void YSE::INTERNAL::soundFile::loadStreaming() {
File file;
if (IO().getActive()) {
// will be deleted by AudioFormatReader
customFileReader * cfr = new customFileReader;
cfr->create(fileName.c_str());
streamReader = SOUND::Manager().getReader(cfr);
}
else {
file = File::getCurrentWorkingDirectory().getChildFile(juce::String(fileName));
streamReader = SOUND::Manager().getReader(file);
}
if (streamReader != nullptr) {
_fileBuffer.setSize(streamReader->numChannels, STREAM_BUFFERSIZE);
// sample rate adjustment
_sampleRateAdjustment = static_cast<Flt>(streamReader->sampleRate) / static_cast<Flt>(SAMPLERATE);
_length = (Int)streamReader->lengthInSamples;
_buffer = _fileBuffer.getArrayOfReadPointers();
_channels = _fileBuffer.getNumChannels();
_streamPos = 0;
fillStream(false);
// file is ready for use now
state = READY;
}
else {
LogImpl().emit(E_FILEREADER, "Unable to read " + file.getFullPathName().toStdString());
state = INVALID;
}
}
void YSE::INTERNAL::soundFile::run() {
if (_streaming) {
if (source == nullptr) {
if (IO().getActive()) {
// will be deleted by AudioFormatReader
customFileReader * cfr = new customFileReader;
cfr->create(fileName.c_str());
streamReader = SOUND::Manager().getReader(cfr);
}
else {
streamReader = SOUND::Manager().getReader(file);
}
}
else {
streamReader = SOUND::Manager().getReader(source);
}
if (streamReader != nullptr) {
_buffer.setSize(streamReader->numChannels, STREAM_BUFFERSIZE);
// sample rate adjustment
_sampleRateAdjustment = static_cast<Flt>(streamReader->sampleRate) / static_cast<Flt>(SAMPLERATE);
_length = (Int)streamReader->lengthInSamples;
_streamPos = 0;
fillStream(false);
// file is ready for use now
state = READY;
return;
}
else {
LogImpl().emit(E_FILEREADER, "Unable to read " + file.getFullPathName().toStdString());
state = INVALID;
return;
}
}
// load non streaming sounds in one go
ScopedPointer<AudioFormatReader> reader;
if (source == nullptr) {
if (IO().getActive()) {
// will be deleted by AudioFormatReader
customFileReader * cfr = new customFileReader;
cfr->create(fileName.c_str());
reader = SOUND::Manager().getReader(cfr);
}
else {
reader = SOUND::Manager().getReader(file);
}
}
else {
reader = SOUND::Manager().getReader(source);
}
if (reader != nullptr) {
_buffer.setSize(reader->numChannels, (Int)reader->lengthInSamples);
reader->read(&_buffer, 0, (Int)reader->lengthInSamples, 0, true, true);
// sample rate adjustment
_sampleRateAdjustment = static_cast<Flt>(reader->sampleRate) / static_cast<Flt>(SAMPLERATE);
_length = _buffer.getNumSamples();
// file is ready for use now
state = READY;
return;
}
else {
LogImpl().emit(E_FILEREADER, "Unable to read " + file.getFullPathName().toStdString());
state = INVALID;
return;
}
}
Bool YSE::INTERNAL::soundFile::read(std::vector<DSP::sample> & filebuffer, Flt& pos, UInt length, Flt speed, Bool loop, SOUND_STATUS & intent, Flt & volume) {
/** Yes, this function uses goto...
It is highly optimized for speed and this is the best way I could find
to ensure good performance. Suggestions are welcome.
*/
if (state != READY) return false;
// adjust speed for sample rate
speed *= _sampleRateAdjustment;
// don't play streaming sounds backwards
if (_streaming && speed < 0) speed = 0;
Flt realPos = 0;
if (_streaming) {
realPos = pos;
while (pos >= STREAM_BUFFERSIZE) {
pos -= STREAM_BUFFERSIZE;
}
realPos -= pos;
}
Flt ** ptr2;
if (!_audioBuffer) ptr2 = _buffer.getArrayOfChannels();
// this is for a smooth fade-in to avoid glitches
if (intent == SS_WANTSTOPLAY) {
intent = SS_PLAYING;
volume = 0.f;
}
Flt startPos = pos;
YSE::SOUND_STATUS startIntent = intent;
Flt startVolume = volume;
// Now start filling the channels.
