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

[PATCH] ALSA: hda: Fix driver index handling at re-binding
by Takashi Iwai 09 Feb '22

09 Feb '22

HD-audio driver handles the multiple instances and keeps the static index that is incremented at each probe. This becomes a problem when user tries to re-bind the device via sysfs multiple times; as the device index isn't cleared unlike rmmod case, it points to the next element at re-binding, and eventually later you can't probe any more when it reaches to SNDRV_CARDS_MAX (usually 32). This patch is an attempt to improve the handling at rebinding. Instead of a static device index, now we keep a bitmap and assigns to the first zero bit position. At the driver remove, in return, the bitmap slot is cleared again, so that it'll be available for the next probe. Reported-by: Alexander Sergeyev <sergeev917(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4b0338c4c543..a2922233e85f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2064,14 +2064,16 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; +static DECLARE_BITMAP(probed_devs, SNDRV_CARDS); + static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { - static int dev; struct snd_card *card; struct hda_intel *hda; struct azx *chip; bool schedule_probe; + int dev; int err; if (pci_match_id(driver_denylist, pci)) { @@ -2079,10 +2081,11 @@ static int azx_probe(struct pci_dev *pci, return -ENODEV; } + dev = find_first_zero_bit(probed_devs, SNDRV_CARDS); if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { - dev++; + set_bit(dev, probed_devs); return -ENOENT; } @@ -2149,7 +2152,7 @@ static int azx_probe(struct pci_dev *pci, if (schedule_probe) schedule_delayed_work(&hda->probe_work, 0); - dev++; + set_bit(dev, probed_devs); if (chip->disabled) complete_all(&hda->probe_wait); return 0; @@ -2372,6 +2375,7 @@ static void azx_remove(struct pci_dev *pci) cancel_delayed_work_sync(&hda->probe_work); device_lock(&pci->dev); + clear_bit(chip->dev_index, probed_devs); pci_set_drvdata(pci, NULL); snd_card_free(card); } -- 2.34.1

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[PATCH v1 0/6] Add Tegra234 HDA support
by Mohan Kumar 09 Feb '22

09 Feb '22

This series add the support for TEGRA234 HDA driver support Mohan Kumar (6): ALSA: hda/tegra: Add Tegra234 hda driver support ALSA: hda/tegra: Hardcode GCAP ISS value on T234 ALSA: hda/tegra: Update scratch reg. communication dt-bindings: Add HDA support for Tegra234 dt-bindings: Document Tegra234 HDA support arm64: tegra: Add hda dts node for Tegra234 .../bindings/sound/nvidia,tegra30-hda.yaml | 3 + .../nvidia/tegra234-p3737-0000+p3701-0000.dts | 6 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 18 +++ include/dt-bindings/clock/tegra234-clock.h | 4 + include/dt-bindings/memory/tegra234-mc.h | 6 + .../dt-bindings/power/tegra234-powergate.h | 9 ++ include/dt-bindings/reset/tegra234-reset.h | 2 + include/sound/hda_codec.h | 4 + sound/pci/hda/hda_tegra.c | 33 ++++- sound/pci/hda/patch_hdmi.c | 115 ++++++++++++++---- 10 files changed, 176 insertions(+), 24 deletions(-) create mode 100644 include/dt-bindings/power/tegra234-powergate.h -- 2.17.1

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[PATCH 1/4] ALSA: hda/realtek: fix mute/micmute LEDs for HP 855 G8
by Jeremy Szu 08 Feb '22

08 Feb '22

The HP EliteBook 855 G8 Notebook PC is using ALC285 codec which needs ALC285_FIXUP_HP_MUTE_LED fixup to make it works. After applying the fixup, the mute/micmute LEDs work good. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 552e2cb73291..9d68f591c6bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8291,6 +8291,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.31.1

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[PATCH] ASoC: fsl_sai: Enable combine mode soft
by Sascha Hauer 08 Feb '22

