC3XXX_RCH_SEL4_SHIFT 6 + +#define ADC3XXX_RCH_SEL1X_SHIFT 0 +#define ADC3XXX_RCH_SEL2X_SHIFT 2 +#define ADC3XXX_RCH_SEL3X_SHIFT 4 +#define ADC3XXX_RCH_COMMON_MODE 0x40 +#define ADC3XXX_BYPASS_RPGA 0x80 + +/* MICBIAS control bits */ +#define ADC3XXX_MICBIAS_MASK 0x2 +#define ADC3XXX_MICBIAS1_SHIFT 5 +#define ADC3XXX_MICBIAS2_SHIFT 3 + +#define ADC3XXX_ADC_MAX_VOLUME 64 +#define ADC3XXX_ADC_POS_VOL 24 + +/* GPIO control bits (GPIO1_CTRL and GPIO2_CTRL) */ +#define ADC3XXX_GPIO_CTRL_CFG_MASK 0x3c +#define ADC3XXX_GPIO_CTRL_CFG_SHIFT 2 +#define ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_MASK 0x01 +#define ADC3XXX_GPIO_CTRL_OUTPUT_CTRL_SHIFT 0 +#define ADC3XXX_GPIO_CTRL_INPUT_VALUE_MASK 0x02 +#define ADC3XXX_GPIO_CTRL_INPUT_VALUE_SHIFT 1 + +#endif /* _ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 2

only have sound after mute and unmute and set volume in alsamixer
by GitHub issues - opened 01 Nov '21

01 Nov '21

alsa-project/alsa-ucm-conf issue #113 was opened from amazingfate: I have a x86 thinclient with debian bullseye installed. It has no sound in my headphone after boot, but I open alsamixer and then: 1, press M to mute the master 2, press M again to unmute the master 3, change the master volume from 100 to 99 After the above three steps I can hear the sound from my headphone. Here is the alsainfo before and after the three steps: [alsa-info_nosound.txt](https://github.com/alsa-project/alsa-ucm-conf/files/… [alsa-info_sound.txt](https://github.com/alsa-project/alsa-ucm-conf/files/74… I can see the difference in "Amp-Out vals". Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/113 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 01 Nov '21

01 Nov '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 ++++ 2 files changed, 165 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..c4fed6335230 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,137 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + '#sound-dai-cells': + const: 0 + + '#gpio-cells': + const: 2 + + gpio-controller: true + + reset-gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,micbias1-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS1 pin + + ti,micbias2-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_MICBIAS_OFF - Mic bias is powered down + - 1 # ADC3XXX_MICBIAS_2_0V - Mic bias is set to 2.0V + - 2 # ADC3XXX_MICBIAS_2_5V - Mic bias is set to 2.5V + - 3 # ADC3XXX_MICBIAS_AVDD - Mic bias is same as AVDD supply + default: 0 + description: | + Mic bias voltage output on MICBIAS2 pin + +required: + - compatible + - reg + - clocks + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + reset-gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + gpio-controller; + #gpio-cells = <2>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + ti,micbias1-vg = <ADC3XXX_MICBIAS_AVDD>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..59ed510e6cf1 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Devicetree bindings definitions for tlv320adc3xxx driver. + * + * Copyright (C) 2021 Axis Communications AB + */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#define ADC3XXX_MICBIAS_OFF 0 /* Micbias pin powered off */ +#define ADC3XXX_MICBIAS_2_0V 1 /* Micbias pin set to 2.0V */ +#define ADC3XXX_MICBIAS_2_5V 2 /* Micbias pin set to 2.5V */ +#define ADC3XXX_MICBIAS_AVDD 3 /* Use AVDD voltage for micbias pin */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[PATCH v2 0/2] Codec driver for TLV320ADC3001 and 3101
by Ricard Wanderlof 01 Nov '21

