snd_pcm_format_t is more formal than enum asrc_word_width, which has two property, width and physical width, which is more accurate than enum asrc_word_width. So it is better to use in(out)put_format instead of in(out)put_word_width.

Signed-off-by: Shengjiu Wang shengjiu.wang@nxp.com Acked-by: Nicolin Chen nicoleotsuka@gmail.com --- sound/soc/fsl/fsl_asrc.c | 56 +++++++++++++++++++++++++++------------- sound/soc/fsl/fsl_asrc.h | 4 +-- 2 files changed, 40 insertions(+), 20 deletions(-)

diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index cfa40ef6b1ca..4d3804a1ea55 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -265,6 +265,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; enum asrc_pair_index index = pair->index; + enum asrc_word_width input_word_width; + enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; u32 clk_index[2], div[2]; int in, out, channels; @@ -283,9 +285,32 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; }

- /* Validate output width */ - if (config->output_word_width == ASRC_WIDTH_8_BIT) { - pair_err("does not support 8bit width output\n"); + switch (snd_pcm_format_width(config->input_format)) { + case 8: + input_word_width = ASRC_WIDTH_8_BIT; + break; + case 16: + input_word_width = ASRC_WIDTH_16_BIT; + break; + case 24: + input_word_width = ASRC_WIDTH_24_BIT; + break; + default: + pair_err("does not support this input format, %d\n", + config->input_format); + return -EINVAL; + } + + switch (snd_pcm_format_width(config->output_format)) { + case 16: + output_word_width = ASRC_WIDTH_16_BIT; + break; + case 24: + output_word_width = ASRC_WIDTH_24_BIT; + break; + default: + pair_err("does not support this output format, %d\n", + config->output_format); return -EINVAL; }

@@ -383,8 +408,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* Implement word_width configurations */ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR1(index), ASRMCR1i_OW16_MASK | ASRMCR1i_IWD_MASK, - ASRMCR1i_OW16(config->output_word_width) | - ASRMCR1i_IWD(config->input_word_width)); + ASRMCR1i_OW16(output_word_width) | + ASRMCR1i_IWD(input_word_width));

/* Enable BUFFER STALL */ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index), @@ -497,13 +522,13 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); - int width = params_width(params); struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; unsigned int channels = params_channels(params); unsigned int rate = params_rate(params); struct asrc_config config; - int word_width, ret; + snd_pcm_format_t format; + int ret;

ret = fsl_asrc_request_pair(channels, pair); if (ret) { @@ -513,15 +538,10 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream,

pair->config = &config;

- if (width == 16) - width = ASRC_WIDTH_16_BIT; - else - width = ASRC_WIDTH_24_BIT; - if (asrc_priv->asrc_width == 16) - word_width = ASRC_WIDTH_16_BIT; + format = SNDRV_PCM_FORMAT_S16_LE; else - word_width = ASRC_WIDTH_24_BIT; + format = SNDRV_PCM_FORMAT_S24_LE;

config.pair = pair->index; config.channel_num = channels; @@ -529,13 +549,13 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.outclk = OUTCLK_ASRCK1_CLK;

if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - config.input_word_width = width; - config.output_word_width = word_width; + config.input_format = params_format(params); + config.output_format = format; config.input_sample_rate = rate; config.output_sample_rate = asrc_priv->asrc_rate; } else { - config.input_word_width = word_width; - config.output_word_width = width; + config.input_format = format; + config.output_format = params_format(params); config.input_sample_rate = asrc_priv->asrc_rate; config.output_sample_rate = rate; } diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index c60075112570..38af485bdd22 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -342,8 +342,8 @@ struct asrc_config { unsigned int dma_buffer_size; unsigned int input_sample_rate; unsigned int output_sample_rate; - enum asrc_word_width input_word_width; - enum asrc_word_width output_word_width; + snd_pcm_format_t input_format; + snd_pcm_format_t output_format; enum asrc_inclk inclk; enum asrc_outclk outclk; };

When set the runtime hardware parameters, we may need to query the capability of DMA to complete the parameters.

This patch is to Extract this operation from dmaengine_pcm_set_runtime_hwparams function to a separate function snd_dmaengine_pcm_refine_runtime_hwparams, that other components which need this feature can call this function.