08 Feb '22

The fsl_sai driver calculates the number of pins used and enables multiple channels if necessary. This means the SAI expects data in one FIFO per pin. The SDMA engine only services a single FIFO, so multi pin support doesn't work at all. This patch enables the software combine mode in chips that support it. With this the SAI presents only a single FIFO to the outside and distributes the data into the different FIFOs internally. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/fsl/fsl_sai.c | 9 +++++++++ sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 10 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 10544fa27dc0..cab015d96889 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -517,6 +517,10 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } + if (sai->soc_data->pins > 1) + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), + FSL_SAI_CR4_FCOMB_MASK, FSL_SAI_CR4_FCOMB_SOFT); + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), FSL_SAI_CR3_TRCE_MASK, FSL_SAI_CR3_TRCE((1 << pins) - 1)); @@ -1195,6 +1199,7 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .use_imx_pcm = false, .use_edma = false, .fifo_depth = 32, + .pins = 1, .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, @@ -1204,6 +1209,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .use_imx_pcm = true, .use_edma = false, .fifo_depth = 32, + .pins = 1, .reg_offset = 0, .mclk0_is_mclk1 = true, .flags = 0, @@ -1213,6 +1219,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .use_imx_pcm = true, .use_edma = false, .fifo_depth = 16, + .pins = 2, .reg_offset = 8, .mclk0_is_mclk1 = false, .flags = PMQOS_CPU_LATENCY, @@ -1222,6 +1229,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .use_imx_pcm = true, .use_edma = false, .fifo_depth = 128, + .pins = 8, .reg_offset = 8, .mclk0_is_mclk1 = false, .flags = 0, @@ -1231,6 +1239,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_imx_pcm = true, .use_edma = true, .fifo_depth = 64, + .pins = 1, .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 9aaf231bc024..410f6e6a9137 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -223,6 +223,7 @@ struct fsl_sai_soc_data { bool use_edma; bool mclk0_is_mclk1; unsigned int fifo_depth; + unsigned int pins; unsigned int reg_offset; unsigned int flags; }; -- 2.30.2

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[PATCH v1 0/6] ASoC: Fix gpiolib dependencies
by Mark Brown 08 Feb '22

08 Feb '22

This series removes a bunch of spurious selects of gpiolib that were causing noise in randconfig build tests. Mark Brown (6): ASoC: dmic: Remove spurious gpiolib select ASoC: rt9120: Remove spurious gpiolib select ASoC: simple-amplifier: Remove spurious gpiolib select ASoC: max9759: Remove spurious gpiolib select ASoC: zl38060: Remove spurious gpiolib select ASoC: simple-mux: Depend on gpiolib rather than selecting it sound/soc/codecs/Kconfig | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) base-commit: e783362eb54cd99b2cac8b3a9aeac942e6f6ac07 -- 2.30.2

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[PATCH v3] ASoC: SOF: compr: Add compress ops implementation
by Daniel Baluta 08 Feb '22

