//------------------------------------------------------------
bool ofxSoundFile::mpg123ReadFile(ofSoundBuffer & buffer){
size_t done=0;
size_t block_size = mpg123_outblock( mp3File );
int err;
if(buffer.size()==0){
buffer.resize(block_size);
do{
err = mpg123_read(mp3File,(unsigned char*)&buffer[buffer.size()-block_size],block_size*4,&done);
buffer.resize(buffer.size()+block_size);
}while(err==MPG123_OK);
buffer.resize(buffer.size()-(block_size-done/4));
if(err != MPG123_DONE){
ofLogError() << "Warning: Decoding ended prematurely because: " << (err == MPG123_ERR ? mpg123_strerror(mp3File) : mpg123_plain_strerror(err));
return false;
}
duration = float(buffer.size())/float(channels) / float(samplerate);
}else{
err = mpg123_read(mp3File,(unsigned char*)&buffer[0],buffer.size()*sizeof(float),&done);
if(err != MPG123_OK){
ofLogError() << "Warning: Error decoding mp3: " << (err == MPG123_ERR ? mpg123_strerror(mp3File) : mpg123_plain_strerror(err));
return false;
}
}
return true;
}
//--------------------------------------------------------------
bool ofxSoundFile::save(string path, const ofSoundBuffer &buff){
// check that we're writing a wav and complain if the file extension is wrong.
ofFile f(path);
if(ofToLower(f.getExtension())!="wav") {
path += ".wav";
ofLogWarning() << "Can only write wav files - will save file as " << path;
}
fstream file(ofToDataPath(path).c_str(), ios::out | ios::binary);
if(!file.is_open()) {
ofLogError() << "Error opening sound file '" << path << "' for writing";
return false;
}
// write a wav header
short myFormat = 1; // for pcm
int mySubChunk1Size = 16;
int bitsPerSample = 16; // assume 16 bit pcm
int myByteRate = buff.getSampleRate() * buff.getNumChannels() * bitsPerSample/8;
short myBlockAlign = buff.getNumChannels() * bitsPerSample/8;
int myChunkSize = 36 + buff.size()*bitsPerSample/8;
int myDataSize = buff.size()*bitsPerSample/8;
int channels = buff.getNumChannels();
int samplerate = buff.getSampleRate();
file.seekp (0, ios::beg);
file.write ("RIFF", 4);
file.write ((char*) &myChunkSize, 4);
file.write ("WAVE", 4);
file.write ("fmt ", 4);
file.write ((char*) &mySubChunk1Size, 4);
file.write ((char*) &myFormat, 2); // should be 1 for PCM
file.write ((char*) &channels, 2); // # channels (1 or 2)
file.write ((char*) &samplerate, 4); // 44100
file.write ((char*) &myByteRate, 4); //
file.write ((char*) &myBlockAlign, 2);
file.write ((char*) &bitsPerSample, 2); //16
file.write ("data", 4);
file.write ((char*) &myDataSize, 4);
// write the wav file per the wav file format, 4096 bytes of data at a time.
#define WRITE_BUFF_SIZE 4096
short writeBuff[WRITE_BUFF_SIZE];
int pos = 0;
while(pos<buff.size()) {
int len = MIN(WRITE_BUFF_SIZE, buff.size()-pos);
for(int i = 0; i < len; i++) {
writeBuff[i] = (int)(buff[pos]*32767.f);
pos++;
}
file.write((char*)writeBuff, len*bitsPerSample/8);
}
file.close();
return true;
}
bool ofxSoundFile::sfReadFile(ofSoundBuffer & buffer){
samples_read = sf_read_float (sndFile, &buffer[0], buffer.size());
/*if(samples_read<(int)buffer.size()){
ofLogError() << "ofxSoundFile: couldnt read " << path;
return false;
}*/
if (subformat == SF_FORMAT_FLOAT || subformat == SF_FORMAT_DOUBLE){
for (int i = 0 ; i < int(buffer.size()) ; i++){
buffer[i] *= scale ;
}
}
return true;