// Most sounds will probably only have one channel. At any rate, it is
// assumed that soundFile and soundImplementation have the same amount
// of channels.
FOREACH (filebuffer) {
// these are to reset to starting point in case of multi channel sounds
pos = startPos;
volume = startVolume;
intent = startIntent;
// this assumes the output and file have the same number of channels
Flt * out = filebuffer[i].getBuffer();
Flt * in;
if (!_audioBuffer) {
in = ptr2[i];
}
else { // only for audioBuffer sources
in = _audioBuffer->getChannel(i).getBuffer();
_length = _audioBuffer->getLength();
}
UInt l = length;
startAgain:
// evaluate intent and see what course to take
while (l > 0) {
// if intent is stopped or paused, we can just fill the rest of the
// buffer with zeroes. Normally, stopped or paused sounds are not
// even processed, but we get here when a sounds starts as 'wants_to_pause',
// fades out to zero and still has to fill the rest of the buffer.
if (intent == SS_STOPPED || intent == SS_PAUSED) {
// fill the rest with zero's
for (; l > 7; l -= 8, out += 8) {
out[0] = 0.f; out[1] = 0.f;
out[2] = 0.f; out[3] = 0.f;
out[4] = 0.f; out[5] = 0.f;
out[6] = 0.f; out[7] = 0.f;
}
while (l--) *out++ = 0.f;
goto nextBuffer; // this output channel buffer is full if we get here
}
// Most of the time a sound will just play at full volume. This has nothing
// to do with the sound volume itself (which will be applied afterwards) but
// with sounds fading in and out at start, stop and pause (to avoid glitches).
// Therefore, this is the most important part to optimize for speed.
else if (intent == SS_PLAYING_FULL_VOLUME) {
mainLoop:
// if playing backwards and we're past the beginning of the soundFile in
// less than 16 steps, move one by one
if ((speed < 0) && ((pos + speed * 16) < 0)) {
while (l--) {
*out++ = in[(UInt)pos];
pos += speed;
// check if we're past the beginning and readjust position if so
if (pos < 0) goto calibrate;
}
goto nextBuffer; // this output channel buffer is full if we get here
}
// if playing forward and we're past the end of the soundFile in
// less than 16 steps, move one by one
else if ((speed > 0) && ((pos + speed * 16) >= (_streaming ? STREAM_BUFFERSIZE : _length))) {
while (l--) {
*out++ = in[(UInt)pos];
pos += speed;
// check if we're past the end and readjust position if so
if (pos >= (_streaming ? STREAM_BUFFERSIZE : _length)) goto calibrate;
}
goto nextBuffer; // this output channel buffer is full if we get here
}
// at this point, we can be sure our current position allows for
// 16 more frames without recalibration
else {
// if the output still needs more than 16 frames, add 16 at once
if (l > 15) {
out[0] = in[(UInt)pos]; pos += speed;
out[1] = in[(UInt)pos]; pos += speed;
out[2] = in[(UInt)pos]; pos += speed;
out[3] = in[(UInt)pos]; pos += speed;
out[4] = in[(UInt)pos]; pos += speed;
out[5] = in[(UInt)pos]; pos += speed;
out[6] = in[(UInt)pos]; pos += speed;
out[7] = in[(UInt)pos]; pos += speed;
out[8] = in[(UInt)pos]; pos += speed;
out[9] = in[(UInt)pos]; pos += speed;
out[10] = in[(UInt)pos]; pos += speed;
out[11] = in[(UInt)pos]; pos += speed;
out[12] = in[(UInt)pos]; pos += speed;
out[13] = in[(UInt)pos]; pos += speed;
out[14] = in[(UInt)pos]; pos += speed;
out[15] = in[(UInt)pos]; pos += speed;
l -= 16;
out += 16;