08 Feb '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> Implement snd_compress_ops. There are a lot of similarities with PCM implementation. For now we use sof_ipc_pcm_params to transfer compress parameters to SOF firmware. This will be changed in the future once we either add new compress parameters to SOF or enhance existing sof_ipc_pcm_params structure to support all native compress params. Note that get_caps and get_codec_caps are missing and will be added later. This is because we need to find a way to advertise DSP capabilities depending on supported platforms. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Changes since v2: - dropped patch 2/2 because deciding on how to support get_caps and get_codec_caps needs some rework. - annotated commit message for 1/2 to mention about get_caps and get_codec_caps Changes since v1: (https://lore.kernel.org/lkml/20220113161341.371345-1-daniel.baluta@oss.nxp.…) - Addressed review from Cezary and Pierre - fixed layout of declaration blocks to be more consistent - avoid using rtd and runtime simultaneously inside a function (always used rtd and crtd) - check return code for create_page_table - completely remove sof_compr_stream and use snd_compr_tstmap instead to keep compress stream related info.· - add get_caps and get_codec_caps implementations (in patch 2)· sound/soc/sof/compress.c | 273 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 271 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 01ca85f0b87f..91a9c95929cd 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -10,6 +10,22 @@ #include "sof-audio.h" #include "sof-priv.h" +static void sof_set_transferred_bytes(struct snd_compr_tstamp *tstamp, + u64 host_pos, u64 buffer_size) +{ + u64 prev_pos; + unsigned int copied; + + div64_u64_rem(tstamp->copied_total, buffer_size, &prev_pos); + + if (host_pos < prev_pos) + copied = (buffer_size - prev_pos) + host_pos; + else + copied = host_pos - prev_pos; + + tstamp->copied_total += copied; +} + static void snd_sof_compr_fragment_elapsed_work(struct work_struct *work) { struct snd_sof_pcm_stream *sps = @@ -29,14 +45,16 @@ void snd_sof_compr_init_elapsed_work(struct work_struct *work) */ void snd_sof_compr_fragment_elapsed(struct snd_compr_stream *cstream) { + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *crtd = cstream->runtime; struct snd_soc_component *component; - struct snd_soc_pcm_runtime *rtd; + struct snd_compr_tstamp *tstamp; struct snd_sof_pcm *spcm; if (!cstream) return; - rtd = cstream->private_data; + tstamp = crtd->private_data; component = snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); spcm = snd_sof_find_spcm_dai(component, rtd); @@ -46,6 +64,257 @@ void snd_sof_compr_fragment_elapsed(struct snd_compr_stream *cstream) return; } + sof_set_transferred_bytes(tstamp, spcm->stream[cstream->direction].posn.host_posn, + crtd->buffer_size); + /* use the same workqueue-based solution as for PCM, cf. snd_sof_pcm_elapsed */ schedule_work(&spcm->stream[cstream->direction].period_elapsed_work); } + +static int create_page_table(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + unsigned char *dma_area, size_t size) +{ + struct snd_dma_buffer *dmab = cstream->runtime->dma_buffer_p; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + int dir = cstream->direction; + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + return snd_sof_create_page_table(component->dev, dmab, + spcm->stream[dir].page_table.area, size); +} + +int sof_compr_open(struct snd_soc_component *component, + struct snd_compr_stream *cstream) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *crtd = cstream->runtime; + struct snd_compr_tstamp *tstamp; + struct snd_sof_pcm *spcm; + int dir; + + tstamp = kzalloc(sizeof(*tstamp), GFP_KERNEL); + if (!tstamp) + return -ENOMEM; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) { + kfree(tstamp); + return -EINVAL; + } + + dir = cstream->direction; + + if (spcm->stream[dir].cstream) { + kfree(tstamp); + return -EBUSY; + } + + spcm->stream[dir].cstream = cstream; + spcm->stream[dir].posn.host_posn = 0; + spcm->stream[dir].posn.dai_posn = 0; + spcm->prepared[dir] = false; + + crtd->private_data = tstamp; + + return 0; +} + +int sof_compr_free(struct snd_soc_component *component, + struct snd_compr_stream *cstream) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_compr_tstamp *tstamp = cstream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct sof_ipc_stream stream; + struct sof_ipc_reply reply; + struct snd_sof_pcm *spcm; + int ret = 0; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE; + stream.comp_id = spcm->stream[cstream->direction].comp_id; + + if (spcm->prepared[cstream->direction]) { + ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, + &stream, sizeof(stream), + &reply, sizeof(reply)); + if (!ret) + spcm->prepared[cstream->direction] = false; + } + + cancel_work_sync(&spcm->stream[cstream->direction].period_elapsed_work); + spcm->stream[cstream->direction].cstream = NULL; + kfree(tstamp); + + return ret; +} + +int sof_compr_set_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_compr_params *params) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *crtd = cstream->runtime; + struct sof_ipc_pcm_params_reply ipc_params_reply; + struct snd_compr_tstamp *tstamp; + struct sof_ipc_pcm_params pcm; + struct snd_sof_pcm *spcm; + int ret; + + tstamp = crtd->private_data; + + spcm = snd_sof_find_spcm_dai(component, rtd); + + if (!spcm) + return -EINVAL; + + cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG; + cstream->dma_buffer.dev.dev = sdev->dev; + ret = snd_compr_malloc_pages(cstream, crtd->buffer_size); + if (ret < 0) + return ret; + + ret = create_page_table(component, cstream, crtd->dma_area, crtd->dma_bytes); + if (ret < 0) + return ret; + + memset(&pcm, 0, sizeof(pcm)); + + pcm.params.buffer.pages = PFN_UP(crtd->dma_bytes); + pcm.hdr.size = sizeof(pcm); + pcm.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_PARAMS; + + pcm.comp_id = spcm->stream[cstream->direction].comp_id; + pcm.params.hdr.size = sizeof(pcm.params); + pcm.params.buffer.phy_addr = spcm->stream[cstream->direction].page_table.addr; + pcm.params.buffer.size = crtd->dma_bytes; + pcm.params.direction = cstream->direction; + pcm.params.channels = params->codec.ch_out; + pcm.params.rate = params->codec.sample_rate; + pcm.params.buffer_fmt = SOF_IPC_BUFFER_INTERLEAVED; + pcm.params.frame_fmt = SOF_IPC_FRAME_S32_LE; + pcm.params.sample_container_bytes = + snd_pcm_format_physical_width(SNDRV_PCM_FORMAT_S32) >> 3; + pcm.params.host_period_bytes = params->buffer.fragment_size; + + ret = sof_ipc_tx_message(sdev->ipc, pcm.hdr.cmd, &pcm, sizeof(pcm), + &ipc_params_reply, sizeof(ipc_params_reply)); + if (ret < 0) { + dev_err(component->dev, "error ipc failed\n"); + return ret; + } + + tstamp->byte_offset = sdev->stream_box.offset + ipc_params_reply.posn_offset; + tstamp->sampling_rate = params->codec.sample_rate; + + spcm->prepared[cstream->direction] = true; + + return 0; +} + +int sof_compr_get_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_codec *params) +{ + /* TODO: we don't query the supported codecs for now, if the + * application asks for an unsupported codec the set_params() will fail. + */ + return 0; +} + +int sof_compr_trigger(struct snd_soc_component *component, + struct snd_compr_stream *cstream, int cmd) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct sof_ipc_stream stream; + struct sof_ipc_reply reply; + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG; + stream.comp_id = spcm->stream[cstream->direction].comp_id; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_START; + break; + case SNDRV_PCM_TRIGGER_STOP: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_PAUSE; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_RELEASE; + break; + default: + dev_err(component->dev, "error: unhandled trigger cmd %d\n", cmd); + break; + } + + return sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, + &stream, sizeof(stream), + &reply, sizeof(reply)); +} + +int sof_compr_copy(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + unsigned int offset, n; + void *ptr; + int ret; + + if (count > rtd->buffer_size) + count = rtd->buffer_size; + + div_u64_rem(rtd->total_bytes_available, rtd->buffer_size, &offset); + ptr = rtd->dma_area + offset; + n = rtd->buffer_size - offset; + + if (count < n) { + ret = copy_from_user(ptr, buf, count); + } else { + ret = copy_from_user(ptr, buf, n); + ret += copy_from_user(rtd->dma_area, buf + n, count - n); + } + + return count - ret; +} + +static int sof_compr_pointer(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct snd_compr_tstamp *pstamp = cstream->runtime->private_data; + + tstamp->sampling_rate = pstamp->sampling_rate; + tstamp->copied_total = pstamp->copied_total; + + return 0; +} + +struct snd_compress_ops sof_compressed_ops = { + .open = sof_compr_open, + .free = sof_compr_free, + .set_params = sof_compr_set_params, + .get_params = sof_compr_get_params, + .trigger = sof_compr_trigger, + .pointer = sof_compr_pointer, + .copy = sof_compr_copy, +}; +EXPORT_SYMBOL(sof_compressed_ops); -- 2.27.0

