Alsa-devel January 2022

alsa-devel@alsa-project.org

134 participants
273 discussions

[PATCH] soundwire: Replace acpi_bus_get_device()
by Rafael J. Wysocki 10 Feb '22

10 Feb '22

From: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Replace acpi_bus_get_device() that is going to be dropped with acpi_fetch_acpi_dev(). No intentional functional impact. Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> --- drivers/soundwire/intel_init.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) Index: linux-pm/drivers/soundwire/intel_init.c =================================================================== --- linux-pm.orig/drivers/soundwire/intel_init.c +++ linux-pm/drivers/soundwire/intel_init.c @@ -180,7 +180,8 @@ static struct sdw_intel_ctx if (!res) return NULL; - if (acpi_bus_get_device(res->handle, &adev)) + adev = acpi_fetch_acpi_dev(res->handle); + if (!adev) return NULL; if (!res->count) @@ -294,13 +295,13 @@ err: static int sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) { - struct acpi_device *adev; + struct acpi_device *adev = acpi_fetch_acpi_dev(ctx->handle); struct sdw_intel_link_dev *ldev; u32 caps; u32 link_mask; int i; - if (acpi_bus_get_device(ctx->handle, &adev)) + if (!adev) return -EINVAL; /* Check SNDWLCAP.LCOUNT */

2 1

[PATCH] soundwire: bus: add dev_warn() messages to track UNATTACHED devices
by Bard Liao 10 Feb '22

10 Feb '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> In rare cases, some devices seem to lose sync and never re-attach on the bus. This seems to happen only when there are more than one device per link, which suggests either an electrical issue, a race condition or a state machine issue. Add two dev_warn() messages to identify the sequence by which the devices become UNATTACHED. BugLink: https://github.com/thesofproject/linux/issues/3063 BugLink: https://github.com/thesofproject/linux/issues/3325 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 67369e941d0d..354d3f89366f 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1749,8 +1749,11 @@ int sdw_handle_slave_status(struct sdw_bus *bus, continue; if (status[i] == SDW_SLAVE_UNATTACHED && - slave->status != SDW_SLAVE_UNATTACHED) + slave->status != SDW_SLAVE_UNATTACHED) { + dev_warn(&slave->dev, "Slave %d state check1: UNATTACHED, status was %d\n", + i, slave->status); sdw_modify_slave_status(slave, SDW_SLAVE_UNATTACHED); + } } if (status[0] == SDW_SLAVE_ATTACHED) { @@ -1785,6 +1788,9 @@ int sdw_handle_slave_status(struct sdw_bus *bus, if (slave->status == SDW_SLAVE_UNATTACHED) break; + dev_warn(&slave->dev, "Slave %d state check2: UNATTACHED, status was %d\n", + i, slave->status); + sdw_modify_slave_status(slave, SDW_SLAVE_UNATTACHED); break; -- 2.17.1

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[PATCH] soundwire: intel: fix wrong register name in intel_shim_wake
by Bard Liao 10 Feb '22

10 Feb '22

From: Libin Yang <libin.yang(a)intel.com> When clearing the sdw wakests status, we should use SDW_SHIM_WAKESTS. Fixes: 4a17c441c7cb ("soundwire: intel: revisit SHIM programming sequences.") Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 122f7a29d8ca..63101f1ba271 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -448,8 +448,8 @@ static void intel_shim_wake(struct sdw_intel *sdw, bool wake_enable) /* Clear wake status */ wake_sts = intel_readw(shim, SDW_SHIM_WAKESTS); - wake_sts |= (SDW_SHIM_WAKEEN_ENABLE << link_id); - intel_writew(shim, SDW_SHIM_WAKESTS_STATUS, wake_sts); + wake_sts |= (SDW_SHIM_WAKESTS_STATUS << link_id); + intel_writew(shim, SDW_SHIM_WAKESTS, wake_sts); } mutex_unlock(sdw->link_res->shim_lock); } -- 2.17.1

2 1

[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 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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[RFC 00/37] ASoC: Intel: AVS - Audio DSP for cAVS
by Cezary Rojewski 07 Feb '22

