void collectLiveEvents( BaseInstrument* instrument )
{
AudioChannel* channel = instrument->audioChannel;
std::vector<BaseAudioEvent*>* liveEvents = instrument->getLiveEvents();
// removal queue
std::vector<BaseAudioEvent*> removes;
int i = 0;
for ( i; i < liveEvents->size(); i++ )
{
BaseAudioEvent* audioEvent = liveEvents->at( i );
if ( !audioEvent->isDeletable())
channel->addLiveEvent( audioEvent );
else
removes.push_back( audioEvent );
}
// removal queue filled ? process it so we can safely
// remove "deleted" AudioEvents without errors occurring
if ( removes.size() > 0 )
{
int i = 0;
for ( i; i < removes.size(); i++ )
{
BaseAudioEvent* audioEvent = removes[ i ];
instrument->removeEvent( audioEvent, true );
}
}
}
void ChannelMergerNode::process(size_t framesToProcess)
{
AudioNodeOutput* output = this->output(0);
ASSERT(output);
ASSERT_UNUSED(framesToProcess, framesToProcess == output->bus()->length());
// Output bus not updated yet, so just output silence.
if (m_desiredNumberOfOutputChannels != output->numberOfChannels()) {
output->bus()->zero();
return;
}
// Merge all the channels from all the inputs into one output.
unsigned outputChannelIndex = 0;
for (unsigned i = 0; i < numberOfInputs(); ++i) {
AudioNodeInput* input = this->input(i);
if (input->isConnected()) {
unsigned numberOfInputChannels = input->bus()->numberOfChannels();
// Merge channels from this particular input.
for (unsigned j = 0; j < numberOfInputChannels; ++j) {
AudioChannel* inputChannel = input->bus()->channel(j);
AudioChannel* outputChannel = output->bus()->channel(outputChannelIndex);
outputChannel->copyFrom(inputChannel);
++outputChannelIndex;
}
}
}
ASSERT(outputChannelIndex == output->numberOfChannels());
}
void ChannelMergerHandler::process(size_t framesToProcess) {
AudioNodeOutput& output = this->output(0);
DCHECK_EQ(framesToProcess, output.bus()->length());
unsigned numberOfOutputChannels = output.numberOfChannels();
DCHECK_EQ(numberOfInputs(), numberOfOutputChannels);
// Merge multiple inputs into one output.
for (unsigned i = 0; i < numberOfOutputChannels; ++i) {
AudioNodeInput& input = this->input(i);
DCHECK_EQ(input.numberOfChannels(), 1u);
AudioChannel* outputChannel = output.bus()->channel(i);
if (input.isConnected()) {
// The mixing rules will be applied so multiple channels are down-
// mixed to mono (when the mixing rule is defined). Note that only
// the first channel will be taken for the undefined input channel
// layout.
//
// See:
// http://webaudio.github.io/web-audio-api/#channel-up-mixing-and-down-mixing
AudioChannel* inputChannel = input.bus()->channel(0);
outputChannel->copyFrom(inputChannel);
} else {
// If input is unconnected, fill zeros in the channel.
outputChannel->zero();
}
}
}
void EqualPowerPanner::pan(double azimuth, double /*elevation*/, AudioBus* inputBus, AudioBus* outputBus, size_t framesToProcess)
{
// FIXME: implement stereo sources
bool isInputSafe = inputBus && inputBus->numberOfChannels() == 1 && framesToProcess <= inputBus->length();
ASSERT(isInputSafe);
if (!isInputSafe)
return;
bool isOutputSafe = outputBus && outputBus->numberOfChannels() == 2 && framesToProcess <= outputBus->length();
ASSERT(isOutputSafe);
if (!isOutputSafe)
return;
AudioChannel* channel = inputBus->channel(0);
float* sourceP = channel->data();
float* destinationL = outputBus->channelByType(AudioBus::ChannelLeft)->data();
float* destinationR = outputBus->channelByType(AudioBus::ChannelRight)->data();
if (!sourceP || !destinationL || !destinationR)
return;
// Pan smoothly from left to right with azimuth going from -30 -> +30 degrees.
double desiredPanPosition;
if (azimuth > 30.0)
desiredPanPosition = 1.0;
else if (azimuth < -30.0)
desiredPanPosition = 0.0;
else
desiredPanPosition = (azimuth + 30.0) / 60.0;
double desiredGainL = 0.5 * cos(piDouble * desiredPanPosition) + 0.5;
double desiredGainR = sqrt(1.0 - desiredGainL*desiredGainL);
// Don't de-zipper on first render call.