01 Nov '21

This is a codec driver for the TLV320ADC3001 and TLV320ADC3101 2 channel audio ADC chips from Texas Instruments. Based on an old driver for these chips that I was referred to by Texas Instruments, but which apparently was never upstreamed, I've taken patches from various incarnations, ported them and augmented them with various new features in order to make a working driver. There is currently no support for the on-chip "miniDSP". Although the TLV320ADC3001 and TLV320ADC3101 have similarities with other TI codec chips for which there are existing drivers, upon closer inspection I decided the differences were too great to warrant attempting to modify an existing driver, especially without access to all supported chips and resources for testing them all. There are certainly things that could be improved, among others in the area of PLL register value calculations, but it has been tested on the Axis ARTPEC-8 platform and has been found to work satisfactorily at least with a TLV320ADC3101 operating as an I2S slave at 48 kHz with 32 bit bit depth, 1 or 2 channels, both with the on-chip PLL enabled and disabled. I have not tested this on a TLV320ADC3001 chip; in the original driver the only differences were related to the now-removed initialization code, so I've just retained the compatible strings for both chips. Limitations: - GPIO pins on chip can only be set up as outputs. - Using BCLK as an input to clock chain is supported by the hw but not by driver. Large portions of the driver have been reworked from v1 based on remarks from Mark. More specifically: - Remove an unneeded #ifdef and EXPORT_SYMBOL_GPL. - Add ADC3XXX_ prefix to all macro names to avoid namespace contention. - Clean up certain macro names with mixed lower case - upper case names. - Do away with the D0..D7 macros representing bit values. - Fix general formatting: spaces before and after {, }, replace multi spaces with tabs, maximum 2 blank lines anywhere, and other minor issues. - Page register does not need to be volatile. - Rather than use 0 / -6dB enum for input attenuator, use standard TLV. - The dither offset control contains reserved values so use VALUE_ENUM instead of straight ENUM. - Replace vector of SOC_ENUMs with individual variables. - Fix ALSA control names: - Use Volume at the end of all gain controls, and remove "Gain" where applicable. - Use "Switch" at the end of all switch controls, either adding it or changing "switch" into "Switch" as applicable. - Switches and Volumes related to capture should be set up as such, so add "Capture" to widget names where appropriate. This is mostly important for the PGA Capture and Switch widgets, not just for consistency, but also because amixer simple Switch controls expect different names (cap/nocap instead of on/off) for control values for capture. - Make widget names have capital first letters for all consitutent words. - When resuming, the registers will not have changed, so no need to call snd_soc_component_cache_sync(). - Move reset gpio init to I2C probe - Remove needless _priv at end of private struct name. - Make hex constants lower case - _CBM_CFM/_CBS_CFS -> _CBP_CFP/_CBC_CFC - Instead of using device specific ALSA controls to control the two GPIO pins, use GPIOLIB. GPIO pin functions still can be set up in the devicetree as the pins can be used for other functions which are not GPIO related. - Limitation: The GPIO pins can only be used as outputs, even though the hardware alternatively supports using them as inputs. - Handle GPIO configs in a vector rather than separately. - Use more appropriate module_i2c_driver instead of generic module_init/module_exit. - Remove useless initialization data. - As the ADC mute function had no control, add it. - Clean up GPIO initialization which was done both in the I2C probe and in the component probe. - Remove probe, remove, suspend and resume callbacks, which either do not do anything at all anymore, or uselessly just do set_bias_level. - The PLL should not be configured in the devicetree, so move the corresponding definitions to tlv320adc3xxx.h where they can be picked up by a machine driver if needed. - Default to PLL AUTO mode. - Use "reset-gpios" instead of too-generic "gpios" for dt property for configuring codec reset pin. - Use function for printing PLL mode rather than macro. - dev_info is too verbose for most of the printouts, change them to dev_dbg. - Add support for configuring micbias voltage in DT. - Remove ALSA control for mic bias. - Move PLL and NADC, MADC divisor enable to DAPM, including 10 ms post PLL start up delay via POST_PMU event. - ADC is powered on using DAPM, so needless to do it in set_bias_level, thus remove it. - Let DAPM handle BCLK divisor enable. With this, we can remove set_bias_level as it doesn't do anything anymore. - Fix incorrect argument order to a snd_soc_component_update_bits call. - Although the chip itself can operate using BCLK as the clock source, either directly to the divider chain or via the PLL, this is not yet supported in the driver, so require the clocks property for the MCLK for now. - Move PLL rate structure out of .h file where it does not belong. /Ricard Ricard Wanderlof (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 137 +++ include/dt-bindings/sound/tlv320adc3xxx.h | 28 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1071 +++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 379 ++++++ 6 files changed, 1624 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[PATCH] ALSA: usb-audio: Line6 HX-Stomp XL USB_ID
by Jason Ormes 01 Nov '21

01 Nov '21

Adding the Line6 HX-Stomp XL USB_ID as it needs this fixed frequency quirk as well. The device is basically just the HX-Stomp with some more buttons on the face. I've Done some recording with it after adding it, and it seems to function properly with this fix. The Midi features appear to be working as well. Signed-off-by: Jason Ormes <skryking(a)gmail.com> --- sound/usb/format.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 50efccbffb8a..e8fc2c350a9f 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -414,6 +414,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0e41, 0x4242): /* Line6 Helix Rack */ case USB_ID(0x0e41, 0x4244): /* Line6 Helix LT */ case USB_ID(0x0e41, 0x4246): /* Line6 HX-Stomp */ + case USB_ID(0x0e41, 0x4253): /* Line6 HX-Stomp XL*/ case USB_ID(0x0e41, 0x4247): /* Line6 Pod Go */ case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */ case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ -- 2.25.1