}
//--------------------------------------------------------------
void ofApp::audioOut(ofSoundBuffer & buffer){
if(spectrum->playing){
// float *ptr = output;
for (int i = 0; i < buffer.getNumFrames(); i++){
wave = 0.0;
for(int n=0; n<BIT; n++){
if (amp[n]>0.00001) {
phases[n] += 512./(44100.0/(hertzScale[n]));
if ( phases[n] >= 511 ) phases[n] -=512;
if ( phases[n] < 0 ) phases[n] = 0;
//remainder = phases[n] - floor(phases[n]);
//wave+=(float) ((1-remainder) * sineBuffer[1+ (long) phases[n]] + remainder * sineBuffer[2+(long) phases[n]])*amp[n];
wave += ( sineBuffer[1 + (long) phases[n]] ) * amp[n];
}
}
wave/=10.0;
if(wave>1.0) wave=1.0;
if(wave<-1.0) wave=-1.0;
buffer[i * buffer.getNumChannels() ] = wave * volume;
buffer[i * buffer.getNumChannels() + 1] = wave * volume;
// outp[i] = wave;
// *ptr++ = wave * volume;
}
} else {
// for (int i = 0; i < bufferSize; i++){
// output[i*nChannels ] = 0;
// output[i*nChannels + 1] = 0;
// outp[i] = 0;
// }
}
}
void ofApp::audioOut(ofSoundBuffer& buffer){
for (unsigned i = 0 ; i<bufferSize; i++) {
currentSample = (osc.square(frequency))/2;
buffer[i * buffer.getNumChannels()] = currentSample;
buffer[i * buffer.getNumChannels()+1] = currentSample;
}
}
void WaveStripe::newSoundBuffer(ofSoundBuffer & buffer){
float ampL = buffer.getRMSAmplitudeChannel(0);
float ampR = buffer.getRMSAmplitudeChannel(1);
mutex.lock();
amplitudesLeftBack.push_back(ampL);
amplitudesRightBack.push_back(ampR);
mutex.unlock();
}
void ofApp::audioOut(ofSoundBuffer &buffer) {
for (unsigned i = 0; i<bufferSize; i++) {
currentSample = sample.playOnce(1.0);
mix.stereo(currentSample, outputs, 0.5);
buffer[i*buffer.getNumChannels()] = outputs[0];
buffer[i*buffer.getNumChannels()] = outputs[1];
}
}
//--------------------------------------------------------------
void ofApp::audioIn(ofSoundBuffer & input){
sb = input;
for (int i = 0; i < input.getNumFrames(); i++){
left[i] = input[i*2];
right[i] = input[i*2+1];
mono[i] = (left[i] + right[i]) / 2;
}
}
void ofApp::audioOut(ofSoundBuffer &buffer){
for (unsigned i = 0; i<bufferSize; i++){
currentSample = osc.sinewave(freq + env.adsr(mod.sinewave(modFreq), env.trigger) * modIndex);
mix.stereo(currentSample, outputs, 0.5);
buffer[i*buffer.getNumChannels()] = outputs[0];
buffer[i*buffer.getNumChannels()+1] = outputs[1];
}
}
void ofApp::audioOut(ofSoundBuffer &buffer){
for (unsigned i = 0; i < bufferSize; i++){
currentSample = osc.saw(frequency) * modulator.sinewave(modSpeed);
// currentSample = osc.saw(frequency) * modulator.square(modSpeed);
mix.stereo(currentSample, outputs, 0.5);
buffer[i*buffer.getNumChannels()] = outputs[0];
buffer[i*buffer.getNumChannels() + 1] = outputs[1];
}
}
//--------------------------------------------------------------
void ofApp::audioOut(ofSoundBuffer &outBuffer) {
// base frequency of the lowest sine wave in cycles per second (hertz)
float frequency = 172.5;
// mapping frequencies from Hz into full oscillations of sin() (two pi)
float wavePhaseStep = (frequency / sampleRate) * TWO_PI;
float pulsePhaseStep = (0.5 / sampleRate) * TWO_PI;
// this loop builds a buffer of audio containing 3 sine waves at different
// frequencies, and pulses the volume of each sine wave individually. In
// other words, 3 oscillators and 3 LFOs.