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[PATCH 0/2] ASoC: SOF: dma-trace: Change trace_init() ops parameter list
by Peter Ujfalusi 08 Feb '22

08 Feb '22

Hi, the DMA trace implementation on AMD platform assumes that the stream_tag pointer is pointing the stream_tag member of struct sof_ipc_dma_trace_params_ext, which is true at the moment, but it can not be guarantied and a change in the dtrace core can cause out of bound accesses for AMD. For this reason, change the API to pass the struct itself which will remove the assumption and makes it clear from both sides what is expected to be sent via the parameter list. This opens up a window to clean up the intel and AMD implementation at the same time. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: intel: hda-trace: Pass the dma buffer pointer to hda_dsp_trace_prepare ASoC: SOF: dma-trace: Pass pointer to params_ext struct in trace_init() sound/soc/sof/amd/acp-trace.c | 38 ++++++++------------------------- sound/soc/sof/amd/acp.h | 3 ++- sound/soc/sof/intel/hda-trace.c | 17 ++++++++------- sound/soc/sof/intel/hda.h | 3 ++- sound/soc/sof/ops.h | 4 ++-- sound/soc/sof/sof-priv.h | 2 +- sound/soc/sof/trace.c | 2 +- 7 files changed, 26 insertions(+), 43 deletions(-) -- 2.35.0

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[PATCH v5] ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema
by Jayesh Choudhary 08 Feb '22