2 1

[PATCH] ALSA: usb-audio: Add registration quirk for JBL Quantum 400
by Alexander Tsoy 01 Nov '21

01 Nov '21

Add another device ID for JBL Quantum 400. It requires the same quirk as other JBL Quantum devices. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 8929d9abe8aa..1d418c3fc598 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1749,6 +1749,7 @@ static const struct registration_quirk registration_quirks[] = { REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ REG_QUIRK_ENTRY(0x0ecb, 0x1f47, 2), /* JBL Quantum 800 */ + REG_QUIRK_ENTRY(0x0ecb, 0x1f4c, 2), /* JBL Quantum 400 */ REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ -- 2.32.0

2 1

[PATCH v1 0/4] Add tfa9897 speaker/receiver modes switching
by Vincent Knecht 31 Oct '21

31 Oct '21

Two tfa9897 are used in tandem on some devices (eg. alcatel-idol347) to drive two speakers, for both receiver (earpiece) and speaker modes. This series: 1. adds a control to allow enabling only one tfa9897/speaker (for the typical phone-call usecase) 2. adds a control to switch between speaker and receiver modes 3. adds support for an optional gpio also used to switch modes Stephan Gerhold (1): ASoC: codecs: tfa989x: Add switch to allow disabling amplifier Vincent Knecht (3): ASoC: codecs: tfa989x: Add support for tfa9897 RCV bit ASoC: dt-bindings: nxp, tfa989x: Add rcv-gpios property for tfa9897 ASoC: codecs: tfa989x: Add support for tfa9897 optional rcv-gpios .../bindings/sound/nxp,tfa989x.yaml | 10 ++++ sound/soc/codecs/tfa989x.c | 47 ++++++++++++++++++- 2 files changed, 55 insertions(+), 2 deletions(-) -- 2.31.1

4 9

RFT: implementation for MOTU FireWire series including CueMix DSP and CueMix FX
by Takashi Sakamoto 31 Oct '21

