Soundship/Loopp
1
0
Fork 0
Code Issues Pull requests Releases Wiki Activity

Use stereo input and stereo audio buffer

Browse source
This commit is contained in:
coderkun 2017-03-19 12:45:04 +01:00 committed by Harry van Haaren
parent 6e8687592b
commit 59b146f052
11 changed files with 107 additions and 53 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

28
src/audiobuffer.cxx
View file

@ -17,6 +17,7 @@
*/
#include "audiobuffer.hxx"
#include "config.hxx"
#include <stdio.h>
@ -31,7 +32,8 @@ AudioBuffer::AudioBuffer(unsigned long size)
// FIXME recorded buffers don't get an ID, using garbage IDs
/// no ID assigned: it *takes* the one from the previous buffer!
init();
buffer.resize(size);
bufferL.resize(size);
bufferR.resize(size);
}
void AudioBuffer::init()
@ -98,18 +100,32 @@ long AudioBuffer::getAudioFrames()
return audioFrames;
}
std::vector<float>& AudioBuffer::getData()
long AudioBuffer::getSize()
{
return buffer;
if(bufferL.size() != bufferR.size()) {
LUPPP_WARN("left and right channels of audio buffer have different size: %i vs %i", bufferL.size(), bufferR.size() );
}
return std::min(bufferL.size(), bufferR.size());
}
void AudioBuffer::nonRtSetSample(std::vector<float>& sample)
std::vector<float>& AudioBuffer::getDataL()
{
buffer.swap(sample);
return bufferL;
}
std::vector<float>& AudioBuffer::getDataR()
{
return bufferR;
}
void AudioBuffer::nonRtSetSample(std::vector<float>& sampleL, std::vector<float>& sampleR)
{
bufferL.swap(sampleL);
bufferR.swap(sampleR);
}
void AudioBuffer::nonRtResize(unsigned long size)
{
buffer.resize(size);
bufferL.resize(size);
bufferR.resize(size);
}
/*

10
src/audiobuffer.hxx
View file

@ -52,9 +52,12 @@ public:
long getAudioFrames();
std::vector<float>& getData();
long getSize();
void nonRtSetSample(std::vector<float>& sample);
std::vector<float>& getDataL();
std::vector<float>& getDataR();
void nonRtSetSample(std::vector<float>& sampleL, std::vector<float>& sampleR);
void nonRtResize(unsigned long size);
@ -72,7 +75,8 @@ protected:
char name[20];
std::vector<float> buffer;
std::vector<float> bufferL;
std::vector<float> bufferR;
};
#endif

3
src/buffers.hxx
View file

@ -28,7 +28,8 @@ public:
enum BUFFER {
// AUDIO
MASTER_INPUT = 0,
MASTER_INPUT_L = 0,
MASTER_INPUT_R,
MASTER_OUT_L,
MASTER_OUT_R,

3
src/diskreader.cxx
View file

@ -237,7 +237,8 @@ int DiskReader::loadSample( int track, int scene, string path )
/// create buffer, and set the data
AudioBuffer* ab = new AudioBuffer();
ab->setAudioFrames( buf.size() );
ab->nonRtSetSample( buf );
// TODO right (stereo)
ab->nonRtSetSample( buf, buf );
//cout << "DiskReader::loadSample() " << path << endl;

3
src/diskwriter.cxx
View file

@ -321,7 +321,8 @@ int DiskWriter::writeAudioBuffer(int track, int scene, AudioBuffer* ab,
// FIXME: the size of the buffer is bigger than the audio contained in it:
// calculate the length that needs saving using getBeats() * framesPerBeat
if ( ab->getAudioFrames() > 0 )
outfile.write( &ab->getData()[0], ab->getAudioFrames() );
// TODO right (stereo)
outfile.write( &ab->getDataL()[0], ab->getAudioFrames() );
else {
LUPPP_WARN("%s","Sample has zero samples");
}

3
src/gaudioeditor.cxx
View file

@ -102,7 +102,8 @@ void AudioEditor::show( AudioBuffer* buf, bool modal )
if ( !ab ) {
LUPPP_WARN("called with ab == 0");
} else {
std::vector<float>& tmp = ab->getData();
// TODO right (stereo)
std::vector<float>& tmp = ab->getDataL();
int size = tmp.size();
waveform->setData( &tmp[0], size );

