audio: Added SDL_AudioStream. Non-power-of-two resampling now works!
--HG-- extra : rebase_source : f6b2eb0c8a4d8869879112d60abe32962839fe7a
This commit is contained in:
Ryan C. Gordon
parent
02fd7b4a1e
commit
70608017eb
4 changed files with 436 additions and 109 deletions
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@ -26,6 +26,7 @@
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#include "SDL_audio_c.h"
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#include "SDL_assert.h"
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#include "../SDL_dataqueue.h"
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/* Effectively mix right and left channels into a single channel */
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@ -590,5 +591,280 @@ SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
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return (cvt->needed);
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}
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struct SDL_AudioStream
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{
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SDL_AudioCVT cvt_before_resampling;
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SDL_AudioCVT cvt_after_resampling;
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SDL_DataQueue *queue;
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Uint8 *work_buffer;
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int work_buffer_len;
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Uint8 *resample_buffer;
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int resample_buffer_len;
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int src_sample_frame_size;
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SDL_AudioFormat src_format;
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Uint8 src_channels;
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int src_rate;
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int dst_sample_frame_size;
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SDL_AudioFormat dst_format;
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Uint8 dst_channels;
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int dst_rate;
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double rate_incr;
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Uint8 pre_resample_channels;
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SDL_bool resampler_seeded;
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float resampler_state[8];
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int packetlen;
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};
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SDL_AudioStream *SDL_NewAudioStream(const SDL_AudioFormat src_format,
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const Uint8 src_channels,
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const int src_rate,
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const SDL_AudioFormat dst_format,
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const Uint8 dst_channels,
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const int dst_rate)
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{
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const int packetlen = 4096; /* !!! FIXME: good enough for now. */
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Uint8 pre_resample_channels;
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SDL_AudioStream *retval;
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retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
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if (!retval) {
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return NULL;
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}
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/* If increasing channels, do it after resampling, since we'd just
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do more work to resample duplicate channels. If we're decreasing, do
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it first so we resample the interpolated data instead of interpolating
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the resampled data (!!! FIXME: decide if that works in practice, though!). */
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pre_resample_channels = SDL_min(src_channels, dst_channels);
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retval->src_sample_frame_size = SDL_AUDIO_BITSIZE(src_format) * src_channels;
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retval->src_format = src_format;
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retval->src_channels = src_channels;
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retval->src_rate = src_rate;
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retval->dst_sample_frame_size = SDL_AUDIO_BITSIZE(dst_format) * dst_channels;
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retval->dst_format = dst_format;
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retval->dst_channels = dst_channels;
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retval->dst_rate = dst_rate;
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retval->pre_resample_channels = pre_resample_channels;
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retval->packetlen = packetlen;
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retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
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/* Not resampling? It's an easy conversion (and maybe not even that!). */
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if (src_rate == dst_rate) {
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retval->cvt_before_resampling.needed = SDL_FALSE;
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retval->cvt_before_resampling.len_mult = 1;
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if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) == -1) {
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SDL_free(retval);
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return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
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}
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} else {
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/* Don't resample at first. Just get us to Float32 format. */
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/* !!! FIXME: convert to int32 on devices without hardware float. */
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if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) == -1) {
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SDL_free(retval);
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return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
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}
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/* Convert us to the final format after resampling. */
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if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) == -1) {
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SDL_free(retval);
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return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
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}
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}
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retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
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if (!retval->queue) {
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SDL_free(retval);
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return NULL; /* SDL_NewDataQueue should have called SDL_SetError. */
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}
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return retval;
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}
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static int
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ResampleAudioStream(SDL_AudioStream *stream, const float *inbuf, const int inbuflen, float *outbuf, const int outbuflen)
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{
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/* !!! FIXME: this resampler sucks, but not much worse than our usual resampler. :) */ /* ... :( */
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const int chans = (int) stream->pre_resample_channels;
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const int framelen = chans * sizeof (float);
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const int total = (inbuflen / framelen);
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const int finalpos = total - chans;
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const double src_incr = 1.0 / stream->rate_incr;
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double idx = 0.0;
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float *dst = outbuf;
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float last_sample[SDL_arraysize(stream->resampler_state)];
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int consumed = 0;
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int i;
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SDL_assert(chans <= SDL_arraysize(last_sample));
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SDL_assert((inbuflen % framelen) == 0);
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if (!stream->resampler_seeded) {
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for (i = 0; i < chans; i++) {
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stream->resampler_state[i] = inbuf[i];
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}
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stream->resampler_seeded = SDL_TRUE;
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}
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for (i = 0; i < chans; i++) {
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last_sample[i] = stream->resampler_state[i];
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}
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while (consumed < total) {
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const int pos = ((int) idx) * chans;
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const float *src = &inbuf[(pos >= finalpos) ? finalpos : pos];
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SDL_assert(dst < (outbuf + (outbuflen / framelen)));
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for (i = 0; i < chans; i++) {