Signed-off-by: Shengjiu Wang shengjiu.wang@nxp.com Reviewed-by: Nicolin Chen nicoleotsuka@gmail.com --- include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 61 ++------------------ 3 files changed, 94 insertions(+), 55 deletions(-)

diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index c679f6116580..b65220685920 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -83,6 +83,11 @@ void snd_dmaengine_pcm_set_config_from_dai_data( const struct snd_dmaengine_dai_dma_data *dma_data, struct dma_slave_config *config);

+int snd_dmaengine_pcm_refine_runtime_hwparams( + struct snd_pcm_substream *substream, + struct snd_dmaengine_dai_dma_data *dma_data, + struct snd_pcm_hardware *hw, + struct dma_chan *chan);

/* * Try to request the DMA channel using compat_request_channel or diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 89a05926ac73..5749a8a49784 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -369,4 +369,87 @@ int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close_release_chan);

+/** + * snd_dmaengine_pcm_refine_runtime_hwparams - Refine runtime hw params + * @substream: PCM substream + * @dma_data: DAI DMA data + * @hw: PCM hw params + * @chan: DMA channel to use for data transfers + * + * Returns 0 on success, a negative error code otherwise. + * + * This function will query DMA capability, then refine the pcm hardware + * parameters. + */ +int snd_dmaengine_pcm_refine_runtime_hwparams( + struct snd_pcm_substream *substream, + struct snd_dmaengine_dai_dma_data *dma_data, + struct snd_pcm_hardware *hw, + struct dma_chan *chan) +{ + struct dma_slave_caps dma_caps; + u32 addr_widths = BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | + BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + snd_pcm_format_t i; + int ret = 0; + + if (!hw || !chan || !dma_data) + return -EINVAL; + + ret = dma_get_slave_caps(chan, &dma_caps); + if (ret == 0) { + if (dma_caps.cmd_pause && dma_caps.cmd_resume) + hw->info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; + if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) + hw->info |= SNDRV_PCM_INFO_BATCH; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + addr_widths = dma_caps.dst_addr_widths; + else + addr_widths = dma_caps.src_addr_widths; + } + + /* + * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep + * hw.formats set to 0, meaning no restrictions are in place. + * In this case it's the responsibility of the DAI driver to + * provide the supported format information. + */ + if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) + /* + * Prepare formats mask for valid/allowed sample types. If the + * dma does not have support for the given physical word size, + * it needs to be masked out so user space can not use the + * format which produces corrupted audio. + * In case the dma driver does not implement the slave_caps the + * default assumption is that it supports 1, 2 and 4 bytes + * widths. + */ + for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) { + int bits = snd_pcm_format_physical_width(i); + + /* + * Enable only samples with DMA supported physical + * widths + */ + switch (bits) { + case 8: + case 16: + case 24: + case 32: + case 64: + if (addr_widths & (1 << (bits / 8))) + hw->formats |= pcm_format_to_bits(i); + break; + default: + /* Unsupported types */ + break; + } + } + + return ret; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_refine_runtime_hwparams); + MODULE_LICENSE("GPL"); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 748f5f641002..b9f147eaf7c4 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -118,12 +118,7 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea struct device *dma_dev = dmaengine_dma_dev(pcm, substream); struct dma_chan *chan = pcm->chan[substream->stream]; struct snd_dmaengine_dai_dma_data *dma_data; - struct dma_slave_caps dma_caps; struct snd_pcm_hardware hw; - u32 addr_widths = BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | - BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | - BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); - snd_pcm_format_t i; int ret;

if (pcm->config && pcm->config->pcm_hardware) @@ -145,56 +140,12 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) hw.info |= SNDRV_PCM_INFO_BATCH;

- ret = dma_get_slave_caps(chan, &dma_caps); - if (ret == 0) { - if (dma_caps.cmd_pause && dma_caps.cmd_resume) - hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; - if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) - hw.info |= SNDRV_PCM_INFO_BATCH; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - addr_widths = dma_caps.dst_addr_widths; - else - addr_widths = dma_caps.src_addr_widths; - } - - /* - * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep - * hw.formats set to 0, meaning no restrictions are in place. - * In this case it's the responsibility of the DAI driver to - * provide the supported format information. - */ - if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) - /* - * Prepare formats mask for valid/allowed sample types. If the - * dma does not have support for the given physical word size, - * it needs to be masked out so user space can not use the - * format which produces corrupted audio. - * In case the dma driver does not implement the slave_caps the - * default assumption is that it supports 1, 2 and 4 bytes - * widths. - */ - for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) { - int bits = snd_pcm_format_physical_width(i); - - /* - * Enable only samples with DMA supported physical - * widths - */ - switch (bits) { - case 8: - case 16: - case 24: - case 32: - case 64: - if (addr_widths & (1 << (bits / 8))) - hw.formats |= pcm_format_to_bits(i); - break; - default: - /* Unsupported types */ - break; - } - } + ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream, + dma_data, + &hw, + chan); + if (ret) + return ret;