07 Feb '22

A continuation of cleanup work of Intel SST solutions found in sound/soc/intel/. With two major chapters released last year catpt [1] and removal of haswell solution [2], time has come for Skylake-driver. Througout 2019, 2020 and 2021 Skylake-driver has had many fixes applied and even attempts of refactors as seen in fundamental overhaul [3], IPC flow adjustments [4] and LARGE_CONFIG overhaul [5] series. Unfortunately, story repeats itself - problems are found within the core of a driver. Painting it with different colors does not change the fact that is it still a house of cards. As changes needed to address those issues would make Skylake solution incompatible with its previous revisions, a decision has been made to provide a new solution instead. In time it would deprecate and replace Skylake-driver. That solution has been called AVS - from AudioDSP architecture name: Audio-Voice-Speech. It is meant to provide support for the exact same range of platforms as its predecessor: SKL, KBL, AML and APL. Several functions found within HDAudio and ASoC framework have been exported and are re-used later by the avs-driver to prevent code being duplicated in this solution. All of these act as driver dependencies and are found at the beginning of the series to make it clear what's needed for avs-driver to compile. Note: this series is dependent upon NHLT-series [6] which was recently merged by Takashi to his for-next branch yet is unavailable on current broonie/for-next. Note: this series does not add fully functional driver as its size would get out of control. Focus is put on adding new code. A SND_SOC_INTEL_SKYLAKE_FAMILY=n dependency is added so no probing attempts are performed. Until all patches are merged, Skylake-driver remains the recommended option. Series which will later follow this one, focus on filling all the remaining functionality gaps and provide filesystems support. Also, a range of machine boards will be added to match range of configurations supported by the avs-driver. Once avs-driver and its collaterals are merged, skylake-driver deprecation and removal process begins, along with all other components that are connected to it e.g.: skylake-driver-only machine boards. As AudioDSP firmware which avs-driver communicates with supports a wide range of audio formats, module configurations and multi-pipeline streams, couple of new concepts are introduce to enable all those functionalities: - Path template and path variants - Runtime path - Conditional path - Granular sound cards (as opposed to 'super-cards') These are later better explained by their respective patches: 'Topology parsing', 'Path management', 'Conditional-path support' and 'Machine board registration'. A 'path' represents a DSP side of audio stream in runtime - is a logical container for pipelines which are themselves composed of modules - processing units. Due to high range of possible audio format combinations, there can be more variants of given path (and thus, its pipelines and modules) than total number of pipelines and module instances which firmware supports concurrently, all the instance IDs are allocated dynamically with help of IDA interface. 'Path templates' come from topology file and describe a pattern which is later used to actually create runtime 'path'. To support modern audio usecases such as WaveOnVoice and EchoReference, conditional paths concept came into existence. These work similarly to standard paths except are a consequence of other paths being created or deleted, rather then being created when userspace app opens specific FE for streaming. Their state machine is controlled by source and sink paths which created them in the first place. Granular machine boards is a contrast to 'super-card' idea which is currently widely used throughout Intel ASoC drivers. Major reasons are: complexity reduction (each board now focuses on a single, concrete device) and overall reduction of topology file size when entire configuration is taken into account. This has functional benefits too: one card failing won't prevent others from probing and being operative. Changes [internal] RFC v2 -> [public] RFC v1: - dropped any sysfs related changes from this series, moved to follow up one - dropped entire subscription-mechanism found in ipc.c. Handlers that are delegated to service certain firmware notifications are now called directly - fixed kernel doc for snd_soc_dapm_new_dai_widgets() as reported by ikp - prefixed snd_hda_codec_device_init() as suggested by Amadeo - improved comments for d0ix transitions for APL-based platforms as suggested by Pierre - a ton of spelling related fixes in most of the commit messages - fixed remaining warnings pointed by scan-build (variable assigned but not used) - replaced most of 'cAVS X.Y' expression usages with 'platform-based' equivalents as suggested by Pierre e.g.: cAVS 1.5 -> SKL-based Changes [internal] RFC v1 -> [internal] RFC v2: - fixed memleak caused by lack of kfree(vols) if memory allocation fails in avs_peakvol_create() as reported by Curtis - fixed missing 'i' iterator incrementation in avs_widget_ready() causing reference loss as reported by Curtis - replace hardcode: 0x40 usage with snd_hdac_calc_stream_format as suggested by Curtis. In consequence, readability for all code loading (CL) procedues has increased and such approach auto-documents the CL stream preparation - updated behavior of all index-fetching functions found in utils.c: avs_module_entry_index(), avs_module_id_entry_index() and follow ups: avs_get_module_entry(), avs_get_module_id_entry() to better conform to linux-kernel standard when no entry is found (return -ENOENT) rather than C++ standard (return -1, what in kernel case translated to -EPERM) as suggested by Curtis and Peter - several suggestions have been made regarding spacing, and so far, I've agreed and applied with all of them. None proposed seemed out of place or redundant - avs_path_stop() renamed to avs_path_pause() pipeline states are represented by RESET/PAUSED/RUNNING. avs_path_reset() and avs_path_run() were already there and avs_path_stop() just didn't look cohesive - added missing parsers for num_output_pin and num_input_pin which are required for custom modules such as WAVES or DSM - dropped 'priv_param_length' from custom module descriptor as this field is obsolete in firmware - parse_dictionary() has been split into parse_dictionary_header() and parse_dictionary_entries() to drop code duplication present in several parsing function which could not re-use entire parse_dictionary() - added avs_tplg_vendor_array_lookup_next() and avs_tplg_vendor_entry_next() to drop code duplicated present in several parsing functions. This change greatly impacted readability of all parsers - parsing helpers such avs_tplg_vendor_array_lookup() now return offset by updating specified in function argument list u32 *offset variable. This is to address problem when u32 offset would be greater than max int, which is the return type for these functions - AVS_DEFINE_PTR_PARSER() macro