31 Oct '21

RFT: implementation of MOTU FireWire series CueMix DSP and CueMix FX Hi all, After investigation for protocols of MOTU FireWire series including features of CueMix DSP and CueMix FX, I've written implementation corresponding to the features in snd-firewire-ctl-services project[1]. If you are enough interested in it, would you please test the implementation? The first aim of this test request is to check whether the protocol implementation works as well out of my development environment. # Disclaimer * The aim of the work is to reveal relevant protocols so adequately and write the implementation. It is out of my scope to produce user-friendly applications such as GUI with cool user-delighted look and feel. (This is also the reason that the message is so long for general users.) # Acknowledgment When I made investigation plan for the protocols and evaluated the result of investigation, documents included in ffado source code[13] is helpful. The documentation includes a part of the result, while some parts are newly revealed in this time. I appreciate for document writers. # Current status * Below models are supported and I tested * 828 * 896 * 828mk2 * 896hd * 8 pre * Ultralite * 4 pre * Audio Express * 828mk3 (FireWire only/Hybrid) * Ultralite mk3 (FireWire only/Hybrid) * The most of control functions are available via ALSA control interface * Hardware metering support * Interaction support between on-board control component such as knob, utilizing notification mechanism of ALSA control interface # Requirements For the test, you need to solve long list of requirements since the protocol implementation is new and lays down user space as well as kernel space. I'm sorry for testers to discourage your motivation, but they are required... * Linux kernel v5.13 or later * a commit 66c6d1ef86ff ('ALSA: control: Add memory consumption limit to user controls')[2] is required since many user-defined control element sets are used. * Installation of backport codes for ALSA firewire stack * They comes from Linux kernel v5.16 prepatch (not released yet). * Please follow to instructions in my remote repository[3] * Installation of alsa-gobject v0.2.1 or later. * alsa-gobject project[4] provides libraries which perform as glue for several kind of language bindings to use UAPI of Linux sound subsystem. For detail, please refer to release announcement[5]. * Installation of libhinawa topic branch * libhinawa project[6] also provides library as glue for UAPI of ALSA firewire stack. * The topic branch (topic/motu-mixer-ioctl)[7] includes new APIs to use functions added in the kernel v5.16 prepatch. * Rust v1.51 or later * I use Rust programming language for the protocol implementation in user space side. * Build topic branch of snd-firewire-ctl-services * snd-firewire-ctl-services project[8] includes the series of implementation for protocols specific to ASICs/Vendors/Models. Additionally, as reference of service programs, it includes runtime implementation based on ALSA control interface and ALSA Sequencer interface. * The topic branch ('topic/motu-mixer-ioctl') includes implementation for the test. # Test procedure When the above requirements are satisfied, the service program can be build by below command line. It takes a bit long than compiling with C language since Rust compiler does more work than C (mainly due to type inference and lifetime check for safe runtime). $ cargo build -p snd-firewire-ctl-services The service program for MOTU FireWire series runs by below command line: $ cargo run --bin snd-firewire-motu-ctl-service CARD_ID The CARD_ID is the numeric ID of sound card corresponding to your device. You need to load ALSA firewire-motu driver in advance: $ sudo modprobe snd-firewire-motu $ cat /proc/asound/cards ... 2 [UltraLiteMk3 ]: FW-MOTU - UltraLiteMk3 MOTU UltraLiteMk3 (version:100800), GUID ... In the above example, CARD_ID should be 2. When the service program runs, you can see control elements via ALSA control interface. You can operate them by usual tools of ALSA control application, but I prefer 'qashctl' in 'qastools' project[11]. Due to specification of protocols defined by MOTU, notifications from the device for some controls are firstly available when isochronous packets are transmitted by the device. Please run any of ALSA PCM applications when testing. The aim of the request is whether the service program can be built and run out of my development environment. So I don't expect you so much. However, If you are interested, please check the other parts about which I concern: * The range of value for each control is valid or not. * Inverse evaluation of each control with boolean value. * The lack of control which should be produced by the service program. # Filing issues * Please use issue service in github.com[10]. * Preferable issues * The service program aborts when I operate control A. * The service program doesn't interact with on-board operation B. * The service program looks to be frozen when I operate C. * The behaviour of control element D is odd. * Unexpected issues * Why device A is not supported * It's preferable for you to take your time to cooperate with me if you wish. It might take a long time since the implementation lays down user space and kernel space. * If so, please file the issue to my remote repository for kernel prepatch[12]. * Why we have no GUI with cool look and feel * It's your work. The GUI application might be an ALSA control application which interacts with the service program. * I can not get which control value corresponds to actual ports * It is restriction under current design of ALSA control interface. We need to work for any extension to the design. * The name of control B is not user-friendly. * The name of controls are not fixed yet since we have few name convention for controls in studio-use equipments. It might corresponds to discussion in ALSA upstream. In my opinion, Linux audio developers have been enough lazy to the issue for recent decade. [1] https://github.com/alsa-project/snd-firewire-ctl-services/ [2] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… [3] https://github.com/takaswie/snd-firewire-improve [4] https://github.com/alsa-project/alsa-gobject/ [5] https://lore.kernel.org/alsa-devel/20200623093239.GA68404@workstation/ [6[ https://github.com/alsa-project/libhinawa [7] https://github.com/alsa-project/libhinawa/tree/topic/motu-mixer-ioctl [8] https://github.com/alsa-project/snd-firewire-ctl-services/ [9] https://github.com/alsa-project/snd-firewire-ctl-services/tree/topic/motu-m… [10] https://github.com/alsa-project/snd-firewire-ctl-services/issues [11] https://gitlab.com/sebholt/qastools [12] https://github.com/takaswie/snd-firewire-improve/issues [13] https://web.archive.org/web/20110704073441/http://subversion.ffado.org/brow… Cheers Takashi Sakamoto

1 0

[PATCH] HX-Stomp XL USB_ID.
by Jason Ormes 30 Oct '21

30 Oct '21

Signed-off-by: Jason Ormes <skryking(a)gmail.com> --- sound/usb/format.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 50efccbffb8a..e8fc2c350a9f 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -414,6 +414,7 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0e41, 0x4242): /* Line6 Helix Rack */ case USB_ID(0x0e41, 0x4244): /* Line6 Helix LT */ case USB_ID(0x0e41, 0x4246): /* Line6 HX-Stomp */ + case USB_ID(0x0e41, 0x4253): /* Line6 HX-Stomp XL*/ case USB_ID(0x0e41, 0x4247): /* Line6 Pod Go */ case USB_ID(0x0e41, 0x4248): /* Line6 Helix >= fw 2.82 */ case USB_ID(0x0e41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ -- 2.25.1

2 3

[PATCH] dt-bindings: sound: gtm601: convert to YAML
by David Heidelberg 30 Oct '21

