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Added a staging buffer to the audio stream so that we can accumulate small amounts of data if needed when resampling

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This commit is contained in:
Sam Lantinga 2017-10-18 19:26:36 -07:00
parent 80f8464d97
commit 653ab5d9c4
1 changed files with 82 additions and 28 deletions
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110
src/audio/SDL_audiocvt.c
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@ -1083,6 +1083,9 @@ struct _SDL_AudioStream
SDL_AudioCVT cvt_after_resampling;
SDL_DataQueue *queue;
SDL_bool first_run;
Uint8 *staging_buffer;
int staging_buffer_size;
int staging_buffer_filled;
Uint8 *work_buffer_base; /* maybe unaligned pointer from SDL_realloc(). */
int work_buffer_len;
int src_sample_frame_size;
@ -1293,7 +1296,17 @@ SDL_NewAudioStream(const SDL_AudioFormat src_format,
return NULL;
}
/* Not resampling? It's an easy conversion (and maybe not even that!). */
retval->staging_buffer_size = ((retval->resampler_padding_samples / retval->pre_resample_channels) * retval->src_sample_frame_size);
if (retval->staging_buffer_size > 0) {
retval->staging_buffer = (Uint8 *) SDL_malloc(retval->staging_buffer_size);
if (retval->resampler_padding == NULL) {
SDL_FreeAudioStream(retval);
SDL_OutOfMemory();
return NULL;
}
}
/* Not resampling? It's an easy conversion (and maybe not even that!) */
if (src_rate == dst_rate) {
retval->cvt_before_resampling.needed = SDL_FALSE;
if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
@ -1348,8 +1361,8 @@ SDL_NewAudioStream(const SDL_AudioFormat src_format,
return retval;
}
int
SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len)
static int
SDL_AudioStreamPutInternal(SDL_AudioStream *stream, const void *buf, int len)
{
int buflen = len;
int workbuflen;
@ -1367,36 +1380,11 @@ SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len)
!!! FIXME: isn't a multiple of 16. In these cases, we should chop off
!!! FIXME: a few samples at the end and convert them separately. */
#if DEBUG_AUDIOSTREAM
printf("AUDIOSTREAM: wants to put %d preconverted bytes\n", buflen);
#endif
if (!stream) {
return SDL_InvalidParamError("stream");
} else if (!buf) {
return SDL_InvalidParamError("buf");
} else if (buflen == 0) {
return 0; /* nothing to do. */
} else if ((buflen % stream->src_sample_frame_size) != 0) {
return SDL_SetError("Can't add partial sample frames");
} else if (buflen < ((stream->resampler_padding_samples / stream->pre_resample_channels) * stream->src_sample_frame_size)) {
return SDL_SetError("Need to put a larger buffer");
}
/* no padding prepended on first run. */
neededpaddingbytes = stream->resampler_padding_samples * sizeof (float);
paddingbytes = stream->first_run ? 0 : neededpaddingbytes;
stream->first_run = SDL_FALSE;
if (!stream->cvt_before_resampling.needed &&
(stream->dst_rate == stream->src_rate) &&
!stream->cvt_after_resampling.needed) {
#if DEBUG_AUDIOSTREAM
printf("AUDIOSTREAM: no conversion needed at all, queueing %d bytes.\n", buflen);
#endif
return SDL_WriteToDataQueue(stream->queue, buf, buflen);
}
/* Make sure the work buffer can hold all the data we need at once... */
workbuflen = buflen;
if (stream->cvt_before_resampling.needed) {
@ -1495,6 +1483,71 @@ SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len)
return buflen ? SDL_WriteToDataQueue(stream->queue, resamplebuf, buflen) : 0;
}
int
SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len)
{
/* !!! FIXME: several converters can take advantage of SIMD, but only
!!! FIXME: if the data is aligned to 16 bytes. EnsureStreamBufferSize()
!!! FIXME: guarantees the buffer will align, but the
!!! FIXME: converters will iterate over the data backwards if
!!! FIXME: the output grows, and this means we won't align if buflen
!!! FIXME: isn't a multiple of 16. In these cases, we should chop off
!!! FIXME: a few samples at the end and convert them separately. */
#if DEBUG_AUDIOSTREAM
printf("AUDIOSTREAM: wants to put %d preconverted bytes\n", buflen);
#endif
if (!stream) {
return SDL_InvalidParamError("stream");
} else if (!buf) {
return SDL_InvalidParamError("buf");
} else if (len == 0) {
return 0; /* nothing to do. */
} else if ((len % stream->src_sample_frame_size) != 0) {
return SDL_SetError("Can't add partial sample frames");
}
if (!stream->cvt_before_resampling.needed &&
(stream->dst_rate == stream->src_rate) &&
!stream->cvt_after_resampling.needed) {
#if DEBUG_AUDIOSTREAM
printf("AUDIOSTREAM: no conversion needed at all, queueing %d bytes.\n", len);
#endif
return SDL_WriteToDataQueue(stream->queue, buf, len);
}
while (len > 0) {
int amount;
/* If we don't have a staging buffer or we're given enough data that
we don't need to store it for later, skip the staging process.
*/
if (!stream->staging_buffer_filled && len >= stream->staging_buffer_size) {
return SDL_AudioStreamPutInternal(stream, buf, len);
}
/* If there's not enough data to fill the staging buffer, just save it */
if ((stream->staging_buffer_filled + len) < stream->staging_buffer_size) {
SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, len);
stream->staging_buffer_filled += len;
return 0;
}
/* Fill the staging buffer, process it, and continue */
amount = (stream->staging_buffer_size - stream->staging_buffer_filled);
SDL_assert(amount > 0);
SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, amount);
stream->staging_buffer_filled = 0;
if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size) < 0) {
return -1;
}
buf = (void *)((Uint8 *)buf + amount);
len -= amount;
}
return 0;
}
/* get converted/resampled data from the stream */
int
SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len)
@ -1546,6 +1599,7 @@ SDL_FreeAudioStream(SDL_AudioStream *stream)
stream->cleanup_resampler_func(stream);
}
SDL_FreeDataQueue(stream->queue);
SDL_free(stream->staging_buffer);
SDL_free(stream->work_buffer_base);
SDL_free(stream->resampler_padding);
SDL_free(stream);