43
src/jack.cxx
View file

@ -109,11 +109,17 @@ Jack::Jack( std::string name ) :
uiUpdateCounter = buffers.samplerate / 30;
uiUpdateConstant = buffers.samplerate / 30;
masterInput = jack_port_register( client,
"master_in",
JACK_DEFAULT_AUDIO_TYPE,
JackPortIsInput,
0 );
masterInputL = jack_port_register( client,
"master_in_left",
JACK_DEFAULT_AUDIO_TYPE,
JackPortIsInput,
0 );
masterInputR = jack_port_register( client,
"master_in_right",
JACK_DEFAULT_AUDIO_TYPE,
JackPortIsInput,
0 );
masterReturnL = jack_port_register( client,
"master_return_left",
@ -367,7 +373,8 @@ void Jack::unregisterMidiIO( MidiIO* mo )
int Jack::process (jack_nframes_t nframes)
{
/// get buffers
buffers.audio[Buffers::MASTER_INPUT] = (float*)jack_port_get_buffer( masterInput , nframes );
buffers.audio[Buffers::MASTER_INPUT_L] = (float*)jack_port_get_buffer( masterInputL , nframes );
buffers.audio[Buffers::MASTER_INPUT_R] = (float*)jack_port_get_buffer( masterInputR , nframes );
buffers.audio[Buffers::MASTER_RETURN_L] = (float*)jack_port_get_buffer( masterReturnL , nframes );
buffers.audio[Buffers::MASTER_RETURN_R] = (float*)jack_port_get_buffer( masterReturnR , nframes );
buffers.audio[Buffers::HEADPHONES_OUT] = (float*)jack_port_get_buffer( headphonesPort , nframes );
@ -469,7 +476,8 @@ void Jack::processFrames(int nframes)
/// mix input, reverb & post-sidechain in
for(unsigned int i = 0; i < nframes; i++) {
float input= buffers.audio[Buffers::MASTER_INPUT][i] * inputVol;
float inputL = buffers.audio[Buffers::MASTER_INPUT_L][i] * inputVol;
float inputR = buffers.audio[Buffers::MASTER_INPUT_R][i] * inputVol;
float L = buffers.audio[Buffers::MASTER_OUT_L][i];
float R = buffers.audio[Buffers::MASTER_OUT_R][i];
@ -478,19 +486,23 @@ void Jack::processFrames(int nframes)
if ( inputToMixEnable ) {
// if sending to mix, scale by volume *and* by XSide send
float tmp = input * inputToMixVol * (1-inputToXSideVol);
L += tmp;
R += tmp;
float tmpL = inputL * inputToMixVol * (1-inputToXSideVol);
float tmpR = inputR * inputToMixVol * (1-inputToXSideVol);
L += tmpL;
R += tmpR;
}
if ( inputToSendEnable ) {
// post-mix-send amount: hence * inputToMixVol
buffers.audio[Buffers::SEND][i] += input * inputToSendVol * inputToMixVol;
// TODO input r (stereo send)
buffers.audio[Buffers::SEND][i] += inputL * inputToSendVol * inputToMixVol;
}
if ( inputToKeyEnable ) {
buffers.audio[Buffers::SIDECHAIN_KEY][i] += input;
// TODO input r (stereo key)
buffers.audio[Buffers::SIDECHAIN_KEY][i] += inputL;
}
buffers.audio[Buffers::SIDECHAIN_SIGNAL][i] += input * inputToXSideVol;
// TODO input r (stereo sidechain)
buffers.audio[Buffers::SIDECHAIN_SIGNAL][i] += inputL * inputToXSideVol;
//compute master volume lag;
if(fabs(masterVol-masterVolLag)>=fabs(masterVolDiff/10.0))
@ -507,7 +519,7 @@ void Jack::processFrames(int nframes)
/// db meter on master input & output
inputMeter->process( nframes, buffers.audio[Buffers::MASTER_INPUT], buffers.audio[Buffers::MASTER_INPUT]);
inputMeter->process( nframes, buffers.audio[Buffers::MASTER_INPUT_L], buffers.audio[Buffers::MASTER_INPUT_R]);
masterMeter->process(nframes, buffers.audio[Buffers::JACK_MASTER_OUT_L], buffers.audio[Buffers::JACK_MASTER_OUT_R] );
if ( uiUpdateCounter > uiUpdateConstant ) {
@ -541,7 +553,8 @@ void Jack::processFrames(int nframes)
/// update buffers by nframes
if(lastnframes+nframes<buffers.nframes) {
lastnframes=nframes;
buffers.audio[Buffers::MASTER_INPUT] = &buffers.audio[Buffers::MASTER_INPUT] [nframes];
buffers.audio[Buffers::MASTER_INPUT_L] = &buffers.audio[Buffers::MASTER_INPUT_L][nframes];
buffers.audio[Buffers::MASTER_INPUT_R] = &buffers.audio[Buffers::MASTER_INPUT_R][nframes];
buffers.audio[Buffers::MASTER_RETURN_L] = &buffers.audio[Buffers::MASTER_RETURN_L][nframes];
buffers.audio[Buffers::MASTER_RETURN_R] = &buffers.audio[Buffers::MASTER_RETURN_R][nframes];
buffers.audio[Buffers::HEADPHONES_OUT] = &buffers.audio[Buffers::HEADPHONES_OUT] [nframes];