if (m_isFirstRender) {
m_isFirstRender = false;
m_gainL = desiredGainL;
m_gainR = desiredGainR;
}
// Cache in local variables.
double gainL = m_gainL;
double gainR = m_gainR;
// Get local copy of smoothing constant.
const double SmoothingConstant = m_smoothingConstant;
int n = framesToProcess;
while (n--) {
float input = *sourceP++;
gainL += (desiredGainL - gainL) * SmoothingConstant;
gainR += (desiredGainR - gainR) * SmoothingConstant;
*destinationL++ = static_cast<float>(input * gainL);
*destinationR++ = static_cast<float>(input * gainR);
}
m_gainL = gainL;
m_gainR = gainR;
}
std::vector<AudioChannel*> getAudioEvents( std::vector<AudioChannel*> channels, int bufferPosition,
int bufferEnd, bool addLiveInstruments )
{
// clear previous channel contents (note we don't delete the channels anywhere as we re-use them)
channels.clear();
int i, l;
// note we update the channels mix properties here as they might change during playback
for ( i = 0, l = instruments.size(); i < l; ++i )
{
BaseInstrument* instrument = instruments.at( i );
AudioChannel* instrumentChannel = instrument->audioChannel;
instrumentChannel->reset();
instrumentChannel->mixVolume = instrument->volume;
if ( !instrumentChannel->muted )
{
if ( AudioEngine::playing )
collectSequencedEvents( instrument, bufferPosition, bufferEnd );
if ( addLiveInstruments && instrument->hasLiveEvents() )
collectLiveEvents( instrument );
channels.push_back( instrumentChannel );
}
}
return channels;
}
void synthesize(float hz, float timbre, float volume)
{
LOG("hz=%f timbre=%f volume=%f\n", hz, timbre, volume);
whiteNoiseChannel.setVolume(volume * 96.f);
whiteNoiseChannel.setSpeed(hz * arraysize(whiteNoise));
}
bool Sequencer::getAudioEvents( std::vector<AudioChannel*>* channels, int bufferPosition,
int bufferSize, bool addLiveInstruments, bool flushChannels )
{
channels->clear();
int bufferEnd = bufferPosition + ( bufferSize - 1 ); // the highest SampleEnd value we'll query
bool loopStarted = bufferEnd > AudioEngine::max_buffer_position; // whether this request exceeds the min_buffer_position - max_buffer_position range
int i, l;
// note we update the channels mix properties here as they might change during playback
for ( i = 0, l = ( int ) instruments.size(); i < l; ++i )
{
BaseInstrument* instrument = instruments.at( i );
AudioChannel* instrumentChannel = instrument->audioChannel;
// clear previous channel contents when requested
if ( flushChannels )
instrumentChannel->reset();
if ( !instrumentChannel->muted )
{
if ( playing )
collectSequencedEvents( instrument, bufferPosition, bufferEnd );
if ( addLiveInstruments && instrument->hasLiveEvents() )
collectLiveEvents( instrument );
channels->push_back( instrumentChannel );
}
}
return loopStarted;
}
/**
* used by the getAudioEvents-method of the sequencer, this validates
* the present AudioEvents against the requested position
* and updates and flushes the removal queue
*
* @param instrument {BaseInstrument*} instrument to gather events from
* @param bufferPosition {int} the current buffers start pointer
* @param bufferEnd {int} the current buffers end pointer
*/
void collectSequencedEvents( BaseInstrument* instrument, int bufferPosition, int bufferEnd )
{
if ( !instrument->hasEvents() )
return;
AudioChannel* channel = instrument->audioChannel;
std::vector<BaseAudioEvent*>* audioEvents = instrument->getEvents();
// removal queue
std::vector<BaseAudioEvent*> removes;
// channel has an internal loop (e.g. drum machine) ? recalculate requested
// buffer position by subtracting all measures above the first
if ( channel->maxBufferPosition > 0 )
{
int samplesPerBar = AudioEngine::samples_per_bar;
while ( bufferPosition >= channel->maxBufferPosition )
{
bufferPosition -= samplesPerBar;
bufferEnd -= samplesPerBar;
}
}
int i = 0, amount = audioEvents->size();
for ( i; i < amount; i++ )
{
BaseAudioEvent* audioEvent = audioEvents->at( i );
if ( audioEvent->isEnabled() )
{
int sampleStart = audioEvent->getSampleStart();
int sampleEnd = audioEvent->getSampleEnd();
if ( audioEvent->isLoopeable() ||
( sampleStart >= bufferPosition && sampleStart <= bufferEnd ) ||
( sampleStart < bufferPosition && sampleEnd >= bufferPosition ))
{
if ( !audioEvent->isDeletable())
channel->addEvent( audioEvent );
else
removes.push_back( audioEvent );
}
}
}
// removal queue filled ? process it so we can safely
// remove "deleted" AudioEvents without errors occurring
if ( removes.size() > 0 )
{
int i = 0;
for ( i; i < removes.size(); i++ )
{
BaseAudioEvent* audioEvent = removes[ i ];
instrument->removeEvent( audioEvent, false );
}
}
}
// Pulls on our provider to get rendered audio stream.