30 Oct '21

Convert GTM601 binding to the YAML format. Signed-off-by: David Heidelberg <david(a)ixit.cz> --- .../devicetree/bindings/sound/gtm601.txt | 19 ----------- .../devicetree/bindings/sound/gtm601.yaml | 33 +++++++++++++++++++ 2 files changed, 33 insertions(+), 19 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/gtm601.txt create mode 100644 Documentation/devicetree/bindings/sound/gtm601.yaml diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt deleted file mode 100644 index efa32a486c4a..000000000000 --- a/Documentation/devicetree/bindings/sound/gtm601.txt +++ /dev/null @@ -1,19 +0,0 @@ -GTM601 UMTS modem audio interface CODEC - -This device has no configuration interface. The sample rate and channels are -based on the compatible string - "option,gtm601" = 8kHz mono - "broadmobi,bm818" = 48KHz stereo - -Required properties: - - - compatible : one of - "option,gtm601" - "broadmobi,bm818" - - -Example: - -codec: gtm601_codec { - compatible = "option,gtm601"; -}; diff --git a/Documentation/devicetree/bindings/sound/gtm601.yaml b/Documentation/devicetree/bindings/sound/gtm601.yaml new file mode 100644 index 000000000000..804402c72798 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/gtm601.yaml @@ -0,0 +1,33 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/gtm601.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: GTM601 UMTS modem audio interface CODEC + +maintainers: + - Angus Ainslie (Purism) <angus(a)akkea.ca> + +description: > + This device has no configuration interface. The sample rate and channels are + based on the compatible string + +properties: + compatible: + oneOf: + - items: + - const: broadmobi,bm818 + - const: option,gtm601 + - items: + - enum: + - broadmobi,bm818 # 48 kHz stereo + - option,gtm601 # 8 kHz mono + +additionalProperties: false + +examples: + - | + gtm601_codec { + compatible = "option,gtm601"; + }; -- 2.33.0

2 1

[PATCH v4 0/6] Multiple headphone codec driver support
by Brent Lu 29 Oct '21

29 Oct '21

Support multiple headphone drivers in same machine driver. In this case, both rt5682 and rt5682s are supported and enumerated by different ACPI HID "10EC5682" and "RTL5682". V2 Changes: - remove useless 'NULL', 'false' in if-condition - can use 'comp_ids' field alone to enumerate driver - add comma to the end of entry in structure initialization - keep the table of byt/cht/cml/icl untouched V3 Changes: - upstreamd from SOF github, PR#3200 - use new compatiable IDs to shrink the enumerate table of BYT and CHT - add 'const' to snd_soc_acpi_codecs structures V4 Changes: - add signoff to patch 4~6 Brent Lu (3): ASoC: soc-acpi: add comp_ids field for machine driver matching ASoC: Intel: sof_rt5682: detect codec variant in probe function ASoC: Intel: sof_rt5682: use comp_ids to enumerate rt5682s Pierre-Louis Bossart (3): ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs ASoC: Intel: soc-acpi: use const for all uses of snd_soc_acpi_codecs include/sound/soc-acpi.h | 3 + sound/soc/intel/boards/sof_rt5682.c | 34 ++------- .../intel/common/soc-acpi-intel-adl-match.c | 11 ++- .../intel/common/soc-acpi-intel-bxt-match.c | 2 +- .../intel/common/soc-acpi-intel-byt-match.c | 68 +++++++----------- .../intel/common/soc-acpi-intel-cht-match.c | 69 +++++++------------ .../intel/common/soc-acpi-intel-cml-match.c | 8 +-- .../intel/common/soc-acpi-intel-glk-match.c | 2 +- .../intel/common/soc-acpi-intel-jsl-match.c | 43 ++++-------- .../intel/common/soc-acpi-intel-kbl-match.c | 12 ++-- .../intel/common/soc-acpi-intel-skl-match.c | 2 +- .../intel/common/soc-acpi-intel-tgl-match.c | 11 ++- sound/soc/soc-acpi.c | 24 ++++++- 13 files changed, 119 insertions(+), 170 deletions(-) -- 2.25.1

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[PATCH 0/6] ASoC: Sanity checks and soc-topology updates
by Cezary Rojewski 29 Oct '21

29 Oct '21

Couple of soc-topology related changes and a use-after-free fix. Said fix and two sanity checks for soc-topology lead the way. While the use-after-free is quite obvious, the sanity checks are here to cover for cases where user malformed the topology file -or- access to filesystem somehow got interrupted during copy operation. We shouldn't be reading outside the file boundary. Afterward a change to soc_tplg_add_kcontrol(): device being passed to soc_tplg_add_dcontrol() from comp->dev to tplg->dev which corrects dev_xxx() invoked later on. Also, device used for topology memory allocations from component->dev to component->card->dev so memory gets freed each time card device (usually platform device) is removed rather than the component device what may happen less frequently. Dummy component gets smarter and no longer overrides hw_params if there are other components accociated with related struct snd_soc_pcm_runtime instance. Amadeusz Sławiński (5): ASoC: core: Remove invalid snd_soc_component_set_jack call ASoC: topology: Check for dapm widget completeness ASoC: topology: Use correct device for prints ASoC: topology: Change topology device to card device ASoC: Stop dummy from overriding hwparams Cezary Rojewski (1): ASoC: topology: Add header payload_size verification sound/soc/soc-core.c | 3 --- sound/soc/soc-topology.c | 34 ++++++++++++++++++++++++++++++---- sound/soc/soc-utils.c | 13 +++++++++++++ 3 files changed, 43 insertions(+), 7 deletions(-) -- 2.25.1

