June 2023 - Alsa-devel - mailman.alsa-project.org

use-case: Add new LatencyOffsetMs field
by GitHub pull_request - opened 21 Jun '23

21 Jun '23

alsa-project/alsa-lib pull request #332 was opened from afq984: Some drivers do not report accurate latencies. Add a LatencyOffsetMs UCM Value allow adding a static offset to the reported latency. Request URL : https://github.com/alsa-project/alsa-lib/pull/332 Patch URL : https://github.com/alsa-project/alsa-lib/pull/332.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH V4 0/9] ASoC: amd: ps: add SoundWire support
by Vijendar Mukunda 21 Jun '23

21 Jun '23

This patch series add support for - Platform device creation for SoundWire Manager instances and PDM controller. - SoundWire DMA driver. - Interrupt handling for SoundWire manager related interrupts, SoundWire DMA interrupts and ACP error interrupts. - ACP PCI driver PM ops modification with respect to SoundWire Power modes. Changes since v3: - use pdev_config instead of pdev_mask in the code. - define platform device configuration macros rather than enums - add comments for MIPI DisCo version - refactor SoundWire DMA start/stop sequence using single function - refactor "acp_reset" flag related functionality. Changes since v2: - add comments in irq handler. - remove workqueue for SoundWire DMA interrupts and use thread implementation for DMA interrupt handling. - add error checks in sdw_amd_scan_controller() - remove passing "acp_lock" as platform resource for SoundWire DMA driver and PDM driver. - retrieve "acp_lock" reference from parent driver directly and use the reference in SoundWire DMA driver. - add handling for acp pci driver probe even when no ACP PDM or SoundWire manager platform devices created. - Fix indentation for acp_sdw_dma_count structure variables. - Use macro instead of hard coded values for FIFO offset and PTE offset. - Change pm_runtime enable sequence in SoundWire DMA driver probe function. - Refactor system level resume callback in SoundWire DMA Changes since v1: - update "soundwire" as "SoundWire" in code. - add comments for platform device mask and platform device count - remove unused variables in acp pci driver private data structure - refactor dma enable register structures in SoundWire DMA driver - add TODO comments in IRQ handler - update IRQ mask values using bit macros - Fix build error reported in Makefile - rename "sdw_dma_stream_instance" structure name as "acp_sdw_dma_stream" Vijendar Mukunda (9): ASoC: amd: ps: create platform devices based on acp config ASoC: amd: ps: handle SoundWire interrupts in acp pci driver ASoC: amd: ps: add SoundWire dma driver ASoC: amd: ps: add SoundWire dma driver dma ops ASoC: amd: ps: add support for SoundWire DMA interrupts ASoC: amd: ps: add pm ops support for SoundWire dma driver ASoC: amd: ps: enable SoundWire dma driver build ASoC: amd: update comments in Kconfig file ASoC: amd: ps: add acp_reset flag check in acp pci driver pm ops. sound/soc/amd/Kconfig | 3 +- sound/soc/amd/ps/Makefile | 2 + sound/soc/amd/ps/acp63.h | 172 ++++++++++- sound/soc/amd/ps/pci-ps.c | 419 +++++++++++++++++++++++-- sound/soc/amd/ps/ps-sdw-dma.c | 555 ++++++++++++++++++++++++++++++++++ 5 files changed, 1115 insertions(+), 36 deletions(-) create mode 100644 sound/soc/amd/ps/ps-sdw-dma.c -- 2.34.1

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[PATCH v3 0/9] add snd_soc_{of_}get_dlc()
by Kuninori Morimoto 21 Jun '23