return snd_soc_set_runtime_hwparams(substream, &hw); }

On Fri, Sep 27, 2019 at 09:46:12AM +0800, Shengjiu Wang wrote:

There is error "aplay: pcm_write:2023: write error: Input/output error" on i.MX8QM/i.MX8QXP platform for S24_3LE format.

In i.MX8QM/i.MX8QXP, the DMA is EDMA, which don't support 24bit sample, but we didn't add any constraint, that cause issues.

So we need to query the caps of dma, then update the hw parameters according to the caps.

Signed-off-by: Shengjiu Wang shengjiu.wang@nxp.com

Acked-by: Nicolin Chen nicoleotsuka@gmail.com

---

sound/soc/fsl/fsl_asrc.c | 4 +-- sound/soc/fsl/fsl_asrc.h | 3 ++ sound/soc/fsl/fsl_asrc_dma.c | 64 ++++++++++++++++++++++++++++++++---- 3 files changed, 62 insertions(+), 9 deletions(-)

diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 584badf956d2..0bf91a6f54b9 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -115,7 +115,7 @@ static void fsl_asrc_sel_proc(int inrate, int outrate,

• within range [ANCA, ANCA+ANCB-1], depends on the channels of pair A
• while pair A and pair C are comparatively independent.

*/ -static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) +int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) { enum asrc_pair_index index = ASRC_INVALID_PAIR; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -158,7 +158,7 @@ static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair)

• It clears the resource from asrc_priv and releases the occupied channels.

*/ -static void fsl_asrc_release_pair(struct fsl_asrc_pair *pair) +void fsl_asrc_release_pair(struct fsl_asrc_pair *pair) { struct fsl_asrc *asrc_priv = pair->asrc_priv; enum asrc_pair_index index = pair->index; diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index 38af485bdd22..2b57e8c53728 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -462,4 +462,7 @@ struct fsl_asrc { #define DRV_NAME "fsl-asrc-dai" extern struct snd_soc_component_driver fsl_asrc_component; struct dma_chan *fsl_asrc_get_dma_channel(struct fsl_asrc_pair *pair, bool dir); +int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair); +void fsl_asrc_release_pair(struct fsl_asrc_pair *pair);

#endif /* _FSL_ASRC_H */ diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 01052a0808b0..2a60fc6142b1 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -16,13 +16,11 @@

#define FSL_ASRC_DMABUF_SIZE (256 * 1024)

-static const struct snd_pcm_hardware snd_imx_hardware = { +static struct snd_pcm_hardware snd_imx_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP |

• SNDRV_PCM_INFO_MMAP_VALID |
  

• SNDRV_PCM_INFO_PAUSE |
  

• SNDRV_PCM_INFO_RESUME,
  

• SNDRV_PCM_INFO_MMAP_VALID,
  

  .buffer_bytes_max = FSL_ASRC_DMABUF_SIZE, .period_bytes_min = 128, .period_bytes_max = 65535, /* Limited by SDMA engine */

@@ -270,12 +268,25 @@ static int fsl_asrc_dma_hw_free(struct snd_pcm_substream *substream)

static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) {

• bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, DRV_NAME);

• struct snd_dmaengine_dai_dma_data *dma_data; struct device *dev = component->dev; struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); struct fsl_asrc_pair *pair;

• struct dma_chan *tmp_chan = NULL;

• u8 dir = tx ? OUT : IN;

• bool release_pair = true;

• int ret = 0;

• ret = snd_pcm_hw_constraint_integer(substream->runtime,

• 			    SNDRV_PCM_HW_PARAM_PERIODS);
  

• if (ret < 0) {

• dev_err(dev, "failed to set pcm hw params periods\n");
  

• return ret;
  

• }

  pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL); if (!pair)

@@ -285,11 +296,50 @@ static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream)

runtime->private_data = pair;

• snd_pcm_hw_constraint_integer(substream->runtime,

• 		      SNDRV_PCM_HW_PARAM_PERIODS);
  

• /* Request a dummy pair, which will be released later.