OSStatus AudioDestinationMac::render(UInt32 numberOfFrames, AudioBufferList* ioData)
{
AudioBuffer* buffers = ioData->mBuffers;
m_renderBus.setChannelMemory(0, (float*)buffers[0].mData, numberOfFrames);
m_renderBus.setChannelMemory(1, (float*)buffers[1].mData, numberOfFrames);
//@tofix - add support for local/live audio input.
m_callback.render(m_input->m_audioBus, &m_renderBus, numberOfFrames);
// Clamp values at 0db (i.e., [-1.0, 1.0])
for (unsigned i = 0; i < m_renderBus.numberOfChannels(); ++i) {
AudioChannel* channel = m_renderBus.channel(i);
VectorMath::vclip(channel->data(), 1, &kLowThreshold, &kHighThreshold, channel->mutableData(), 1, numberOfFrames);
}
return noErr;
}
void synthInit()
{
Random gen = Random();
//Generate the White Noise
for (int i = 0; i != arraysize(whiteNoise); i++) {
whiteNoise[i] = gen.random() * 0x7fff;
}
whiteNoiseChannel.play(noiseAsset);
}
void ChannelMergerNode::process(size_t framesToProcess)
{
AudioNodeOutput* output = this->output(0);
ASSERT(output);
ASSERT_UNUSED(framesToProcess, framesToProcess == output->bus()->length());
// Output bus not updated yet, so just output silence.
if (m_desiredNumberOfOutputChannels != output->numberOfChannels()) {
output->bus()->zero();
return;
}
// Merge all the channels from all the inputs into one output.
unsigned outputChannelIndex = 0;
unsigned maxAllowedOutputChannels = output->numberOfChannels();
for (unsigned i = 0; i < numberOfInputs(); ++i) {
AudioNodeInput* input = this->input(i);
if (input->isConnected()) {
unsigned numberOfInputChannels = input->bus()->numberOfChannels();
// Merge channels from this particular input, but be careful not to exceed the number of
// output channels. (This can happen if there are many inputs with each input
// containing many channels.)
for (unsigned j = 0; j < numberOfInputChannels; ++j) {
if (outputChannelIndex < maxAllowedOutputChannels) {
AudioChannel* inputChannel = input->bus()->channel(j);
AudioChannel* outputChannel = output->bus()->channel(outputChannelIndex);
outputChannel->copyFrom(inputChannel);
++outputChannelIndex;
}
}
}
if (outputChannelIndex >= maxAllowedOutputChannels)
break;
}
ASSERT(outputChannelIndex == output->numberOfChannels());
}
void collectLiveEvents( BaseInstrument* instrument )
{
AudioChannel* channel = instrument->audioChannel;
std::vector<BaseAudioEvent*>* liveEvents = instrument->getLiveEvents();
// removal queue
std::vector<BaseAudioEvent*> removes;
int i = 0;
for ( i; i < liveEvents->size(); i++ )
{
BaseAudioEvent* audioEvent = liveEvents->at( i );
if ( !audioEvent->deletable())
channel->addLiveEvent( audioEvent );
else
removes.push_back( audioEvent );
}
// removal queue filled ? process it so we can safely
// remove "deleted" AudioEvents without errors occurring
if ( removes.size() > 0 )
{
int i = 0;
for ( i; i < removes.size(); i++ )
{
BaseAudioEvent* audioEvent = removes[ i ];
// remove audio event from the list
if ( std::find( liveEvents->begin(), liveEvents->end(), audioEvent ) != liveEvents->end())
{
liveEvents->erase( std::find( liveEvents->begin(), liveEvents->end(), audioEvent ));
}
instrument->removeEvent( audioEvent );
}
}
}
void ChannelMergerNode::process(ContextRenderLock& r, size_t framesToProcess)
{
auto output = this->output(0);
ASSERT_UNUSED(framesToProcess, framesToProcess == output->bus(r)->length());
// Output bus not updated yet, so just output silence. See Note * in checkNumberOfChannelsForInput
if (m_desiredNumberOfOutputChannels != output->numberOfChannels())
{
output->bus(r)->zero();
return;
}
// Merge all the channels from all the inputs into one output.