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[RESEND, v2 0/2] ASoC: mediatek: Add support for MT8195 sound card with rt1011 and rt5682
by Trevor Wu 29 Oct '21

29 Oct '21

This series of patches adds support for mt8195 board with mt6359, rt1011 and rt5682. Patches are based on broonie tree "for-next" branch. Changes since v1: - refined code based on Pierre-Louis's suggestions - added missing of_node_put - fixed wrong fixup assignment on HDMITX Trevor Wu (2): ASoC: mediatek: mt8195: add machine driver with mt6359, rt1011 and rt5682 dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1011-rt5682 document .../sound/mt8195-mt6359-rt1011-rt5682.yaml | 47 + sound/soc/mediatek/Kconfig | 15 + sound/soc/mediatek/mt8195/Makefile | 1 + .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 1155 +++++++++++++++++ 4 files changed, 1218 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1011-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c -- 2.18.0

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[PATCH 0/2] ASoC: mediatek: Add support for MT8195 sound card with rt1011 and rt5682
by Trevor Wu 29 Oct '21

29 Oct '21

This series of patches adds support for mt8195 board with mt6359, rt1011 and rt5682. Patches are based on broonie tree "for-next" branch. Trevor Wu (2): ASoC: mediatek: mt8195: add machine driver with mt6359, rt1011 and rt5682 dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1011-rt5682 document .../sound/mt8195-mt6359-rt1011-rt5682.yaml | 47 + sound/soc/mediatek/Kconfig | 15 + sound/soc/mediatek/mt8195/Makefile | 1 + .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 1140 +++++++++++++++++ 4 files changed, 1203 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1011-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c -- 2.18.0

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[PATCH] ASoC: cs42l42: Add control for audio slow-start switch
by Richard Fitzgerald 29 Oct '21

29 Oct '21

This adds an ALSA control so that the slow-start audio ramp feature can be disabled. This is useful for high-definition audio applications. The register field is unusual in that it is a 3-bit field with only two valid values, 000=off and 111=on. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 32 +++++++++++++++++++++++++++++++- sound/soc/codecs/cs42l42.h | 3 +++ 2 files changed, 34 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 27a1c4c73074..0d562c274292 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -42,6 +42,7 @@ static const struct reg_default cs42l42_reg_defaults[] = { { CS42L42_SRC_CTL, 0x10 }, { CS42L42_MCLK_CTL, 0x02 }, { CS42L42_SFTRAMP_RATE, 0xA4 }, + { CS42L42_SLOW_START_ENABLE, 0x70 }, { CS42L42_I2C_DEBOUNCE, 0x88 }, { CS42L42_I2C_STRETCH, 0x03 }, { CS42L42_I2C_TIMEOUT, 0xB7 }, @@ -177,6 +178,7 @@ static bool cs42l42_readable_register(struct device *dev, unsigned int reg) case CS42L42_MCLK_STATUS: case CS42L42_MCLK_CTL: case CS42L42_SFTRAMP_RATE: + case CS42L42_SLOW_START_ENABLE: case CS42L42_I2C_DEBOUNCE: case CS42L42_I2C_STRETCH: case CS42L42_I2C_TIMEOUT: @@ -387,6 +389,30 @@ static const struct regmap_config cs42l42_regmap = { static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true); static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); +static int cs42l42_slow_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + u8 val; + + /* all bits of SLOW_START_EN much change together */ + switch (ucontrol->value.integer.value[0]) { + case 0: + val = 0; + break; + case 1: + val = CS42L42_SLOW_START_EN_MASK; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, CS42L42_SLOW_START_ENABLE, + CS42L42_SLOW_START_EN_MASK, val); + + return 0; +} + static const char * const cs42l42_hpf_freq_text[] = { "1.86Hz", "120Hz", "235Hz", "466Hz" }; @@ -431,7 +457,11 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DAC_HPF_EN_SHIFT, true, false), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, - 0x3f, 1, mixer_tlv) + 0x3f, 1, mixer_tlv), + + SOC_SINGLE_EXT("Slow Start Switch", CS42L42_SLOW_START_ENABLE, + CS42L42_SLOW_START_EN_SHIFT, true, false, + snd_soc_get_volsw, cs42l42_slow_start_put), }; static int cs42l42_hp_adc_ev(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index f45bcc9a3a62..c8b3267a318b 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -62,6 +62,9 @@ #define CS42L42_INTERNAL_FS_MASK (1 << CS42L42_INTERNAL_FS_SHIFT) #define CS42L42_SFTRAMP_RATE (CS42L42_PAGE_10 + 0x0A) +#define CS42L42_SLOW_START_ENABLE (CS42L42_PAGE_10 + 0x0B) +#define CS42L42_SLOW_START_EN_MASK GENMASK(6, 4) +#define CS42L42_SLOW_START_EN_SHIFT 4 #define CS42L42_I2C_DEBOUNCE (CS42L42_PAGE_10 + 0x0E) #define CS42L42_I2C_STRETCH (CS42L42_PAGE_10 + 0x0F) #define CS42L42_I2C_TIMEOUT (CS42L42_PAGE_10 + 0x10) -- 2.11.0