21 Jun '23

Hi Mark Current soc-core.c has snd_soc_{of_}get_dai_name() to get DAI name for dlc (snd_soc_dai_link_component). It gets .dai_name, but we need .of_node too. Therefor user need to arrange. It will be more useful if it gets both .dai_name and .of_node. This patch adds snd_soc_{of_}get_dlc() for it, and convert to use it. v2 -> v3 - care loongson on [2/9] v1 -> v2 - add patch for loongson Link: https://lore.kernel.org/r/87cz1rdmp0.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87mt12hp75.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (9): ASoC: soc-core.c: add snd_soc_{of_}get_dlc() ASoC: soc-core.c: add index on snd_soc_of_get_dai_name() ASoC: fsl: use snd_soc_{of_}get_dlc() ASoC: qcom: use snd_soc_{of_}get_dlc() ASoC: meson: use snd_soc_{of_}get_dlc() ASoC: samsung: use snd_soc_{of_}get_dlc() ASoC: loongson: use snd_soc_{of_}get_dlc() ASoC: soc-core.c: use snd_soc_{of_}get_dlc() ASoC: simple-card.c: use snd_soc_{of_}get_dlc() include/sound/soc.h | 8 ++- sound/soc/fsl/imx-card.c | 14 +---- sound/soc/fsl/imx-rpmsg.c | 3 +- sound/soc/generic/simple-card-utils.c | 4 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/loongson/loongson_card.c | 34 ++++------ sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/gx-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 ++--- sound/soc/meson/meson-card.h | 3 +- sound/soc/qcom/common.c | 13 +--- sound/soc/samsung/odroid.c | 16 +---- sound/soc/soc-core.c | 77 ++++++++++++----------- 14 files changed, 79 insertions(+), 121 deletions(-) -- 2.25.1

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[PATCH v2 0/9] add snd_soc_{of_}get_dlc()
by Kuninori Morimoto 21 Jun '23

21 Jun '23

Hi Mark Current soc-core.c has snd_soc_{of_}get_dai_name() to get DAI name for dlc (snd_soc_dai_link_component). It gets .dai_name, but we need .of_node too. Therefor user need to arrange. It will be more useful if it gets both .dai_name and .of_node. This patch adds snd_soc_{of_}get_dlc() for it, and convert to use it. v1 -> v2 - care loongson Link: https://lore.kernel.org/r/87mt12hp75.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (9): ASoC: soc-core.c: add snd_soc_{of_}get_dlc() ASoC: soc-core.c: add index on snd_soc_of_get_dai_name() ASoC: fsl: use snd_soc_{of_}get_dlc() ASoC: qcom: use snd_soc_{of_}get_dlc() ASoC: meson: use snd_soc_{of_}get_dlc() ASoC: samsung: use snd_soc_{of_}get_dlc() ASoC: loongson: use snd_soc_{of_}get_dlc() ASoC: soc-core.c: use snd_soc_{of_}get_dlc() ASoC: simple-card.c: use snd_soc_{of_}get_dlc() include/sound/soc.h | 8 ++- sound/soc/fsl/imx-card.c | 14 +---- sound/soc/fsl/imx-rpmsg.c | 3 +- sound/soc/generic/simple-card-utils.c | 4 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/loongson/loongson_card.c | 34 ++++------ sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/gx-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 ++--- sound/soc/meson/meson-card.h | 3 +- sound/soc/qcom/common.c | 13 +--- sound/soc/samsung/odroid.c | 16 +---- sound/soc/soc-core.c | 77 ++++++++++++----------- 14 files changed, 79 insertions(+), 121 deletions(-) -- 2.25.1

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[PATCH] ASoC: intel: sof_sdw: Fixup typo in device link checking
by Charles Keepax 21 Jun '23

21 Jun '23

The loop checking for multiple different devices on a single sdw link contains a typo accidentally using i twice instead of j. Correct to the correct index variable. Fixes: dc5a3e60a4b5 ("ASoC: Intel: sof_sdw: append codec type to dai link name") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/intel/boards/sof_sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6faf4a43eaf5a..144f082c63fda 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1347,7 +1347,7 @@ static int sof_card_dai_links_create(struct device *dev, if ((SDW_PART_ID(adr_link->adr_d[i].adr) != SDW_PART_ID(adr_link->adr_d[j].adr)) || (SDW_MFG_ID(adr_link->adr_d[i].adr) != - SDW_MFG_ID(adr_link->adr_d[i].adr))) { + SDW_MFG_ID(adr_link->adr_d[j].adr))) { append_codec_type = true; goto out; } -- 2.30.2

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[PATCH 0/8] add snd_soc_{of_}get_dlc()
by Kuninori Morimoto 21 Jun '23