uint32_t outputChannelIndex = 0;
for (uint32_t i = 0; i < numberOfInputs(); ++i)
{
auto input = this->input(i);
if (input->isConnected())
{
uint32_t numberOfInputChannels = input->bus(r)->numberOfChannels();
// Merge channels from this particular input.
for (uint32_t j = 0; j < numberOfInputChannels; ++j)
{
AudioChannel* inputChannel = input->bus(r)->channel(j);
AudioChannel* outputChannel = output->bus(r)->channel(outputChannelIndex);
outputChannel->copyFrom(inputChannel);
++outputChannelIndex;
}
}
}
ASSERT(outputChannelIndex == output->numberOfChannels());
}
int PulseAudioDriver::setup(bool capture, bool playback, const QString& )
{
PENTER;
sample_spec.rate = frame_rate;
sample_spec.channels = 2;
sample_spec.format = PA_SAMPLE_FLOAT32NE;
assert(pa_sample_spec_valid(&sample_spec));
if (channel_map_set && channel_map.channels != sample_spec.channels) {
fprintf(stderr, "Channel map doesn't match file.\n");
return -1;
}
/* Set up a new main loop */
if (!(mainloop = pa_mainloop_new())) {
fprintf(stderr, "pa_mainloop_new() failed.\n");
return -1;
}
mainloop_api = pa_mainloop_get_api(mainloop);
int r = pa_signal_init(mainloop_api);
assert(r == 0);
/* Create a new connection context */
if (!(context = pa_context_new(mainloop_api, "Traverso"))) {
fprintf(stderr, "pa_context_new() failed.\n");
return -1;
}
pa_context_set_state_callback(context, context_state_callback, this);
/* Connect the context */
pa_context_connect(context, "", (pa_context_flags_t)0, NULL);
int ret;
/* Run the main loop */
// if (pa_mainloop_run(mainloop, &ret) < 0) {
// fprintf(stderr, "pa_mainloop_run() failed.\n");
// return -1;
// }
AudioChannel* audiochannel;
int port_flags;
char buf[32];
// TODO use the found maxchannel count for the playback stream, instead of assuming 2 !!
for (int chn = 0; chn < 2; chn++) {
snprintf (buf, sizeof(buf) - 1, "playback_%d", chn+1);
audiochannel = device->register_playback_channel(buf, "32 bit float audio", port_flags, frames_per_cycle, chn);
audiochannel->set_latency( frames_per_cycle + capture_frame_latency );
playbackChannels.append(audiochannel);
}
// TODO use the found maxchannel count for the capture stream, instead of assuming 0 !!
for (int chn = 0; chn < 2; chn++) {
snprintf (buf, sizeof(buf) - 1, "capture_%d", chn+1);
audiochannel = device->register_capture_channel(buf, "32 bit float audio", port_flags, frames_per_cycle, chn);
audiochannel->set_latency( frames_per_cycle + capture_frame_latency );
captureChannels.append(audiochannel);
}
return 1;
}
void Reverb::process(const AudioBus* sourceBus, AudioBus* destinationBus, size_t framesToProcess)
{
// Do a fairly comprehensive sanity check.
// If these conditions are satisfied, all of the source and destination pointers will be valid for the various matrixing cases.
bool isSafeToProcess = sourceBus && destinationBus && sourceBus->numberOfChannels() > 0 && destinationBus->numberOfChannels() > 0
&& framesToProcess <= MaxFrameSize && framesToProcess <= sourceBus->length() && framesToProcess <= destinationBus->length();
ASSERT(isSafeToProcess);
if (!isSafeToProcess)
return;
// For now only handle mono or stereo output
if (destinationBus->numberOfChannels() > 2) {
destinationBus->zero();
return;
}
AudioChannel* destinationChannelL = destinationBus->channel(0);
const AudioChannel* sourceChannelL = sourceBus->channel(0);
// Handle input -> output matrixing...