for(int i = 0; i < outBuffer.getNumFrames(); i++) {
// build up a chord out of sine waves at 3 different frequencies
float sampleLow = sin(wavePhase);
float sampleMid = sin(wavePhase * 1.5);
float sampleHi = sin(wavePhase * 2.0);
// pulse each sample's volume
sampleLow *= sin(pulsePhase);
sampleMid *= sin(pulsePhase * 1.04);
sampleHi *= sin(pulsePhase * 1.09);
float fullSample = (sampleLow + sampleMid + sampleHi);
// reduce the full sample's volume so it doesn't exceed 1
fullSample *= 0.3;
// write the computed sample to the left and right channels
outBuffer.getSample(i, 0) = fullSample;
outBuffer.getSample(i, 1) = fullSample;
// get the two phase variables ready for the next sample
wavePhase += wavePhaseStep;
pulsePhase += pulsePhaseStep;
}
ofScopedLock lock(audioMutex);
lastBuffer = outBuffer;
}
void ofxBasicSoundPlayer::audioOut(ofSoundBuffer& outputBuffer){
if(bIsPlaying){
int nFrames = outputBuffer.getNumFrames();
int nChannels = outputBuffer.getNumChannels();
if (playerNumChannels != nChannels || playerNumFrames != nFrames || playerSampleRate != outputBuffer.getSampleRate()) {
audioOutBuffersChanged(nFrames, nChannels, outputBuffer.getSampleRate());
}
if(streaming){
int samplesRead = soundFile.readTo(buffer,nFrames);
if ( samplesRead==0 ){
bIsPlaying=false;
soundFile.seekTo(0);
}
else{
buffer.stereoPan(volumesLeft.back(),volumesRight.back());
newBufferE.notify(this,buffer);
buffer.copyTo(outputBuffer);
}
}else{
if (positions.size() == 1 && abs(speed - 1)<FLT_EPSILON) {
buffer.copyTo(outputBuffer,nFrames,nChannels,positions[0],loop);
}else{
for(int i=0;i<(int)positions.size();i++){
//assert( resampledBuffer.getNumFrames() == bufferSize*relativeSpeed[i] );
if(abs(relativeSpeed[i] - 1)<FLT_EPSILON){
buffer.copyTo(resampledBuffer,nFrames,nChannels,positions[i],loop);
}else{
buffer.resampleTo(resampledBuffer,positions[i],nFrames,relativeSpeed[i],loop, ofSoundBuffer::Linear);
}
resampledBuffer.stereoPan(volumesLeft[i],volumesRight[i]);
newBufferE.notify(this,resampledBuffer);
resampledBuffer.addTo(outputBuffer,0,loop);
}
}
updatePositions(nFrames);
}
}
}
//----------
void Focus::audioOut(ofSoundBuffer & out) {
int intervalFrames;
{
lock_guard<mutex> lock(this->resultMutex);
if(!this->result.active) {
return;
}
auto interval = 1.0f / pow (2.0f, this->result.valueNormalised / 0.12f);
intervalFrames = int(interval * 44100.0f);
}
auto & soundEngine = ofxRulr::Utils::SoundEngine::X();
auto & assetRegister = ofxAssets::Register::X();
auto tickBig = assetRegister.getSoundPointer("ofxRulr::tick_big");
auto tickSmall = assetRegister.getSoundPointer("ofxRulr::tick_small");
auto numFrames = out.getNumFrames();
for(int i=0; i<numFrames; i++) {
//check if this frame we start a tick
if(this->ticks.framesUntilNext <= 0) {
//select the tick sound
auto isBigTick = this->ticks.index++ == 0;
this->ticks.index %= 6;
auto tickSoundAsset = isBigTick ? tickBig : tickSmall;
//add it to the active sounds (delayed by 1 buffer always)
ofxRulr::Utils::SoundEngine::ActiveSound activeSound;
activeSound.delay = i;
activeSound.sound = tickSoundAsset;
soundEngine.play(activeSound);
//set the next tick sound
this->ticks.framesUntilNext = intervalFrames;
}
//check interval doesn't go too long
if(this->ticks.framesUntilNext > intervalFrames) {
//e.g. this might happen at next buffer fill
this->ticks.framesUntilNext = intervalFrames;
}
this->ticks.framesUntilNext--;
}
}
//--------------------------------------------------------------
bool ofxSoundFile::readTo(ofSoundBuffer & buffer, unsigned int _samples){
buffer.setNumChannels(channels);
buffer.setSampleRate(samplerate);
if(_samples!=0){
// will read the requested number of samples
// clamp to the number of samples we actually have
_samples = min(_samples,samples);
buffer.resize(_samples*channels);
}
#ifdef OF_USING_SNDFILE
else if (sndFile){
// will read entire file
buffer.resize(samples);
}
#endif
#ifdef OF_USING_LAD
else if (audioDecoder) {
// will read entire file
buffer.resize(samples);
}
#endif
#ifdef OF_USING_MPG123
else if(mp3File){
buffer.clear();
}
#endif
#ifdef OF_USING_SNDFILE
if(sndFile) return sfReadFile(buffer);
#elif defined( OF_USING_MPG123 )
if(mp3File) return mpg123ReadFile(buffer);
#elif defined( OF_USING_LAD )
if(audioDecoder) return ladReadFile(buffer);
#endif
return false;
}
void ofxGSTT::audioIn(ofSoundBuffer & buffer){
//TODO make better use of soundbuffer obj
audioIn(&buffer[0], buffer.getNumFrames(), buffer.getNumChannels(), OFXGSTT_DEFAULTDEVICE_ID); //TODO multidevice business
}
void ofBaseSoundOutput::audioOut( ofSoundBuffer& buffer ){
audioOut(&buffer[0], buffer.getNumFrames(), buffer.getNumChannels(), buffer.getDeviceID(), buffer.getTickCount());
}
bool ofxSoundFile::ladReadFile(ofSoundBuffer &buffer){
int samplesRead = audioDecoder->read( buffer.size(), &buffer[0] );
return samplesRead;
}