21 Jun '23

Hi Mark Current soc-core.c has snd_soc_{of_}get_dai_name() to get DAI name for dlc (snd_soc_dai_link_component). It gets .dai_name, but we need .of_node too. Therefor user need to arrange. It will be more useful if it gets both .dai_name and .of_node. This patch adds snd_soc_{of_}get_dlc() for it, and convert to use it. Kuninori Morimoto (8): ASoC: soc-core.c: add snd_soc_{of_}get_dlc() ASoC: soc-core.c: add index on snd_soc_of_get_dai_name() ASoC: fsl: use snd_soc_{of_}get_dlc() ASoC: qcom: use snd_soc_{of_}get_dlc() ASoC: meson: use snd_soc_{of_}get_dlc() ASoC: samsung: use snd_soc_{of_}get_dlc() ASoC: soc-core.c: use snd_soc_{of_}get_dlc() ASoC: simple-card.c: use snd_soc_{of_}get_dlc() include/sound/soc.h | 8 ++- sound/soc/fsl/imx-card.c | 14 +---- sound/soc/fsl/imx-rpmsg.c | 3 +- sound/soc/generic/simple-card-utils.c | 4 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/gx-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 16 ++--- sound/soc/meson/meson-card.h | 3 +- sound/soc/qcom/common.c | 13 +--- sound/soc/samsung/odroid.c | 16 +---- sound/soc/soc-core.c | 77 ++++++++++++----------- 13 files changed, 68 insertions(+), 98 deletions(-) -- 2.25.1

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[PATCH][next] ASoC: tas2781: Fix spelling mistake "calibraiton" -> "calibration"
by Colin Ian King 20 Jun '23

20 Jun '23

There is a spelling mistake in a dev_err message. Fix it. Also fix grammar and add space between last word and (%d)". Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/tas2781-fmwlib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 432b19ccec8c..cbf0aef2c001 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1863,7 +1863,7 @@ static int fw_parse_calibration_data(struct tasdevice_priv *tas_priv, if (tas_fmw->nr_calibrations != 1) { dev_err(tas_priv->dev, - "%s: only support one calibraiton(%d)!\n", + "%s: only supports one calibration (%d)!\n", __func__, tas_fmw->nr_calibrations); goto out; } -- 2.39.2

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[PATCH 1/1] ASoC: dwc: add DMA handshake control
by Maxim Kochetkov 20 Jun '23

20 Jun '23

DMA mode uses hardware handshake signals. DMACR register is used to enable the DMA Controller interface operation. So add DMA enable/disable to i2s_start()/i2s_stop() functions if using DMA mode. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/dwc/dwc-i2s.c | 39 +++++++++++++++++++++++++++++++++++++-- sound/soc/dwc/local.h | 6 ++++++ 2 files changed, 43 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 9e7065dd854c..02b9894e99a7 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -150,19 +150,51 @@ static irqreturn_t i2s_irq_handler(int irq, void *dev_id) return IRQ_NONE; } +static void i2s_enable_dma(struct dw_i2s_dev *dev, u32 stream) +{ + u32 dma_reg = i2s_read_reg(dev->i2s_base, I2S_DMACR); + + /* Enable DMA handshake for stream */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_reg |= I2S_DMAEN_TXBLOCK; + else + dma_reg |= I2S_DMAEN_RXBLOCK; + + i2s_write_reg(dev->i2s_base, I2S_DMACR, dma_reg); +} + +static void i2s_disable_dma(struct dw_i2s_dev *dev, u32 stream) +{ + u32 dma_reg = i2s_read_reg(dev->i2s_base, I2S_DMACR); + + /* Disable DMA handshake for stream */ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_reg &= ~I2S_DMAEN_TXBLOCK; + i2s_write_reg(dev->i2s_base, I2S_RTXDMA, 1); + } else { + dma_reg &= ~I2S_DMAEN_RXBLOCK; + i2s_write_reg(dev->i2s_base, I2S_RRXDMA, 1); + } + i2s_write_reg(dev->i2s_base, I2S_DMACR, dma_reg); +} + static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { struct i2s_clk_config_data *config = &dev->config; i2s_write_reg(dev->i2s_base, IER, 1); - i2s_enable_irqs(dev, substream->stream, config->chan_nr); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) i2s_write_reg(dev->i2s_base, ITER, 1); else i2s_write_reg(dev->i2s_base, IRER, 1); + if (dev->use_pio) + i2s_enable_irqs(dev, substream->stream, config->chan_nr); + else + i2s_enable_dma(dev, substream->stream); + i2s_write_reg(dev->i2s_base, CER, 1); } @@ -176,7 +208,10 @@ static void i2s_stop(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 0); - i2s_disable_irqs(dev, substream->stream, 8); + if (dev->use_pio) + i2s_disable_irqs(dev, substream->stream, 8); + else + i2s_disable_dma(dev, substream->stream); if (!dev->active) { i2s_write_reg(dev->i2s_base, CER, 0); diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index d64bd4f8fd34..ba4e397099be 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -53,6 +53,12 @@ #define I2S_COMP_VERSION 0x01F8 #define I2S_COMP_TYPE 0x01FC +#define I2S_RRXDMA 0x01C4 +#define I2S_RTXDMA 0x01CC +#define I2S_DMACR 0x0200 +#define I2S_DMAEN_RXBLOCK (1 << 16) +#define I2S_DMAEN_TXBLOCK (1 << 17) + /* * Component parameter register fields - define the I2S block's * configuration. -- 2.40.1