08 Feb '22

Convert the bindings for McASP controllers for TI SOCs from txt to YAML schema. Adds additional properties 'clocks', 'clock-names', 'power-domains', '#sound-dai-cells' and 'port' which were missing from the txt file. Removes properties 'sram-size-playback' and 'sram-size-capture' since they are not used. Adds 'dmas' and 'dma-names' in the example which were missing from the txt file. Changes 'interrupts' and 'interrupt-names' from optional to required properties. Changes 'op-modes', 'serial-dir' to optional properties as they are not needed if the McASP is used only as GPIO. Changes 'tdm-slots' to required property only for I2S operation mode. Adds the yaml file in the 'MAINTAINERS' under the heading 'TEXAS INSTRUMENTS ASoC DRIVERS' Signed-off-by: Jayesh Choudhary <j-choudhary(a)ti.com> --- Changelog: v5: - changes the commit message - changes description for 'reg', 'op-mode', 'tx-num-evt', 'rx-num-evt' and 'interrupts' - adds minimum and maximum for 'tdm-slots' - removes default from 'serial-dir' - removes maxItems from 'tdm-slots', 'tx-num-evt', 'rx-num-evt' and 'auxclk-fs-ratio' as they are of type uint32 - adds 'tdm-slots' as conditionally required property for I2S mode of operation v4: - changes the commit message - adds type and description to relevant properties - changes maxItems for 'serial-dir' - removes properties 'sram-size-playback' and 'sram-size-capture' as they are not used - removes 'function-gpios' - removes 'num-serializer' - marks 'tdm-slots', 'op-mode' and 'serial-dir' as optional properties - adds the yaml file in MAINTAINERS v3: - remove maxItems from 'clock-names' v2: - changes the commit message - modifies the properties 'clocks', 'clock-names', 'dma-names', 'dmas', 'interrupts' and 'interrupt-names' according to the arm SOCs - adds 'ports' and 'num-serializer' as node properties .../bindings/sound/davinci-mcasp-audio.txt | 86 -------- .../bindings/sound/davinci-mcasp-audio.yaml | 201 ++++++++++++++++++ MAINTAINERS | 1 + 3 files changed, 202 insertions(+), 86 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt deleted file mode 100644 index bd863bd69501..000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ /dev/null @@ -1,86 +0,0 @@ -Texas Instruments McASP controller - -Required properties: -- compatible : - "ti,dm646x-mcasp-audio" : for DM646x platforms - "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - "ti,dra7-mcasp-audio" : for DRA7xx platforms - "ti,omap4-mcasp-audio" : for OMAP4 - -- reg : Should contain reg specifiers for the entries in the reg-names property. -- reg-names : Should contain: - * "mpu" for the main registers (required). For compatibility with - existing software, it is recommended this is the first entry. - * "dat" for separate data port register access (optional). -- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, - IEC60958-1, and AES-3 formats. -- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted - or received over one serializer. -- serial-dir : A list of serializer configuration. Each entry is a number - indication for serializer pin direction. - (0 - INACTIVE, 1 - TX, 2 - RX) -- dmas: two element list of DMA controller phandles and DMA request line - ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. The dma - identifiers must be "rx" and "tx". - -Optional properties: - -- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0 -- tx-num-evt : FIFO levels. -- rx-num-evt : FIFO levels. -- dismod : Specify the drive on TX pin during inactive slots - 0 : 3-state - 2 : logic low - 3 : logic high - Defaults to 'logic low' when the property is not present -- sram-size-playback : size of sram to be allocated during playback -- sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP -- interrupt-names : Known interrupt names are "tx" and "rx" -- pinctrl-0: Should specify pin control group used for this controller. -- pinctrl-names: Should contain only one value - "default", for more details - please refer to pinctrl-bindings.txt -- fck_parent : Should contain a valid clock name which will be used as parent - for the McASP fck -- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK - and FS rate if applicable: - AUCLK rate = auxclk-fs-ratio * FS rate - -Optional GPIO support: -If any McASP pin need to be used as GPIO then the McASP node must have: -... - gpio-controller - #gpio-cells = <2>; -... - -When requesting a GPIO, the first parameter is the PIN index in McASP_P* -registers. -For example to request the AXR2 pin of mcasp8: -function-gpios = <&mcasp8 2 0>; - -Or to request the ACLKR pin of mcasp8: -function-gpios = <&mcasp8 29 0>; - -For generic gpio information, please refer to bindings/gpio/gpio.txt - -Example: - -mcasp0: mcasp0@1d00000 { - compatible = "ti,da830-mcasp-audio"; - reg = <0x100000 0x3000>; - reg-names "mpu"; - interrupts = <82>, <83>; - interrupt-names = "tx", "rx"; - op-mode = <0>; /* MCASP_IIS_MODE */ - tdm-slots = <2>; - serial-dir = < - 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ - 0 0 0 0 - 0 0 0 1 - 2 0 0 0 >; - tx-num-evt = <1>; - rx-num-evt = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml new file mode 100644 index 000000000000..f46c66bc6b2d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,201 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McASP Controller for TI SoCs + +maintainers: + - Jayesh Choudhary <j-choudhary(a)ti.com> + +properties: + compatible: + enum: + - ti,dm646x-mcasp-audio + - ti,da830-mcasp-audio + - ti,am33xx-mcasp-audio + - ti,dra7-mcasp-audio + - ti,omap4-mcasp-audio + + reg: + minItems: 1 + items: + - description: CFG registers + - description: data registers + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + op-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + description: 0 - I2S or 1 - DIT operation mode + enum: + - 0 + - 1 + + tdm-slots: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number of channels over one serializer + the property is ignored in DIT mode + minimum: 2 + maximum: 32 + + serial-dir: + description: + A list of serializer configuration + Entry is indication for serializer pin direction + 0 - Inactive, 1 - TX, 2 - RX + All AXR pins should be present in the array even if inactive + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 25 + items: + minimum: 0 + maximum: 2 + + dmas: + minItems: 1 + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + minItems: 1 + items: + - const: tx + - const: rx + + ti,hwmods: + $ref: /schemas/types.yaml#/definitions/string + description: Name of hwmod associated with McASP + maxItems: 1 + deprecated: true + + tx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + configures WFIFO threshold + 0 disables the FIFO use + if property is missing, then also FIFO use is disabled + + rx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + configures RFIFO threshold + 0 disables the FIFO use + if property is missing, then also FIFO use is disabled + + dismod: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + specify the drive on TX pin during inactive time slots + 0 - 3-state, 2 - logic low, 3 - logic high + enum: + - 0 + - 2 + - 3 + default: 2 + + interrupts: + anyOf: + - minItems: 1 + items: + - description: TX interrupt + - description: RX interrupt + - items: + - description: common/combined interrupt + + interrupt-names: + oneOf: + - minItems: 1 + items: + - const: tx + - const: rx + - const: common + + fck_parent: + $ref: /schemas/types.yaml#/definitions/string + description: parent clock name for McASP fck + maxItems: 1 + + auxclk-fs-ratio: + $ref: /schemas/types.yaml#/definitions/uint32 + description: ratio of AUCLK and FS rate if applicable + + gpio-controller: true + + "#gpio-cells": + const: 2 + + clocks: + minItems: 1 + items: + - description: functional clock + - description: module specific optional ahclkx clock + - description: module specific optional ahclkr clock + + clock-names: + minItems: 1 + items: + - const: fck + - const: ahclkx + - const: ahclkr + + power-domains: + description: phandle to the corresponding power-domain + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + port: + description: connection for when McASP is used via graph card + type: object + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - interrupts + - interrupt-names + +allOf: + - if: + properties: + opmode: + enum: + - 0 + + then: + required: + - tdm-slots + +additionalProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index 5ea5655a29c3..ec6bde4efcae 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -18995,6 +18995,7 @@ TEXAS INSTRUMENTS ASoC DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained +F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ TEXAS INSTRUMENTS' DAC7612 DAC DRIVER -- 2.17.1