has been introduced to drop code duplication for all ptr-parsing users - all struct avs_path_module creators have had their declaration updated: function argument *owner ceased to exist as it could already be accessed by mod->owner - fixed the order of operation for conditional paths (e.g.: echo reference) so these are no longer controlled by "source" path and instead are impacted by state changes of source and sink paths both. Previously only source path e.g. playback sourcing echo reference would trigger RUNNING status for conditional path. Equivalent RUNNING on WoV path which is in this case sink path, would not do so, leading to order-of-operation problems. Behavior has been changed to: both source and sink need to be RUNNING for conditional path to be set to RUNNING too. PAUSED for either source or sink will cause PAUSED transition for conditional path. - to achieve the above, path states are now saved in 'state' i.e. new u32 field for struct avs_path - resigned from fw_filename field usage in favour of newly added tplg_filename for machine board descriptors as suggested by Pierre - platform descriptor fields have had their names update better reflect their purpose as suggested by Pierre - fixed comp_list missing locking when manipulated - all message senders now accept request as pointer as suggeseted by Peter - resigned of AZX_ usage for all ADSP-related registers, leaving them only for HOST memory space related operations - fixed disable path for core DSP operations: power/reset/stall as reported by Peter - safety when locking between received responses (reply vs notification) has been lowered as suggested by Pierre. Most usages are not performed in IRQ context and none is done in hard-IRQ one - s/master/main/ plus AVS_MAIN_CORE_MASK has replaced ->master_mask - several functions have had their logging updated - logs have been moved to lower level functions as suggested by Pierre - hdac_ext_stream usage has been streamlined to estream, hdac_streams are represented by hstream instead - hw_params() are resilient to scenarios when they are called mutliple times as reported by Pierre - avs_dsp_enable() now collapses if any of its steps fails as reported by Pierre and Peter - avs_module_ida_empty() now returns value of type bool as suggested by Bard [1]: https://www.spinics.net/lists/alsa-devel/msg116440.html [2]: https://www.spinics.net/lists/alsa-devel/msg116901.html [3]: https://www.spinics.net/lists/alsa-devel/msg94199.html [4]: https://www.spinics.net/lists/alsa-devel/msg92588.html [5]: https://lore.kernel.org/all/20190808181549.12521-1-cezary.rojewski@intel.co… [6]: https://lore.kernel.org/all/YaDq7L1Mu++3UBL7@sirena.org.uk/T/ Cezary Rojewski (37): ALSA: hda: Add snd_hdac_ext_bus_link_at() helper ALSA: hda: Update and expose snd_hda_codec_device_init() ALSA: hda: Update and expose codec register procedures ALSA: hda: Expose codec cleanup and power-save functions ALSA: hda: Add helper macros for DSP capable devices ASoC: Export DAI register and widget ctor and dctor functions ASoC: Intel: Introduce AVS driver ASoC: Intel: avs: Inter process communication ASoC: Intel: avs: Add code loading requests ASoC: Intel: avs: Add pipeline management requests ASoC: Intel: avs: Add module management requests ASoC: Intel: avs: Add power management requests ASoC: Intel: avs: Add ROM requests ASoC: Intel: avs: Add basefw runtime-parameter requests ASoC: Intel: avs: Firmware resources management utilities ASoC: Intel: avs: Declare module configuration types ASoC: Intel: avs: Dynamic firmware resources management ASoC: Intel: avs: Topology parsing ASoC: Intel: avs: Path management ASoC: Intel: avs: Conditional-path support ASoC: Intel: avs: General code loading flow ASoC: Intel: avs: Implement CLDMA transfer ASoC: Intel: avs: Code loading over CLDMA ASoC: Intel: avs: Code loading over HDA ASoC: Intel: avs: Generic soc component driver ASoC: Intel: avs: Generic PCM FE operations ASoC: Intel: avs: non-HDA PCM BE operations ASoC: Intel: avs: HDA PCM BE operations ASoC: Intel: avs: Coredump and recovery flow ASoC: Intel: avs: Prepare for firmware tracing ASoC: Intel: avs: D0ix power state support ASoC: Intel: avs: Event tracing ASoC: Intel: avs: Machine board registration ASoC: Intel: avs: PCI driver implementation ASoC: Intel: avs: Power management ASoC: Intel: avs: SKL-based platforms support ASoC: Intel: avs: APL-based platforms support .../ABI/testing/sysfs-bus-pci-devices-avs | 24 + include/sound/hda_codec.h | 11 +- include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 50 + include/sound/soc-acpi.h | 2 + include/sound/soc-dapm.h | 1 + include/uapi/sound/intel/avs/tokens.h | 147 ++ sound/hda/ext/hdac_ext_controller.c | 31 +- sound/pci/hda/hda_codec.c | 93 +- sound/pci/hda/hda_local.h | 2 - sound/soc/codecs/hdac_hda.c | 2 +- sound/soc/intel/Kconfig | 19 + sound/soc/intel/Makefile | 1 + sound/soc/intel/avs/Makefile | 12 + sound/soc/intel/avs/apl.c | 244 +++ sound/soc/intel/avs/avs.h | 331 ++++ sound/soc/intel/avs/board_selection.c | 459 +++++ sound/soc/intel/avs/cldma.c | 328 ++++ sound/soc/intel/avs/cldma.h | 29 + sound/soc/intel/avs/core.c | 737 +++++++ sound/soc/intel/avs/dsp.c | 326 ++++ sound/soc/intel/avs/ipc.c | 612 ++++++ sound/soc/intel/avs/loader.c | 672 +++++++ sound/soc/intel/avs/messages.c | 674 +++++++ sound/soc/intel/avs/messages.h | 813 ++++++++ sound/soc/intel/avs/path.c | 1287 +++++++++++++ sound/soc/intel/avs/path.h | 85 + sound/soc/intel/avs/pcm.c | 1240 ++++++++++++ sound/soc/intel/avs/registers.h | 83 + sound/soc/intel/avs/skl.c | 127 ++ sound/soc/intel/avs/topology.c | 1700 +++++++++++++++++ sound/soc/intel/avs/topology.h | 207 ++ sound/soc/intel/avs/trace.c | 34 + sound/soc/intel/avs/trace.h | 158 ++ sound/soc/intel/avs/utils.c | 305 +++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 15 + 37 files changed, 10824 insertions(+), 40 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-pci-devices-avs create mode 100644 include/uapi/sound/intel/avs/tokens.h create mode 100644 sound/soc/intel/avs/Makefile create mode 100644 sound/soc/intel/avs/apl.c create mode 100644 sound/soc/intel/avs/avs.h create mode 100644 sound/soc/intel/avs/board_selection.c create mode 100644 sound/soc/intel/avs/cldma.c create mode 100644 sound/soc/intel/avs/cldma.h create mode 100644 sound/soc/intel/avs/core.c create mode 100644 sound/soc/intel/avs/dsp.c create mode 100644 sound/soc/intel/avs/ipc.c create mode 100644 sound/soc/intel/avs/loader.c create mode 100644 sound/soc/intel/avs/messages.c create mode 100644 sound/soc/intel/avs/messages.h create mode 100644 sound/soc/intel/avs/path.c create mode 100644 sound/soc/intel/avs/path.h create mode 100644 sound/soc/intel/avs/pcm.c create mode 100644 sound/soc/intel/avs/registers.h create mode 100644 sound/soc/intel/avs/skl.c create mode 100644 sound/soc/intel/avs/topology.c create mode 100644 sound/soc/intel/avs/topology.h create mode 100644 sound/soc/intel/avs/trace.c create mode 100644 sound/soc/intel/avs/trace.h create mode 100644 sound/soc/intel/avs/utils.c -- 2.25.1