• * Request pair function needs channel num as input, for this
  

• * dummy pair, we just request "1" channel temporarily.
  

• */
  

• ret = fsl_asrc_request_pair(1, pair);
• if (ret < 0) {

• dev_err(dev, "failed to request asrc pair\n");
  

• goto req_pair_err;
  

• }
• /* Request a dummy dma channel, which will be released later. */
• tmp_chan = fsl_asrc_get_dma_channel(pair, dir);
• if (!tmp_chan) {

• dev_err(dev, "failed to get dma channel\n");
  

• ret = -EINVAL;
  

• goto dma_chan_err;
  

• }
• dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
• /* Refine the snd_imx_hardware according to caps of DMA. */
• ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream,

• 					dma_data,
  

• 					&snd_imx_hardware,
  

• 					tmp_chan);
  

• if (ret < 0) {

• dev_err(dev, "failed to refine runtime hwparams\n");
  

• goto out;
  

• }
• release_pair = false; snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);

• return 0;

+out:

• dma_release_channel(tmp_chan);

+dma_chan_err:

• fsl_asrc_release_pair(pair);

+req_pair_err:

• if (release_pair)

• kfree(pair);
  

• return ret;

}

static int fsl_asrc_dma_shutdown(struct snd_pcm_substream *substream)

2.21.0

The patch

ASoC: fsl_asrc: update supported sample format

has been applied to the asoc tree at

https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5

All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying to this mail.

Thanks, Mark

From 109539c986cee525e5ff9ae98793f23c2b29e54d Mon Sep 17 00:00:00 2001

From: Shengjiu Wang shengjiu.wang@nxp.com Date: Fri, 27 Sep 2019 09:46:10 +0800 Subject: [PATCH] ASoC: fsl_asrc: update supported sample format

The ASRC support 24bit/16bit/8bit input width, which is data width, not slot width.

For the S20_3LE format, the data with is 20bit, slot width is 24bit, if we set ASRMCR1n.IWD to be 24bits, the result is the volume is lower than expected, it likes 24bit data right shift 4 bits

So replace S20_3LE with S24_3LE in supported list and add S8 format in TX supported list

Signed-off-by: Shengjiu Wang shengjiu.wang@nxp.com Acked-by: Nicolin Chen nicoleotsuka@gmail.com Link: https://lore.kernel.org/r/45a7c383f43cc1dd9d0934846447aee653278c03.156949393... Signed-off-by: Mark Brown broonie@kernel.org --- sound/soc/fsl/fsl_asrc.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-)

diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 4d3804a1ea55..584badf956d2 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -624,7 +624,7 @@ static int fsl_asrc_dai_probe(struct snd_soc_dai *dai)

#define FSL_ASRC_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S20_3LE) + SNDRV_PCM_FMTBIT_S24_3LE)

static struct snd_soc_dai_driver fsl_asrc_dai = { .probe = fsl_asrc_dai_probe, @@ -635,7 +635,8 @@ static struct snd_soc_dai_driver fsl_asrc_dai = { .rate_min = 5512, .rate_max = 192000, .rates = SNDRV_PCM_RATE_KNOT, - .formats = FSL_ASRC_FORMATS, + .formats = FSL_ASRC_FORMATS | + SNDRV_PCM_FMTBIT_S8, }, .capture = { .stream_name = "ASRC-Capture",

The patch

ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width

has been applied to the asoc tree at

https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5

All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying to this mail.

Thanks, Mark

From 4bf62571070dd1021556e275d9221f736b2ffcf3 Mon Sep 17 00:00:00 2001

From: Shengjiu Wang shengjiu.wang@nxp.com Date: Fri, 27 Sep 2019 09:46:09 +0800 Subject: [PATCH] ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width

snd_pcm_format_t is more formal than enum asrc_word_width, which has two property, width and physical width, which is more accurate than enum asrc_word_width. So it is better to use in(out)put_format instead of in(out)put_word_width.