// check if we're past the end and readjust position if so
if (pos < 0 || pos >= (_streaming ? STREAM_BUFFERSIZE : _length)) goto calibrate;
// since l > 15, the output buffer is not full yet
goto mainLoop;
}
// almost at the end of the output buffer, handle one
// frame at a time now
else {
while (l-- > 0) {
*out++ = in[(UInt)pos];
pos += speed;
// check if we're past the end and readjust position if so
if (pos < 0 || pos >= (_streaming ? STREAM_BUFFERSIZE : _length)) goto calibrate;
}
}
goto nextBuffer; // this output channel buffer is full if we get here
}
}
// Sound plays, but not at full volume. So we're fading in
else if (intent == SS_PLAYING) {
while (l--) {
*out++ = in[(UInt)pos] * volume;
pos += speed;
volume += 0.005f;
if (pos < 0 || pos >= (_streaming ? STREAM_BUFFERSIZE : _length)) goto calibrate;
if ((volume >= 1.f)) {
// full volume is reached. Move to the optimized version
volume = 1.f;
intent = SS_PLAYING_FULL_VOLUME;
goto startAgain;
}
}
goto nextBuffer; // this output channel buffer is full if we get here
}
// Still playing, but fading out to pause
else if (intent == SS_WANTSTOPAUSE) {
while (l--) {
*out++ = in[(UInt)pos] * volume;
pos += speed;
volume -= 0.005f;
if (pos < 0 || pos >= (_streaming ? STREAM_BUFFERSIZE : _length)) goto calibrate;
if ((volume <= 0.f)) {
// fade out complete, switch intent to paused and restart
// The rest of the buffer will be filled with zeroes
volume = 0.f;
intent = SS_PAUSED;
goto startAgain;
}
}
goto nextBuffer; // this output channel buffer is full if we get here
}
// Still playing, but fading out to stop
else if (intent == SS_WANTSTOSTOP) {
while (l--) {
*out++ = in[(UInt)pos] * volume;
pos += speed;
volume -= 0.005f;
if (pos < 0 || pos >= (_streaming ? STREAM_BUFFERSIZE : _length)) goto calibrate;
if ((volume <= 0.f)) {
// fade out complete, switch intent to stopped and restart
// The rest of the buffer will be filled with zeroes
volume = 0.f;
intent = SS_STOPPED;
goto startAgain;
}
}
goto nextBuffer; // this output channel buffer is full if we get here
}
calibrate:
if (_streaming) {
while (pos >= STREAM_BUFFERSIZE) {
if(fillStream(loop)) {
pos -= STREAM_BUFFERSIZE;
realPos += STREAM_BUFFERSIZE;
if (realPos >= streamReader->lengthInSamples) {
realPos -= streamReader->lengthInSamples;
}
}
else {
pos = 0;
intent = SS_STOPPED;
volume = 0.f;
_streamPos = 0;
continue;
}
}
}
else
{
// if we get here, pos is past the end or before the beginning
// recalibrate position now. We can't simply set it to the end
// or the beginning because this won't work with speed <> 1
while (pos < 0) pos += _length; // looping backwards
if (pos >= _length) {
if (loop) while (pos >= _length) pos -= _length;
else {
// if no loop, fill the rest of the buffer with zero's
pos = 0;
intent = SS_STOPPED;
}
}
}
}
// This label creates a safe point to start processing the next channel when the
// current one is done
nextBuffer: ;
}
if (_streaming) pos += realPos;
// make sure position is reset to zero if playing has stopped during this read
if (intent == SS_STOPPED) {
pos = 0;
if (_streaming) {
_streamPos = 0;
if (fillStream(loop)) {
pos -= STREAM_BUFFERSIZE;
realPos += STREAM_BUFFERSIZE;
if (realPos >= streamReader->lengthInSamples) {
realPos -= streamReader->lengthInSamples;
}
}
pos += realPos;
}
}
return true;
}