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AM_PATH_ALSA. error
by GitHub issues - edited 29 Oct '21

29 Oct '21

alsa-project/alsa-utils issue #116 was edited from steeley: Hi, I'm installing on latest raspios buster lite 32bit: (Linux cm432 5.10.63-v7l+ #1459 SMP Wed Oct 6 16:41:57 BST 2021 armv7l GNU/Linux) my build process is: `git clone https://github.com/alsa-project/alsa-utils.git cd alsa-utils/ git checkout v1.1.9 autoreconf -i ./configure --prefix=/usr/local make sudo make install` I have built and installed alsa-libs, and alsa-plugins correctly, but get this error with alsa-utils when I autoreconf: `configure.ac:22: warning: macro 'AM_PATH_ALSA' not found in library configure.ac:22: error: possibly undefined macro: AM_PATH_ALSA If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. autoreconf: /usr/bin/autoconf failed with exit status: 1` How to resolve? TIA Issue URL : https://github.com/alsa-project/alsa-utils/issues/116 Repository URL: https://github.com/alsa-project/alsa-utils

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AM_PATH_ALSA. error
by GitHub issues - edited 29 Oct '21

29 Oct '21

alsa-project/alsa-utils issue #116 was edited from steeley: Hi, I'm installing on latest raspios buster lite 32bit: (Linux cm432 5.10.63-v7l+ #1459 SMP Wed Oct 6 16:41:57 BST 2021 armv7l GNU/Linux) my build process is: `git clone https://github.com/alsa-project/alsa-utils.git cd alsa-utils/ git checkout v1.1.9 autoreconf -i ./configure --prefix=/usr/local make sudo make install` I have built and installed alsa-libs, and alia-plugins correctly, but get this error with alsa-utils when I autoreconf: `configure.ac:22: warning: macro 'AM_PATH_ALSA' not found in library configure.ac:22: error: possibly undefined macro: AM_PATH_ALSA If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. autoreconf: /usr/bin/autoconf failed with exit status: 1` How to resolve? TIA Issue URL : https://github.com/alsa-project/alsa-utils/issues/116 Repository URL: https://github.com/alsa-project/alsa-utils

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AM_PATH_ALSA. error
by GitHub issues - opened 29 Oct '21

29 Oct '21

alsa-project/alsa-utils issue #116 was opened from steeley: Hi, I'm installing on latest raspios buster lite 32bit: (Linux cm432 5.10.63-v7l+ #1459 SMP Wed Oct 6 16:41:57 BST 2021 armv7l GNU/Linux) my build process is: `git clone https://github.com/alsa-project/alsa-utils.git cd alsa-utils/ git checkout v1.1.9 autoreconf -i ./configure --prefix=/usr/local make sudo make install` I have built and installed alsa-libs correctly, but get this error with alas-utils when I autoreconf: `configure.ac:22: warning: macro 'AM_PATH_ALSA' not found in library configure.ac:22: error: possibly undefined macro: AM_PATH_ALSA If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. autoreconf: /usr/bin/autoconf failed with exit status: 1` How to resolve? TIA Issue URL : https://github.com/alsa-project/alsa-utils/issues/116 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ASoC: amd: acp: select CONFIG_SND_SOC_ACPI
by Arnd Bergmann 29 Oct '21