Signed-off-by: Shengjiu Wang shengjiu.wang@nxp.com Acked-by: Nicolin Chen nicoleotsuka@gmail.com Link: https://lore.kernel.org/r/7937c1404ee327ce141cb03b3575b02ea01a740c.156949393... Signed-off-by: Mark Brown broonie@kernel.org --- sound/soc/fsl/fsl_asrc.c | 56 +++++++++++++++++++++++++++------------- sound/soc/fsl/fsl_asrc.h | 4 +-- 2 files changed, 40 insertions(+), 20 deletions(-)

diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index cfa40ef6b1ca..4d3804a1ea55 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -265,6 +265,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; enum asrc_pair_index index = pair->index; + enum asrc_word_width input_word_width; + enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; u32 clk_index[2], div[2]; int in, out, channels; @@ -283,9 +285,32 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; }

- /* Validate output width */ - if (config->output_word_width == ASRC_WIDTH_8_BIT) { - pair_err("does not support 8bit width output\n"); + switch (snd_pcm_format_width(config->input_format)) { + case 8: + input_word_width = ASRC_WIDTH_8_BIT; + break; + case 16: + input_word_width = ASRC_WIDTH_16_BIT; + break; + case 24: + input_word_width = ASRC_WIDTH_24_BIT; + break; + default: + pair_err("does not support this input format, %d\n", + config->input_format); + return -EINVAL; + } + + switch (snd_pcm_format_width(config->output_format)) { + case 16: + output_word_width = ASRC_WIDTH_16_BIT; + break; + case 24: + output_word_width = ASRC_WIDTH_24_BIT; + break; + default: + pair_err("does not support this output format, %d\n", + config->output_format); return -EINVAL; }

@@ -383,8 +408,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* Implement word_width configurations */ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR1(index), ASRMCR1i_OW16_MASK | ASRMCR1i_IWD_MASK, - ASRMCR1i_OW16(config->output_word_width) | - ASRMCR1i_IWD(config->input_word_width)); + ASRMCR1i_OW16(output_word_width) | + ASRMCR1i_IWD(input_word_width));

/* Enable BUFFER STALL */ regmap_update_bits(asrc_priv->regmap, REG_ASRMCR(index), @@ -497,13 +522,13 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsl_asrc *asrc_priv = snd_soc_dai_get_drvdata(dai); - int width = params_width(params); struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_asrc_pair *pair = runtime->private_data; unsigned int channels = params_channels(params); unsigned int rate = params_rate(params); struct asrc_config config; - int word_width, ret; + snd_pcm_format_t format; + int ret;

ret = fsl_asrc_request_pair(channels, pair); if (ret) { @@ -513,15 +538,10 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream,

pair->config = &config;

- if (width == 16) - width = ASRC_WIDTH_16_BIT; - else - width = ASRC_WIDTH_24_BIT; - if (asrc_priv->asrc_width == 16) - word_width = ASRC_WIDTH_16_BIT; + format = SNDRV_PCM_FORMAT_S16_LE; else - word_width = ASRC_WIDTH_24_BIT; + format = SNDRV_PCM_FORMAT_S24_LE;

config.pair = pair->index; config.channel_num = channels; @@ -529,13 +549,13 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.outclk = OUTCLK_ASRCK1_CLK;

if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - config.input_word_width = width; - config.output_word_width = word_width; + config.input_format = params_format(params); + config.output_format = format; config.input_sample_rate = rate; config.output_sample_rate = asrc_priv->asrc_rate; } else { - config.input_word_width = word_width; - config.output_word_width = width; + config.input_format = format; + config.output_format = params_format(params); config.input_sample_rate = asrc_priv->asrc_rate; config.output_sample_rate = rate; } diff --git a/sound/soc/fsl/fsl_asrc.h b/sound/soc/fsl/fsl_asrc.h index c60075112570..38af485bdd22 100644 --- a/sound/soc/fsl/fsl_asrc.h +++ b/sound/soc/fsl/fsl_asrc.h @@ -342,8 +342,8 @@ struct asrc_config { unsigned int dma_buffer_size; unsigned int input_sample_rate; unsigned int output_sample_rate; - enum asrc_word_width input_word_width; - enum asrc_word_width output_word_width; + snd_pcm_format_t input_format; + snd_pcm_format_t output_format; enum asrc_inclk inclk; enum asrc_outclk outclk; };

On Wed, Jan 8, 2020 at 8:58 PM John Stultz john.stultz@linaro.org wrote:

On Thu, Sep 26, 2019 at 6:50 PM Shengjiu Wang shengjiu.wang@nxp.com wrote:

...