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[PATCH 1/3] ASoC: rt5682s: do not block workqueue if card is unbound
by Kai Vehmanen 08 Feb '22

08 Feb '22

The current rt5682s_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index efa1016831dd..1e662d1be2b3 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -824,11 +824,13 @@ static void rt5682s_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5682s_priv, jack_detect_work.work); int val, btn_type; - while (!rt5682s->component) - usleep_range(10000, 15000); - - while (!rt5682s->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682s->component || !rt5682s->component->card || + !rt5682s->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682s->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5682s->jdet_mutex); mutex_lock(&rt5682s->calibrate_mutex); base-commit: d466706b9d0dae4754a47c34fd5f4654ee54f609 -- 2.35.1

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[PATCH v2] ASoC: tas2770: Insert post reset delay
by Martin Povišer 08 Feb '22

08 Feb '22

Per TAS2770 datasheet there must be a 1 ms delay from reset to first command. So insert delays into the driver where appropriate. Fixes: 1a476abc723e ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- In previous submission of this patch some people got their copy encrypted. That was by accident, pardon that. I have changed my email provider since. v1: https://lore.kernel.org/alsa-devel/20220201234612.74401-1-povik+lin@protonm… sound/soc/codecs/tas2770.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index b52c0861edc8..b397a2122dd4 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -38,10 +38,12 @@ static void tas2770_reset(struct tas2770_priv *tas2770) gpiod_set_value_cansleep(tas2770->reset_gpio, 0); msleep(20); gpiod_set_value_cansleep(tas2770->reset_gpio, 1); + usleep_range(1000, 2000); } snd_soc_component_write(tas2770->component, TAS2770_SW_RST, TAS2770_RST); + usleep_range(1000, 2000); } static int tas2770_set_bias_level(struct snd_soc_component *component, @@ -110,6 +112,7 @@ static int tas2770_codec_resume(struct snd_soc_component *component) if (tas2770->sdz_gpio) { gpiod_set_value_cansleep(tas2770->sdz_gpio, 1); + usleep_range(1000, 2000); } else { ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, @@ -529,8 +532,10 @@ static int tas2770_codec_probe(struct snd_soc_component *component) tas2770->component = component; - if (tas2770->sdz_gpio) + if (tas2770->sdz_gpio) { gpiod_set_value_cansleep(tas2770->sdz_gpio, 1); + usleep_range(1000, 2000); + } tas2770_reset(tas2770); -- 2.33.0

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