29 Oct '21

From: Arnd Bergmann <arnd(a)arndb.de> The acp-platform driver now needs the ACPI helpers: ld.lld: error: undefined symbol: snd_soc_acpi_find_machine >>> referenced by acp-platform.c >>> soc/amd/acp/acp-platform.o:(acp_machine_select) in archive sound/built-in.a ld.lld: error: undefined symbol: snd_soc_acpi_codec_list >>> referenced by acp-renoir.c >>> soc/amd/acp/acp-renoir.o:(snd_soc_acpi_amd_acp_machines) in archive sound/built-in.a Other drivers using this interface, select SND_SOC_ACPI, so do the same thing here. Fixes: e646b51f5dd5 ("ASoC: amd: acp: Add callback for machine driver on ACP") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/amd/acp/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index e3d5cf993f5f..52a1371f9e61 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -20,6 +20,7 @@ config SND_SOC_AMD_ACP_I2S config SND_SOC_AMD_ACP_PCM tristate + select SND_SOC_ACPI if ACPI config SND_AMD_ASOC_RENOIR tristate "AMD ACP ASOC Renoir Support" -- 2.29.2

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[PATCH] ASoC: es8316: add support for ESSX8336 ACPI _HID
by Bard Liao 29 Oct '21

29 Oct '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The same codec seems to have different personalities. ESSX8316 was used for Baytrail/CherryTrail, ESSX8336 seems to be used for AppoloLake, GeminiLake, JasperLake and TigerLake devices. BugLink: https://github.com/thesofproject/linux/issues/2955 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>-e Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/codecs/es8316.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 5fb02635c140..8f30a3ea8bfe 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -840,6 +840,7 @@ MODULE_DEVICE_TABLE(of, es8316_of_match); #ifdef CONFIG_ACPI static const struct acpi_device_id es8316_acpi_match[] = { {"ESSX8316", 0}, + {"ESSX8336", 0}, {}, }; MODULE_DEVICE_TABLE(acpi, es8316_acpi_match); -- 2.17.1

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[PATCH] ASoC: fix unmet dependencies on GPIOLIB for SND_SOC_RT1015P
by Julian Braha 29 Oct '21

29 Oct '21

When SND_SOC_MT8192_MT6359_RT1015_RT5682, SND_SOC_MT8192_MT6359_RT1015_RT5682, SND_SOC_MT8183_DA7219_MAX98357A, or SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A is selected, and GPIOLIB is not selected, Kbuild gives the following warnings, respectively: WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_MT8192_MT6359_RT1015_RT5682 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8192 [=y] && MTK_PMIC_WRAP [=y] WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_MT8192_MT6359_RT1015_RT5682 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8192 [=y] && MTK_PMIC_WRAP [=y] WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_MT8183_DA7219_MAX98357A [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_MT8183 [=y] && I2C [=y] WARNING: unmet direct dependencies detected for SND_SOC_RT1015P Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8183 [=y] This is because these config options select SND_SOC_RT1015P without selecting or depending on GPIOLIB, despite SND_SOC_RT1015P depending on GPIOLIB. These unmet dependency bugs were detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha <julianbraha(a)gmail.com> --- sound/soc/mediatek/Kconfig | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 81ad2dcee9eb..d10d6a3bf5eb 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -120,7 +120,7 @@ config SND_SOC_MT8183 config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A tristate "ASoC Audio driver for MT8183 with MT6358 TS3A227E MAX98357A RT1015 codec" - depends on I2C + depends on I2C && GPIOLIB depends on SND_SOC_MT8183 select SND_SOC_MT6358 select SND_SOC_MAX98357A @@ -138,7 +138,7 @@ config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A config SND_SOC_MT8183_DA7219_MAX98357A tristate "ASoC Audio driver for MT8183 with DA7219 MAX98357A RT1015 codec" - depends on SND_SOC_MT8183 && I2C + depends on SND_SOC_MT8183 && I2C && GPIOLIB select SND_SOC_MT6358 select SND_SOC_MAX98357A select SND_SOC_RT1015 @@ -173,7 +173,7 @@ config SND_SOC_MT8192 config SND_SOC_MT8192_MT6359_RT1015_RT5682 tristate "ASoC Audio driver for MT8192 with MT6359 RT1015 RT5682 codec" - depends on I2C + depends on I2C && GPIOLIB depends on SND_SOC_MT8192 && MTK_PMIC_WRAP select SND_SOC_MT6359 select SND_SOC_RT1015 @@ -200,7 +200,7 @@ config SND_SOC_MT8195 config SND_SOC_MT8195_MT6359_RT1019_RT5682 tristate "ASoC Audio driver for MT8195 with MT6359 RT1019 RT5682 codec" - depends on I2C + depends on I2C && GPIOLIB depends on SND_SOC_MT8195 && MTK_PMIC_WRAP select SND_SOC_MT6359 select SND_SOC_RT1015P -- 2.30.2

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