3
src/jack.hxx
View file

@ -173,7 +173,8 @@ private:
bool clientActive;
int samplerate;
jack_port_t* masterInput;
jack_port_t* masterInputL;
jack_port_t* masterInputR;
jack_port_t* masterOutputL;
jack_port_t* masterOutputR;

18
src/looper.cxx
View file

@ -118,8 +118,9 @@ void Looper::process(unsigned int nframes, Buffers* buffers)
}
// copy data from input buffer to recording buffer
float* input = buffers->audio[Buffers::MASTER_INPUT];
clips[clip]->record( nframes, input, 0 );
float* inputL = buffers->audio[Buffers::MASTER_INPUT_L];
float* inputR = buffers->audio[Buffers::MASTER_INPUT_R];
clips[clip]->record( nframes, inputL, inputR);
} else if ( clips[clip]->playing() ) {
// copy data into tmpBuffer, then pitch-stretch into track buffer
long targetFrames = clips[clip]->getBeats() * fpb;
@ -130,11 +131,14 @@ void Looper::process(unsigned int nframes, Buffers* buffers)
playSpeed = float(actualFrames) / targetFrames;
}
// TODO stereo sends
float* out = buffers->audio[Buffers::SEND_TRACK_0 + track];
for(unsigned int i = 0; i < nframes; i++ ) {
// REFACTOR into system that is better than per sample function calls
float tmp = clips[clip]->getSample(playSpeed);
float tmpL = 0;
float tmpR = 0;
clips[clip]->getSample(playSpeed, &tmpL, &tmpR);
float deltaPitch = 12 * log ( playSpeed ) / log (2);
semitoneShift = -deltaPitch;
@ -143,9 +147,13 @@ void Looper::process(unsigned int nframes, Buffers* buffers)
//FIXME: pitchShift adds delay even for playSpeed = 1.0!!
//we should use something better (e.g librubberband)
if(playSpeed!=1.0f)
pitchShift( 1, &tmp, &out[i] );
pitchShift( 1, &tmpL, &out[i] );
// TODO stereo sends
//pitchShift( 1, tmpR, &out[i] );
else
out[i]+=tmp;
// TODO stereo sends
out[i]+=tmpL;
//outR[i]+=tmpR;
}
//printf("Looper %i playing(), speed = %f\n", track, playSpeed );

