[alsa-devel] [PATCH 0/3] update supported sample format
update supported sample format
Shengjiu Wang (3): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8
sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++++--------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 93 +++++++++++++++++++++++++++++++++--- 3 files changed, 135 insertions(+), 30 deletions(-)
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 --- 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 Mon, Sep 09, 2019 at 06:33:19PM -0400, Shengjiu Wang wrote:
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.
Hmm...I don't really see the benefit of using snd_pcm_format_t here...I mean, I know it's a generic one, and would understand if we use it as a param for a common API. But this patch merely packs the "width" by intentionally using this snd_pcm_format_t and then adds another translation to unpack it.. I feel it's a bit overcomplicated. Or am I missing something?
And I feel it's not necessary to use ALSA common format in our own "struct asrc_config" since it is more IP/register specific.
Thanks Nicolin
Signed-off-by: Shengjiu Wang shengjiu.wang@nxp.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",
return -EINVAL; }config->output_format);
@@ -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;
elseformat = SNDRV_PCM_FORMAT_S16_LE;
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.input_sample_rate = rate; config.output_sample_rate = asrc_priv->asrc_rate; } else {config.output_format = format;
config.input_word_width = word_width;
config.output_word_width = width;
config.input_format = format;
config.input_sample_rate = asrc_priv->asrc_rate; config.output_sample_rate = rate; }config.output_format = params_format(params);
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;
};
2.21.0
Hi
On Mon, Sep 09, 2019 at 06:33:19PM -0400, Shengjiu Wang wrote:
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.
Hmm...I don't really see the benefit of using snd_pcm_format_t here...I mean, I know it's a generic one, and would understand if we use it as a param for a common API. But this patch merely packs the "width" by intentionally using this snd_pcm_format_t and then adds another translation to unpack it.. I feel it's a bit overcomplicated. Or am I missing something?
And I feel it's not necessary to use ALSA common format in our own "struct asrc_config" since it is more IP/register specific.
Thanks Nicolin
As you know, we have another M2M function internally, when user want to Set the format through M2M API, it is better to use snd_pcm_format_t instead the Width, for snd_pcm_format_t include two property, data with and physical width In driver some place need data width, some place need physical width. For example how to distinguish S24_LE and S24_3LE in driver, DMA setting needs The physical width, but ASRC need data width.
Another purpose is that we have another new designed ASRC, which support more Formats, I would like it can share same API with this ASRC, using snd_pcm_format_t That we can use the common API, like snd_pcm_format_linear, snd_pcm_format_big_endian to get the property of the format, which is needed by driver.
Best regards Wang shengjiu
On Tue, Sep 10, 2019 at 02:22:06AM +0000, S.j. Wang wrote:
Hi
On Mon, Sep 09, 2019 at 06:33:19PM -0400, Shengjiu Wang wrote:
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.
Hmm...I don't really see the benefit of using snd_pcm_format_t here...I mean, I know it's a generic one, and would understand if we use it as a param for a common API. But this patch merely packs the "width" by intentionally using this snd_pcm_format_t and then adds another translation to unpack it.. I feel it's a bit overcomplicated. Or am I missing something?
And I feel it's not necessary to use ALSA common format in our own "struct asrc_config" since it is more IP/register specific.
Thanks Nicolin
As you know, we have another M2M function internally, when user want to Set the format through M2M API, it is better to use snd_pcm_format_t instead the Width, for snd_pcm_format_t include two property, data with and physical width In driver some place need data width, some place need physical width. For example how to distinguish S24_LE and S24_3LE in driver, DMA setting needs The physical width, but ASRC need data width.
Another purpose is that we have another new designed ASRC, which support more Formats, I would like it can share same API with this ASRC, using snd_pcm_format_t That we can use the common API, like snd_pcm_format_linear, snd_pcm_format_big_endian to get the property of the format, which is needed by driver.
I see. Just acked the patch.
On Mon, Sep 09, 2019 at 06:33:19PM -0400, Shengjiu Wang wrote:
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",
return -EINVAL; }config->output_format);
@@ -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;
elseformat = SNDRV_PCM_FORMAT_S16_LE;
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.input_sample_rate = rate; config.output_sample_rate = asrc_priv->asrc_rate; } else {config.output_format = format;
config.input_word_width = word_width;
config.output_word_width = width;
config.input_format = format;
config.input_sample_rate = asrc_priv->asrc_rate; config.output_sample_rate = rate; }config.output_format = params_format(params);
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;
};
2.21.0
The ASRC support 24bit/16bit/8bit input width, so S20_3LE format should not be supported, it is word width is 20bit. 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 --- 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",
On Mon, Sep 09, 2019 at 06:33:20PM -0400, Shengjiu Wang wrote:
The ASRC support 24bit/16bit/8bit input width, so S20_3LE format should not be supported, it is word width is 20bit.
I thought 3LE used 24-bit physical width. And the driver assigns ASRC_WIDTH_24_BIT to "width" for all non-16bit cases, so 20-bit would go for that 24-bit slot also. I don't clearly recall if I had explicitly tested S20_3LE, but I feel it should work since I put there...