4 56

[PATCH] ASoC: amd: acp: Set gpio_spkr_en to None for max speaker amplifer in machine driver
by V sujith kumar Reddy 03 Feb '22

03 Feb '22

Maxim codec driver already enabling/disabling spk_en_gpio in form of sd_mode gpio hence remove such gpio access control from machine driver to avoid conflict Signed-off-by: V sujith kumar Reddy <vsujithkumar.reddy(a)amd.com> --- sound/soc/amd/acp/acp-mach.h | 1 - sound/soc/amd/acp/acp-sof-mach.c | 4 ++-- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index fd6299844ebe..c855f50d6b34 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -21,7 +21,6 @@ #include <linux/gpio/consumer.h> #define EN_SPKR_GPIO_GB 0x11F -#define EN_SPKR_GPIO_NK 0x146 #define EN_SPKR_GPIO_NONE -EINVAL enum be_id { diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index c1d9650fc222..3346677949e3 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -37,7 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NK, + .gpio_spkr_en = EN_SPKR_GPIO_NONE, }; static struct acp_card_drvdata sof_rt5682s_rt1019_data = { @@ -56,7 +56,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NK, + .gpio_spkr_en = EN_SPKR_GPIO_NONE, }; static const struct snd_kcontrol_new acp_controls[] = { -- 2.25.1

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Alsa-devel January 2022