44
src/looperclip.cxx
View file

@ -121,7 +121,7 @@ void LooperClip::load( AudioBuffer* ab )
jack->getControllerUpdater()->setTrackSceneProgress(track, scene, 0 );
// set the endpoint to the buffer's size
_recordhead = _buffer->getData().size();
_recordhead = _buffer->getSize();
#ifdef DEBUG_BUFFER
char buffer [50];
@ -134,8 +134,9 @@ void LooperClip::load( AudioBuffer* ab )
void LooperClip::setRequestedBuffer( AudioBuffer* ab )
{
if ( _buffer ) {
size_t size = _buffer->getData().size();
memcpy( &ab->getData().at(0), &_buffer->getData().at(0), sizeof(float)*size);
size_t size = _buffer->getSize();
memcpy( &ab->getDataL().at(0), &_buffer->getDataL().at(0), sizeof(float)*size);
memcpy( &ab->getDataR().at(0), &_buffer->getDataR().at(0), sizeof(float)*size);
ab->setID ( _buffer->getID() );
ab->setBeats( _buffer->getBeats() );
@ -153,10 +154,13 @@ void LooperClip::setRequestedBuffer( AudioBuffer* ab )
void LooperClip::recieveSaveBuffer( AudioBuffer* saveBuffer )
{
if ( saveBuffer->getData().size() >= _buffer->getData().at(0) ) {
if ( saveBuffer->getSize() >= _buffer->getDataL().at(0) ||
saveBuffer->getSize() >= _buffer->getDataR().at(0) ) {
// copy current contents into save buffer,
// getData() contains L and R buffer, so twice the size is needed
size_t framesBySize = _buffer->getAudioFrames();
memcpy( &saveBuffer->getData().at(0), &_buffer->getData().at(0), sizeof(float)*framesBySize);
memcpy( &saveBuffer->getDataL().at(0), &_buffer->getDataL().at(0), sizeof(float)*framesBySize);
memcpy( &saveBuffer->getDataR().at(0), &_buffer->getDataR().at(0), sizeof(float)*framesBySize);
saveBuffer->setID ( _buffer->getID() );
saveBuffer->setBeats( _buffer->getBeats() );
@ -189,11 +193,12 @@ void LooperClip::record(int count, float* L, float* R)
{
// write "count" samples into current buffer.
if ( _buffer ) {
size_t size = _buffer->getData().size();
size_t size = _buffer->getSize();
for(int i = 0; i < count; i++) {
if ( _recordhead < size ) {
_buffer->getData().at( _recordhead ) = *L++;
_buffer->getDataL().at( _recordhead ) = *L++;
_buffer->getDataR().at( _recordhead ) = *R++;
_recordhead++;
} else {
// break: this is *BAD*, audio data is lost but the buffer isn't here
@ -218,7 +223,8 @@ void LooperClip::record(int count, float* L, float* R)
unsigned long LooperClip::recordSpaceAvailable()
{
if ( _buffer )
return _buffer->getData().size() - _recordhead;
// getData() contains L and R buffer, so it is twice the size
return _buffer->getSize() - _recordhead;
return 0;
}
@ -226,7 +232,7 @@ unsigned long LooperClip::recordSpaceAvailable()
size_t LooperClip::audioBufferSize()
{
if ( _buffer ) {
return _buffer->getData().size();
return _buffer->getSize();
}
return 0;
}
@ -410,11 +416,11 @@ bool LooperClip::newBufferInTransit()
return _newBufferInTransit;
}
float LooperClip::getSample(float playSpeed)
void LooperClip::getSample(float playSpeed, float* L, float* R)
{
if ( _buffer && _buffer->getData().size() > 0 ) {
if ( _buffer && (_buffer->getSize() > 0)) {
if ( _playhead >= _recordhead ||
_playhead >= _buffer->getData().size() ||
_playhead >= _buffer->getSize() ||
_playhead < 0 ) {
_playhead = 0;
@ -422,14 +428,16 @@ float LooperClip::getSample(float playSpeed)
//writeToGuiRingbuffer( &e );
}
std::vector<float>& v = _buffer->getData();
float tmp = v.at(_playhead);
std::vector<float>& vL = _buffer->getDataL();
std::vector<float>& vR = _buffer->getDataR();
*L = vL.at(_playhead);
*R = vR.at(_playhead);
_playhead +=playSpeed;
return tmp;
}
return 0.f;
else {
*L = 0.f;
*R = 0.f;
}
}
float LooperClip::getProgress()

2
src/looperclip.hxx
View file

@ -55,7 +55,7 @@ public:
void load( AudioBuffer* ab );
/// audio functionality
float getSample(float playSpeed);
void getSample(float playSpeed, float* L, float* R);
void record(int count, float* L, float* R);
/// TimeObserver override