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Merged
lgirdwood merged 3 commits into from
Aug 31, 2020
Merged

Pcm converter: return error code #3251

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c8c2af2
pcm_converter: Return error code from converting function
ktrzcinx Jul 10, 2020
c787abd
dma: Return error code from dma_buffer_copy_* function
ktrzcinx Jul 10, 2020
71ed7f5
dai: host: dma: Check returned value in dma_buffer_copy_* clients
ktrzcinx Jul 10, 2020
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26 changes: 22 additions & 4 deletions src/audio/dai.c
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Original file line number Diff line number Diff line change
Expand Up @@ -73,7 +73,10 @@ static void dai_dma_cb(void *arg, enum notify_id type, void *data)
struct comp_dev *dev = arg;
struct dai_data *dd = comp_get_drvdata(dev);
uint32_t bytes = next->elem.size;
struct comp_buffer *source;
struct comp_buffer *sink;
void *buffer_ptr;
int ret;

comp_dbg(dev, "dai_dma_cb()");

Expand All @@ -99,17 +102,32 @@ static void dai_dma_cb(void *arg, enum notify_id type, void *data)
}

if (dev->direction == SOF_IPC_STREAM_PLAYBACK) {
dma_buffer_copy_to(dd->local_buffer, dd->dma_buffer,
dd->process, bytes);
ret = dma_buffer_copy_to(dd->local_buffer, dd->dma_buffer,
dd->process, bytes);

buffer_ptr = dd->local_buffer->stream.r_ptr;
} else {
dma_buffer_copy_from(dd->dma_buffer, dd->local_buffer,
dd->process, bytes);
ret = dma_buffer_copy_from(dd->dma_buffer, dd->local_buffer,
dd->process, bytes);

buffer_ptr = dd->local_buffer->stream.w_ptr;
}

/* assert dma_buffer_copy succeed */
if (ret < 0) {
source = dev->direction == SOF_IPC_STREAM_PLAYBACK ?
dd->local_buffer : dd->dma_buffer;
sink = dev->direction == SOF_IPC_STREAM_PLAYBACK ?
dd->dma_buffer : dd->local_buffer;
comp_err(dev, "dai_dma_cb() dma buffer copy failed, dir %d bytes %d avail %d free %d",
dev->direction, bytes,
audio_stream_get_avail_samples(&source->stream) *
audio_stream_frame_bytes(&source->stream),
audio_stream_get_free_samples(&sink->stream) *
audio_stream_frame_bytes(&sink->stream));
return;
}

/* update host position (in bytes offset) for drivers */
dev->position += bytes;
if (dd->dai_pos) {
Expand Down
27 changes: 23 additions & 4 deletions src/audio/host.c
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Original file line number Diff line number Diff line change
Expand Up @@ -130,13 +130,32 @@ static uint32_t host_dma_get_split(struct host_data *hd, uint32_t bytes)
static void host_update_position(struct comp_dev *dev, uint32_t bytes)
{
struct host_data *hd = comp_get_drvdata(dev);
struct comp_buffer *source;
struct comp_buffer *sink;
int ret;


if (dev->direction == SOF_IPC_STREAM_PLAYBACK)
dma_buffer_copy_from(hd->dma_buffer, hd->local_buffer,
hd->process, bytes);
ret = dma_buffer_copy_from(hd->dma_buffer, hd->local_buffer,
hd->process, bytes);
else
dma_buffer_copy_to(hd->local_buffer, hd->dma_buffer,
hd->process, bytes);
ret = dma_buffer_copy_to(hd->local_buffer, hd->dma_buffer,
hd->process, bytes);

/* assert dma_buffer_copy succeed */
if (ret < 0) {
source = dev->direction == SOF_IPC_STREAM_PLAYBACK ?
hd->dma_buffer : hd->local_buffer;
sink = dev->direction == SOF_IPC_STREAM_PLAYBACK ?
hd->local_buffer : hd->dma_buffer;
comp_err(dev, "host_update_position() dma buffer copy failed, dir %d bytes %d avail %d free %d",
dev->direction, bytes,
audio_stream_get_avail_samples(&source->stream) *
audio_stream_frame_bytes(&source->stream),
audio_stream_get_free_samples(&sink->stream) *
audio_stream_frame_bytes(&sink->stream));
return;
}

dev->position += bytes;

Expand Down
15 changes: 10 additions & 5 deletions src/audio/pcm_converter/pcm_converter.c
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Original file line number Diff line number Diff line change
Expand Up @@ -14,9 +14,9 @@
#include <sof/audio/pcm_converter.h>
#include <sof/debug/panic.h>

void pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset,
struct audio_stream *sink, uint32_t ooffset,
uint32_t samples, pcm_converter_lin_func converter)
int pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset,
struct audio_stream *sink, uint32_t ooffset,
uint32_t samples, pcm_converter_lin_func converter)
{
const int s_size_in = audio_stream_sample_bytes(source);
const int s_size_out = audio_stream_sample_bytes(sink);
Expand All @@ -30,8 +30,11 @@ void pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset,
int chunk;
int N1, N2;

assert(audio_stream_get_avail_samples(source) >= samples + ioffset);
assert(audio_stream_get_free_samples(sink) >= samples + ooffset);
/* assert enough avail/free samples in source and sink buffer */
if (audio_stream_get_avail_samples(source) < samples + ioffset)
return -EINVAL;
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lgirdwood marked this conversation as resolved.
Resolved
if (audio_stream_get_free_samples(sink) < samples + ooffset)
return -EINVAL;

while (i < samples) {
/* calculate chunk size */
Expand All @@ -51,4 +54,6 @@ void pcm_convert_as_linear(const struct audio_stream *source, uint32_t ioffset,
w_ptr = audio_stream_wrap(sink, w_ptr + chunk * s_size_out);
i += chunk;
}

return samples;
}
108 changes: 60 additions & 48 deletions src/audio/pcm_converter/pcm_converter_generic.c
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Original file line number Diff line number Diff line change
Expand Up @@ -26,9 +26,9 @@