When set the runtime hardware parameters, we may need to query the capability of DMA to complete the parameters.

This patch is to Extract this operation from dmaengine_pcm_set_runtime_hwparams function to a separate function snd_dmaengine_pcm_refine_runtime_hwparams, that other components which need this feature can call this function.

Signed-off-by: Shengjiu Wang shengjiu.wang@nxp.com Reviewed-by: Nicolin Chen nicoleotsuka@gmail.com

As a heads up, this patch seems to be causing a regression on the HiKey board.

On boot up I'm seeing: [ 17.721424] hi6210_i2s f7118000.i2s: ASoC: can't open component f7118000.i2s: -6

And HDMI audio isn't working. With this patch reverted, audio works again.

...

diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 89a05926ac73..5749a8a49784 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -369,4 +369,87 @@ int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream)

...

...

•   ret = dma_get_slave_caps(chan, &dma_caps);
  

•   if (ret == 0) {
  

•           if (dma_caps.cmd_pause && dma_caps.cmd_resume)
  

•                   hw->info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME;
  

•           if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT)
  

•                   hw->info |= SNDRV_PCM_INFO_BATCH;
  

•           if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
  

•                   addr_widths = dma_caps.dst_addr_widths;
  

•           else
  

•                   addr_widths = dma_caps.src_addr_widths;
  

•   }
  

It seems a failing ret from dma_get_slave_caps() here is being returned...

...

•   /*
  

•    * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep
  

•    * hw.formats set to 0, meaning no restrictions are in place.
  

•    * In this case it's the responsibility of the DAI driver to
  

•    * provide the supported format information.
  

•    */
  

•   if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK))
  

•           /*
  

•            * Prepare formats mask for valid/allowed sample types. If the
  

•            * dma does not have support for the given physical word size,
  

•            * it needs to be masked out so user space can not use the
  

•            * format which produces corrupted audio.
  

•            * In case the dma driver does not implement the slave_caps the
  

•            * default assumption is that it supports 1, 2 and 4 bytes
  

•            * widths.
  

•            */
  

•           for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) {
  

•                   int bits = snd_pcm_format_physical_width(i);
  

•                   /*
  

•                    * Enable only samples with DMA supported physical
  

•                    * widths
  

•                    */
  

•                   switch (bits) {
  

•                   case 8:
  

•                   case 16:
  

•                   case 24:
  

•                   case 32:
  

•                   case 64:
  

•                           if (addr_widths & (1 << (bits / 8)))
  

•                                   hw->formats |= pcm_format_to_bits(i);
  

•                           break;
  

•                   default:
  

•                           /* Unsupported types */
  

•                           break;
  

•                   }
  

•           }
  

•   return ret;
  

... down here.

Where as in the old code...

...

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 748f5f641002..b9f147eaf7c4 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c

...

@@ -145,56 +140,12 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) hw.info |= SNDRV_PCM_INFO_BATCH;

•   ret = dma_get_slave_caps(chan, &dma_caps);
  

•   if (ret == 0) {
  

•           if (dma_caps.cmd_pause && dma_caps.cmd_resume)
  

•                   hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME;
  

•           if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT)
  

•                   hw.info |= SNDRV_PCM_INFO_BATCH;
  

•           if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
  

•                   addr_widths = dma_caps.dst_addr_widths;
  

•           else
  

•                   addr_widths = dma_caps.src_addr_widths;
  

•   }
  

...the ret from dma_get_slave_caps() checked above, but is not actually returned.

Suggestions on how to sort this out?

Just wanted to check in on this, as I'm still seeing this regression with -rc6.

thanks -john

On Thu, Jan 16, 2020 at 11:11 PM Shengjiu Wang shengjiu.wang@gmail.com wrote:

Hi

On Thu, Jan 16, 2020 at 1:56 PM John Stultz john.stultz@linaro.org wrote:

...

On Wed, Jan 8, 2020 at 8:58 PM John Stultz john.stultz@linaro.org wrote:

...

On Thu, Sep 26, 2019 at 6:50 PM Shengjiu Wang shengjiu.wang@nxp.com wrote:

...