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[PATCH 1/1] ASoC: dwc: don't assign addr_width for dt configs
by Maxim Kochetkov 20 Jun '23

20 Jun '23

For proper DMA operation addr_width must corresponds with audio format (S16, S24, S32, etc). Proper bus width calculations is performed by snd_hwparams_to_dma_slave_config(). So drop wrong addr_width asignment for dt configs and let snd_hwparams_to_dma_slave_config() do the job. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/dwc/dwc-i2s.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 9e7065dd854c..21398ee924e4 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -558,13 +558,9 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, u32 comp1 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_1); u32 comp2 = i2s_read_reg(dev->i2s_base, I2S_COMP_PARAM_2); u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1)); - u32 idx = COMP1_APB_DATA_WIDTH(comp1); u32 idx2; int ret; - if (WARN_ON(idx >= ARRAY_SIZE(bus_widths))) - return -EINVAL; - ret = dw_configure_dai(dev, dw_i2s_dai, SNDRV_PCM_RATE_8000_192000); if (ret < 0) return ret; @@ -574,7 +570,6 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->capability |= DWC_I2S_PLAY; dev->play_dma_data.dt.addr = res->start + I2S_TXDMA; - dev->play_dma_data.dt.addr_width = bus_widths[idx]; dev->play_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2]) >> 8; dev->play_dma_data.dt.maxburst = 16; @@ -584,7 +579,6 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, dev->capability |= DWC_I2S_RECORD; dev->capture_dma_data.dt.addr = res->start + I2S_RXDMA; - dev->capture_dma_data.dt.addr_width = bus_widths[idx]; dev->capture_dma_data.dt.fifo_size = fifo_depth * (fifo_width[idx2] >> 8); dev->capture_dma_data.dt.maxburst = 16; -- 2.40.1

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[PATCH v3 00/11] ASoC: qcom: audioreach: add compress offload support
by Srinivas Kandagatla 20 Jun '23

20 Jun '23

This patchset adds compressed offload support to Qualcomm audioreach drivers. Currently it supports AAC, MP3 and FALC along with gapless. Tested this on SM8450 and sc7280. thanks, srini Changes since v2: - fixed order of sign-off - added placeholder decoder and encoder to audioreach_set_media_format Mohammad Rafi Shaik (4): ASoC: qcom: SC7280: audioreach: Add sc7280 hardware param fixup callback ASoC: q6dsp: q6apm: add end of stream events ASoC: q6dsp: audioreach: Add support to set compress format params ASoC: q6dsp: audioreach: Add gapless feature support Srinivas Kandagatla (7): ASoC: q6dsp: audioreach: add helper function to set u32 param ASoC: q6dsp: audioreach: Add placeholder decoder for compress playback ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress DAI and codec caps get callbacks ASoC: q6dsp: q6apm-dai: Add trigger/pointer compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress set params and metadata DAI callbacks ASoC: q6dsp: q6apm-dai: Add mmap and copy compress DAI callbacks sound/soc/qcom/qdsp6/audioreach.c | 250 ++++++++++------- sound/soc/qcom/qdsp6/audioreach.h | 51 ++++ sound/soc/qcom/qdsp6/q6apm-dai.c | 445 ++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6apm.c | 68 +++++ sound/soc/qcom/qdsp6/q6apm.h | 6 + sound/soc/qcom/sc7280.c | 23 +- 6 files changed, 747 insertions(+), 96 deletions(-) -- 2.21.0

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