#if CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S24LE

static void pcm_convert_s16_to_s24(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_s16_to_s24(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
uint32_t buff_frag = 0;
int16_t *src;
Expand All @@ -41,11 +41,13 @@ static void pcm_convert_s16_to_s24(const struct audio_stream *source,
*dst = *src << 8;
buff_frag++;
}

return samples;
}

static void pcm_convert_s24_to_s16(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_s24_to_s16(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
uint32_t buff_frag = 0;
int32_t *src;
Expand All @@ -58,15 +60,17 @@ static void pcm_convert_s24_to_s16(const struct audio_stream *source,
*dst = sat_int16(Q_SHIFT_RND(sign_extend_s24(*src), 23, 15));
buff_frag++;
}

return samples;
}

#endif /* CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S24LE */

#if CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S32LE

static void pcm_convert_s16_to_s32(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_s16_to_s32(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
uint32_t buff_frag = 0;
int16_t *src;
Expand All @@ -79,11 +83,13 @@ static void pcm_convert_s16_to_s32(const struct audio_stream *source,
*dst = *src << 16;
buff_frag++;
}

return samples;
}

static void pcm_convert_s32_to_s16(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_s32_to_s16(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
uint32_t buff_frag = 0;
int32_t *src;
Expand All @@ -96,15 +102,17 @@ static void pcm_convert_s32_to_s16(const struct audio_stream *source,
*dst = sat_int16(Q_SHIFT_RND(*src, 31, 15));
buff_frag++;
}

return samples;
}

#endif /* CONFIG_FORMAT_S16LE && CONFIG_FORMAT_S32LE */

#if CONFIG_FORMAT_S24LE && CONFIG_FORMAT_S32LE

static void pcm_convert_s24_to_s32(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_s24_to_s32(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
uint32_t buff_frag = 0;
int32_t *src;
Expand All @@ -117,11 +125,13 @@ static void pcm_convert_s24_to_s32(const struct audio_stream *source,
*dst = *src << 8;
buff_frag++;
}

return samples;
}

static void pcm_convert_s32_to_s24(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_s32_to_s24(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
uint32_t buff_frag = 0;
int32_t *src;
Expand All @@ -134,6 +144,8 @@ static void pcm_convert_s32_to_s24(const struct audio_stream *source,
*dst = sat_int24(Q_SHIFT_RND(*src, 31, 23));
buff_frag++;
}

return samples;
}

#endif /* CONFIG_FORMAT_S24LE && CONFIG_FORMAT_S32LE */
Expand Down Expand Up @@ -272,20 +284,20 @@ static void pcm_convert_f_to_s16_lin(const void *psrc, void *pdst,
dst[i] = sat_int16(_pcm_convert_f_to_i(src[i], 15));
}

static void pcm_convert_s16_to_f(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_s16_to_f(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s16_to_f_lin);
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s16_to_f_lin);
}

static void pcm_convert_f_to_s16(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_f_to_s16(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_f_to_s16_lin);
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_f_to_s16_lin);
}
#endif /* CONFIG_FORMAT_FLOAT && CONFIG_FORMAT_S16LE */

Expand Down Expand Up @@ -316,20 +328,20 @@ static void pcm_convert_f_to_s24_lin(const void *psrc, void *pdst,
dst[i] = sat_int24(_pcm_convert_f_to_i(src[i], 23));
}

static void pcm_convert_s24_to_f(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_s24_to_f(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s24_to_f_lin);
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s24_to_f_lin);
}

static void pcm_convert_f_to_s24(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_f_to_s24(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_f_to_s24_lin);
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_f_to_s24_lin);
}
#endif /* CONFIG_FORMAT_FLOAT && CONFIG_FORMAT_S24LE */

Expand Down Expand Up @@ -360,20 +372,20 @@ static void pcm_convert_f_to_s32_lin(const void *psrc, void *pdst,
dst[i] = _pcm_convert_f_to_i(src[i], 31);
}

static void pcm_convert_s32_to_f(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_s32_to_f(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s32_to_f_lin);
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_s32_to_f_lin);
}

static void pcm_convert_f_to_s32(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
static int pcm_convert_f_to_s32(const struct audio_stream *source,
uint32_t ioffset, struct audio_stream *sink,
uint32_t ooffset, uint32_t samples)
{
pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_f_to_s32_lin);
return pcm_convert_as_linear(source, ioffset, sink, ooffset, samples,
pcm_convert_f_to_s32_lin);
}
#endif /* CONFIG_FORMAT_FLOAT && CONFIG_FORMAT_S32LE */

Expand Down
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