When set the runtime hardware parameters, we may need to query the capability of DMA to complete the parameters.

This patch is to Extract this operation from dmaengine_pcm_set_runtime_hwparams function to a separate function snd_dmaengine_pcm_refine_runtime_hwparams, that other components which need this feature can call this function.

Signed-off-by: Shengjiu Wang shengjiu.wang@nxp.com Reviewed-by: Nicolin Chen nicoleotsuka@gmail.com

As a heads up, this patch seems to be causing a regression on the HiKey board.

On boot up I'm seeing: [ 17.721424] hi6210_i2s f7118000.i2s: ASoC: can't open component f7118000.i2s: -6

And HDMI audio isn't working. With this patch reverted, audio works again.

...

diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 89a05926ac73..5749a8a49784 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -369,4 +369,87 @@ int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream)

...

...

•   ret = dma_get_slave_caps(chan, &dma_caps);
  

•   if (ret == 0) {
  

•           if (dma_caps.cmd_pause && dma_caps.cmd_resume)
  

•                   hw->info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME;
  

•           if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT)
  

•                   hw->info |= SNDRV_PCM_INFO_BATCH;
  

•           if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
  

•                   addr_widths = dma_caps.dst_addr_widths;
  

•           else
  

•                   addr_widths = dma_caps.src_addr_widths;
  

•   }
  

It seems a failing ret from dma_get_slave_caps() here is being returned...

...

•   /*
  

•    * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep
  

•    * hw.formats set to 0, meaning no restrictions are in place.
  

•    * In this case it's the responsibility of the DAI driver to
  

•    * provide the supported format information.
  

•    */
  

•   if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK))
  

•           /*
  

•            * Prepare formats mask for valid/allowed sample types. If the
  

•            * dma does not have support for the given physical word size,
  

•            * it needs to be masked out so user space can not use the
  

•            * format which produces corrupted audio.
  

•            * In case the dma driver does not implement the slave_caps the
  

•            * default assumption is that it supports 1, 2 and 4 bytes
  

•            * widths.
  

•            */
  

•           for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) {
  

•                   int bits = snd_pcm_format_physical_width(i);
  

•                   /*
  

•                    * Enable only samples with DMA supported physical
  

•                    * widths
  

•                    */
  

•                   switch (bits) {
  

•                   case 8:
  

•                   case 16:
  

•                   case 24:
  

•                   case 32:
  

•                   case 64:
  

•                           if (addr_widths & (1 << (bits / 8)))
  

•                                   hw->formats |= pcm_format_to_bits(i);
  

•                           break;
  

•                   default:
  

•                           /* Unsupported types */
  

•                           break;
  

•                   }
  

•           }
  

•   return ret;
  

... down here.

Where as in the old code...

...

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 748f5f641002..b9f147eaf7c4 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c

...

@@ -145,56 +140,12 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) hw.info |= SNDRV_PCM_INFO_BATCH;

•   ret = dma_get_slave_caps(chan, &dma_caps);
  

•   if (ret == 0) {
  

•           if (dma_caps.cmd_pause && dma_caps.cmd_resume)
  

•                   hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME;
  

•           if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT)
  

•                   hw.info |= SNDRV_PCM_INFO_BATCH;
  

•           if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
  

•                   addr_widths = dma_caps.dst_addr_widths;
  

•           else
  

•                   addr_widths = dma_caps.src_addr_widths;
  

•   }
  

...the ret from dma_get_slave_caps() checked above, but is not actually returned.

Suggestions on how to sort this out?

Just wanted to check in on this, as I'm still seeing this regression with -rc6.

Compare with the old code. it seems that we shouldn't check the return value.

Could you help to test below changes?

--- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -138,12 +138,10 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component, if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) hw.info |= SNDRV_PCM_INFO_BATCH;

•   ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream,
  

•   snd_dmaengine_pcm_refine_runtime_hwparams(substream,
                                                    dma_data,
                                                    &hw,
                                                    chan);
  

•   if (ret)
  

•           return ret;
  
    return snd_soc_set_runtime_hwparams(substream, &hw);
  

}

Yes, thanks for taking a look at this! Your patch does appear to avoid the regression. (Though you'll want to drop the ret declaration to avoid "warning: unused variable 'ret'" compiler warnings.)

thanks -john