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const float val = *(src++);
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*(dst++) = (val + last_sample[i]) * 0.5f;
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last_sample[i] = val;
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}
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consumed = pos + chans;
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idx += src_incr;
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}
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for (i = 0; i < chans; i++) {
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stream->resampler_state[i] = last_sample[i];
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}
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return (dst - outbuf) * sizeof (float);
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}
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static Uint8 *
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EnsureBufferSize(Uint8 **buf, int *len, const int newlen)
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{
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if (*len < newlen) {
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void *ptr = SDL_realloc(*buf, newlen);
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if (!ptr) {
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SDL_OutOfMemory();
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return NULL;
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}
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*buf = (Uint8 *) ptr;
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*len = newlen;
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}
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return *buf;
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}
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int
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SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, const Uint32 _buflen)
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{
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int buflen = (int) _buflen;
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if (!stream) {
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return SDL_InvalidParamError("stream");
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} else if (!buf) {
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return SDL_InvalidParamError("buf");
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} else if (buflen == 0) {
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return 0; /* nothing to do. */
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} else if ((buflen % stream->src_sample_frame_size) != 0) {
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return SDL_SetError("Can't add partial sample frames");
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}
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if (stream->cvt_before_resampling.needed) {
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const int workbuflen = buflen * stream->cvt_before_resampling.len_mult; /* will be "* 1" if not needed */
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Uint8 *workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
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if (workbuf == NULL) {
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return -1; /* probably out of memory. */
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}
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SDL_memcpy(workbuf, buf, buflen);
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stream->cvt_before_resampling.buf = workbuf;
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stream->cvt_before_resampling.len = buflen;
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if (SDL_ConvertAudio(&stream->cvt_before_resampling) == -1) {
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return -1; /* uhoh! */
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}
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buf = workbuf;
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buflen = stream->cvt_before_resampling.len_cvt;
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}
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if (stream->dst_rate != stream->src_rate) {
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const int workbuflen = buflen * ((int) SDL_ceil(stream->rate_incr));
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float *workbuf = (float *) EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
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if (workbuf == NULL) {
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return -1; /* probably out of memory. */
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}
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buflen = ResampleAudioStream(stream, (float *) buf, buflen, workbuf, workbuflen);
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buf = workbuf;
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}
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if (stream->cvt_after_resampling.needed) {
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const int workbuflen = buflen * stream->cvt_before_resampling.len_mult; /* will be "* 1" if not needed */
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Uint8 *workbuf;
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if (buf == stream->resample_buffer) {
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workbuf = EnsureBufferSize(&stream->resample_buffer, &stream->resample_buffer_len, workbuflen);
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} else {
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const int inplace = (buf == stream->work_buffer);
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workbuf = EnsureBufferSize(&stream->work_buffer, &stream->work_buffer_len, workbuflen);
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if (workbuf && !inplace) {
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SDL_memcpy(workbuf, buf, buflen);
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}
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}
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if (workbuf == NULL) {
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return -1; /* probably out of memory. */
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}
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stream->cvt_after_resampling.buf = workbuf;
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stream->cvt_after_resampling.len = buflen;
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if (SDL_ConvertAudio(&stream->cvt_after_resampling) == -1) {
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return -1; /* uhoh! */
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}
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buf = workbuf;
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buflen = stream->cvt_after_resampling.len_cvt;
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}
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return SDL_WriteToDataQueue(stream->queue, buf, buflen);
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}
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void
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SDL_AudioStreamClear(SDL_AudioStream *stream)
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{
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if (!stream) {
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SDL_InvalidParamError("stream");
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} else {
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SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
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stream->resampler_seeded = SDL_FALSE;
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}
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}
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/* get converted/resampled data from the stream */
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int
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SDL_AudioStreamGet(SDL_AudioStream *stream, Uint32 len, void *buf, const Uint32 buflen)
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{
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if (!stream) {
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return SDL_InvalidParamError("stream");
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} else if (!buf) {
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return SDL_InvalidParamError("buf");
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} else if (len == 0) {
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return 0; /* nothing to do. */
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} else if ((len % stream->dst_sample_frame_size) != 0) {
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return SDL_SetError("Can't request partial sample frames");
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}
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return SDL_ReadFromDataQueue(stream->queue, buf, buflen);
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}
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/* number of converted/resampled bytes available */
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int
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SDL_AudioStreamAvailable(SDL_AudioStream *stream)
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{
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return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
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}
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/* dispose of a stream */
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void
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SDL_FreeAudioStream(SDL_AudioStream *stream)
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{
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if (stream) {
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SDL_FreeDataQueue(stream->queue);
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SDL_free(stream->work_buffer);
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SDL_free(stream->resample_buffer);
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SDL_free(stream);
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}
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}
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/* vi: set ts=4 sw=4 expandtab: */
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