static int TIMIDITY_GetSome(void *context, void *data, int bytes, SDL_bool *done)
{
TIMIDITY_Music *music = (TIMIDITY_Music *)context;
int filled, amount, expected;
if (music->stream) {
filled = SDL_AudioStreamGet(music->stream, data, bytes);
if (filled != 0) {
return filled;
}
}
if (!music->play_count) {
/* All done */
*done = SDL_TRUE;
return 0;
}
if (music->stream) {
expected = music->buffer_size;
amount = Timidity_PlaySome(music->song, music->buffer, music->buffer_size);
if (SDL_AudioStreamPut(music->stream, music->buffer, amount) < 0) {
return -1;
}
} else {
expected = bytes;
amount = Timidity_PlaySome(music->song, data, bytes);
}
if (amount < expected) {
if (music->play_count == 1) {
/* We didn't consume anything and we're done */
music->play_count = 0;
} else {
int play_count = -1;
if (music->play_count > 0) {
play_count = (music->play_count - 1);
}
if (TIMIDITY_Play(music, play_count) < 0) {
return -1;
}
}
}
if (music->stream) {
/* We'll pick it up from the stream next time around */
return 0;
} else {
/* We wrote output data */
return amount;
}
}
// stereo LRLRLR order
void mixerCallback( void* userdata, Uint8* streamData, int len )
{
memset( streamData, len, workingSilence );
memset( workingBuffer, 0, workingBufferSize );
if( workingBuffer == NULL ) {
return;
}
int numSamples = ( len / WORKING_CHANNELS ) / ( ( SDL_AUDIO_MASK_BITSIZE & WORKING_FORMAT ) / 8 );
#if 0
int soundFreq = 440; // wave length
float step = 1.0f / WORKING_RATE;
// for testing audio output
for( int s = 0; s < numSamples; ++s ) {
int streamIdx = ( s * WORKING_CHANNELS );
testTimePassed += 1.0f / WORKING_RATE;
float v = sinf( testTimePassed * soundFreq * M_TWO_PI_F );
workingBuffer[streamIdx] = v * 0.1f;
workingBuffer[streamIdx+1] = v * 0.1f;
}
#else
// advance each playing sound
for( EntityID id = idSet_GetFirstValidID( &playingIDSet ); id != INVALID_ENTITY_ID; id = idSet_GetNextValidID( &playingIDSet, id ) ) {
int i = idSet_GetIndex( id );
Sound* snd = &( playingSounds[i] );
Sample* sample = &( samples[snd->sample] );
// for right now lets assume the pitch will stay the same, when we get it working it'll just involve
// changing the speed at which we move through the array
bool soundDone = false;
float volume = snd->volume * sbSoundGroups[snd->group].volume * masterVolume;
for( int s = 0; ( s < numSamples ) && !soundDone; ++s ) {
int streamIdx = ( s * WORKING_CHANNELS );
// we're assuming stereo output here
if( sample->numChannels == 1 ) {
float data = sample->data[(int)snd->pos] * volume;
/* left */ workingBuffer[streamIdx] += data * inverseLerp( 1.0f, 0.0f, snd->pan );
/* right */ workingBuffer[streamIdx+1] += data * inverseLerp( -1.0f, 0.0f, snd->pan );
snd->pos += snd->pitch;
} else {
// if the sample is stereo then we ignore panning
// NOTE: Pitch change doesn't work with stereo samples yet
workingBuffer[streamIdx] += sample->data[(int)snd->pos] * volume;
workingBuffer[streamIdx+1] += sample->data[(int)snd->pos+1] * volume;
snd->pos += 2.0f;
}
if( snd->pos >= sample->numSamples ) {
if( sample->loops ) {
snd->pos -= (float)sample->numSamples;
} else {
soundDone = true;
}
}
}
if( soundDone ) {
idSet_ReleaseID( &playingIDSet, id ); // this doesn't invalidate the id for the loop
}
}
for( int i = 0; i < MAX_STREAMING_SOUNDS; ++i ) {
if( !streamingSounds[i].playing ) continue;
StreamingSound* stream = &( streamingSounds[i] );
bool soundDone = false;
float volume = stream->volume * sbSoundGroups[stream->group].volume * masterVolume;
// if the next buffer fill would be past what we have loaded then load some more
// note: the SDL_AudioStreamAvailable return value is in bytes
while( ( SDL_AudioStreamAvailable( stream->sdlStream ) < len ) && !( stream->readDone ) ) {
int read = 0;
int request = STREAM_READ_BUFFER_SIZE / sizeof( short );
size_t test = ARRAY_SIZE( streamReadBuffer );
// returns the number of samples stored per channel
int samplesPerChannel = stb_vorbis_get_samples_short_interleaved(
stream->access, stream->access->channels, streamReadBuffer, request );
read = samplesPerChannel * sizeof( short );
SDL_AudioStreamPut( stream->sdlStream, streamReadBuffer, samplesPerChannel * stream->channels * sizeof( short ) );
// reached the end of the file, are we looping?
if( read != request ) {
if( stream->loops ) {
stb_vorbis_seek_start( stream->access );
} else {
stream->readDone = true;
SDL_AudioStreamFlush( stream->sdlStream );
}
}
}
// read data from the audio stream until there is no more left or the end of the buffer to fill
int bytesToStream = numSamples * stream->channels * sizeof( sbStreamWorkingBuffer[0] );
sb_Reserve( sbStreamWorkingBuffer, (size_t)bytesToStream );
int gotten = SDL_AudioStreamGet( stream->sdlStream, sbStreamWorkingBuffer, bytesToStream );
if( gotten < 0 ) {
llog( LOG_ERROR, "Error reading from sdlStream: %s", SDL_GetError( ) );
stream->playing = false;
continue;
} else if( gotten == 0 ) {
// end of stream
stream->playing = false;
}
int samplesGotten = gotten / ( stream->channels * sizeof( sbStreamWorkingBuffer[0] ) );
for( int s = 0; s < samplesGotten; ++s ) {
// then just mix those samples
int streamIdx = ( s * WORKING_CHANNELS );
int workingIdx = ( s * stream->channels );
// we're assuming stereo output here
if( stream->access->channels == 1 ) {
float data = sbStreamWorkingBuffer[workingIdx] * volume;
workingBuffer[streamIdx] += data * inverseLerp( 1.0f, 0.0f, stream->pan ); // left
workingBuffer[streamIdx+1] += data * inverseLerp( -1.0f, 0.0f, stream->pan ); // right
} else {
// if the sample is stereo then we ignore panning
workingBuffer[streamIdx] += sbStreamWorkingBuffer[workingIdx] * volume;
workingBuffer[streamIdx+1] += sbStreamWorkingBuffer[workingIdx + 1] * volume;
}
}
}
#endif
memcpy( streamData, workingBuffer, len );
}
