Added a staging buffer to the audio stream so that we can accumulate small amounts of data if needed when resampling
diff --git a/src/audio/SDL_audiocvt.c b/src/audio/SDL_audiocvt.c
index 25fe903..5bc7016 100644
--- a/src/audio/SDL_audiocvt.c
+++ b/src/audio/SDL_audiocvt.c
@@ -1083,6 +1083,9 @@
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 @@
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 @@
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 @@
!!! 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 @@
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 @@
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);