Thanks Nicolin
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
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 |
}, .capture = { .stream_name = "ASRC-Capture",SNDRV_PCM_FMTBIT_S8,
-- 2.21.0
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 --- sound/soc/fsl/fsl_asrc.c | 4 +- sound/soc/fsl/fsl_asrc.h | 3 ++ sound/soc/fsl/fsl_asrc_dma.c | 93 +++++++++++++++++++++++++++++++++--- 3 files changed, 92 insertions(+), 8 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..30e27917016e 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 */ @@ -276,6 +274,16 @@ static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) struct device *dev = component->dev; struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); struct fsl_asrc_pair *pair; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u8 dir = tx ? OUT : IN; + struct dma_slave_caps dma_caps; + struct dma_chan *tmp_chan; + struct snd_dmaengine_dai_dma_data *dma_data; + u32 addr_widths = BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | + BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | + BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); + int ret; + int i;
pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL); if (!pair) @@ -285,8 +293,81 @@ 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); + 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; + } + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + /* Request a temp pair, which is release in the end */ + fsl_asrc_request_pair(1, pair); + + tmp_chan = fsl_asrc_get_dma_channel(pair, dir); + if (!tmp_chan) { + dev_err(dev, "can't get dma channel\n"); + return -EINVAL; + } + + ret = dma_get_slave_caps(tmp_chan, &dma_caps); + if (ret == 0) { + if (dma_caps.cmd_pause) + snd_imx_hardware.info |= SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME; + if (dma_caps.residue_granularity <= + DMA_RESIDUE_GRANULARITY_SEGMENT) + snd_imx_hardware.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 = 0; 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))) + snd_imx_hardware.formats |= (1LL << i); + break; + default: + /* Unsupported types */ + break; + } + } + + if (tmp_chan) + dma_release_channel(tmp_chan); + fsl_asrc_release_pair(pair); + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
return 0;
On Mon, Sep 09, 2019 at 06:33:21PM -0400, 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.
@@ -285,8 +293,81 @@ 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);
- 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;
- }
- dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
- /* Request a temp pair, which is release in the end */
- fsl_asrc_request_pair(1, pair);
Not sure if it'd be practical, but a pair request could fail. Will probably need to check return value.
And a quick feeling is that below code is mostly identical to what is in the soc-generic-dmaengine-pcm.c file. So I'm wondering if we could abstract a helper function somewhere in the ASoC core: Mark?
Thanks Nicolin
tmp_chan = fsl_asrc_get_dma_channel(pair, dir);
if (!tmp_chan) {
dev_err(dev, "can't get dma channel\n");
return -EINVAL;
}
ret = dma_get_slave_caps(tmp_chan, &dma_caps);
if (ret == 0) {
if (dma_caps.cmd_pause)
snd_imx_hardware.info |= SNDRV_PCM_INFO_PAUSE |
SNDRV_PCM_INFO_RESUME;
if (dma_caps.residue_granularity <=
DMA_RESIDUE_GRANULARITY_SEGMENT)
snd_imx_hardware.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 = 0; 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)))
snd_imx_hardware.formats |= (1LL << i);
break;
default:
/* Unsupported types */
break;
}
}
if (tmp_chan)
dma_release_channel(tmp_chan);
fsl_asrc_release_pair(pair);
snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware);
return 0;
-- 2.21.0
Hi
On Mon, Sep 09, 2019 at 06:33:21PM -0400, 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.
@@ -285,8 +293,81 @@ 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);
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;
}
dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
/* Request a temp pair, which is release in the end */
fsl_asrc_request_pair(1, pair);
Not sure if it'd be practical, but a pair request could fail. Will probably need to check return value.
And a quick feeling is that below code is mostly identical to what is in the soc-generic-dmaengine-pcm.c file. So I'm wondering if we could abstract a helper function somewhere in the ASoC core: Mark?
Thanks Nicolin
Yes, it refers to the code in soc-generic-dmaengine-pcm.c, if there is a common API, this is helpful.
Best regards Wang shengjiu
On Mon, Sep 09, 2019 at 06:52:13PM -0700, Nicolin Chen wrote:
And a quick feeling is that below code is mostly identical to what is in the soc-generic-dmaengine-pcm.c file. So I'm wondering if we could abstract a helper function somewhere in the ASoC core: Mark?
That's roughly what sound/core/pcm_dmaengine.c is doing - possibly we should move more stuff into there.
On Wed, Sep 11, 2019 at 12:08:07PM +0100, Mark Brown wrote:
On Mon, Sep 09, 2019 at 06:52:13PM -0700, Nicolin Chen wrote:
And a quick feeling is that below code is mostly identical to what is in the soc-generic-dmaengine-pcm.c file. So I'm wondering if we could abstract a helper function somewhere in the ASoC core: Mark?
That's roughly what sound/core/pcm_dmaengine.c is doing - possibly we should move more stuff into there.
It looks like a right place to me. Thank you!
participants (4)
-
Mark Brown
-
Nicolin Chen
-
S.j. Wang
-
Shengjiu Wang