diff --git a/src/audio/SDL_audiocvt.c b/src/audio/SDL_audiocvt.c index 25fe903f2..5bc70160d 100644 --- a/src/audio/SDL_audiocvt.c +++ b/src/audio/SDL_audiocvt.c @@ -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);