size_t numInputChannels = sourceBus->numberOfChannels();
size_t numOutputChannels = destinationBus->numberOfChannels();
size_t numReverbChannels = m_convolvers.size();
if (numInputChannels == 2 && numReverbChannels == 2 && numOutputChannels == 2) {
// 2 -> 2 -> 2
const AudioChannel* sourceChannelR = sourceBus->channel(1);
AudioChannel* destinationChannelR = destinationBus->channel(1);
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
m_convolvers[1]->process(sourceChannelR, destinationChannelR, framesToProcess);
} else if (numInputChannels == 1 && numOutputChannels == 2 && numReverbChannels == 2) {
// 1 -> 2 -> 2
for (int i = 0; i < 2; ++i) {
AudioChannel* destinationChannel = destinationBus->channel(i);
m_convolvers[i]->process(sourceChannelL, destinationChannel, framesToProcess);
}
} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 2) {
// 1 -> 1 -> 2
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
// simply copy L -> R
AudioChannel* destinationChannelR = destinationBus->channel(1);
bool isCopySafe = destinationChannelL->data() && destinationChannelR->data() && destinationChannelL->length() >= framesToProcess && destinationChannelR->length() >= framesToProcess;
ASSERT(isCopySafe);
if (!isCopySafe)
return;
memcpy(destinationChannelR->mutableData(), destinationChannelL->data(), sizeof(float) * framesToProcess);
} else if (numInputChannels == 1 && numReverbChannels == 1 && numOutputChannels == 1) {
// 1 -> 1 -> 1
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
} else if (numInputChannels == 2 && numReverbChannels == 4 && numOutputChannels == 2) {
// 2 -> 4 -> 2 ("True" stereo)
const AudioChannel* sourceChannelR = sourceBus->channel(1);
AudioChannel* destinationChannelR = destinationBus->channel(1);
AudioChannel* tempChannelL = m_tempBuffer->channel(0);
AudioChannel* tempChannelR = m_tempBuffer->channel(1);
// Process left virtual source
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
m_convolvers[1]->process(sourceChannelL, destinationChannelR, framesToProcess);
// Process right virtual source
m_convolvers[2]->process(sourceChannelR, tempChannelL, framesToProcess);
m_convolvers[3]->process(sourceChannelR, tempChannelR, framesToProcess);
destinationBus->sumFrom(*m_tempBuffer);
} else if (numInputChannels == 1 && numReverbChannels == 4 && numOutputChannels == 2) {
// 1 -> 4 -> 2 (Processing mono with "True" stereo impulse response)
// This is an inefficient use of a four-channel impulse response, but we should handle the case.
AudioChannel* destinationChannelR = destinationBus->channel(1);
AudioChannel* tempChannelL = m_tempBuffer->channel(0);
AudioChannel* tempChannelR = m_tempBuffer->channel(1);
// Process left virtual source
m_convolvers[0]->process(sourceChannelL, destinationChannelL, framesToProcess);
m_convolvers[1]->process(sourceChannelL, destinationChannelR, framesToProcess);
// Process right virtual source
m_convolvers[2]->process(sourceChannelL, tempChannelL, framesToProcess);
m_convolvers[3]->process(sourceChannelL, tempChannelR, framesToProcess);
destinationBus->sumFrom(*m_tempBuffer);
} else {
// Handle gracefully any unexpected / unsupported matrixing
// FIXME: add code for 5.1 support...
destinationBus->zero();
}
}
/**
* starts the render thread
* NOTE: the render thread is always active, even when the
* sequencer is paused
*/
void start()
{
OPENSL_STREAM *p;
p = android_OpenAudioDevice( AudioEngineProps::SAMPLE_RATE, AudioEngineProps::INPUT_CHANNELS,
AudioEngineProps::OUTPUT_CHANNELS, AudioEngineProps::BUFFER_SIZE );
// hardware unavailable ? halt thread, trigger JNI callback for error handler
if ( p == NULL )
{
Observer::handleHardwareUnavailable();
return;
}
// audio hardware available, start render thread
int buffer_size, i, c, ci;
buffer_size = AudioEngineProps::BUFFER_SIZE;
int outputChannels = AudioEngineProps::OUTPUT_CHANNELS;
bool isMono = outputChannels == 1;
std::vector<AudioChannel*> channels;
std::vector<AudioChannel*> channels2; // used when loop starts for gathering events at the start range
bool loopStarted = false; // whether the current buffer will exceed the end offset of the loop (read remaining samples from the start)
int loopOffset = 0; // the offset within the current buffer where we start reading from the current loops start offset
int loopAmount = 0; // amount of samples we must read from the current loops start offset
float recbufferIn [ buffer_size ]; // used for recording from device input
float outbuffer [ buffer_size * outputChannels ]; // the output buffer rendered by the hardware
// generate buffers for temporary channel buffer writes
AudioBuffer* channelBuffer = new AudioBuffer( outputChannels, buffer_size );
AudioBuffer* inbuffer = new AudioBuffer( outputChannels, buffer_size ); // accumulates all channels ("master strip")
AudioBuffer* recbuffer = new AudioBuffer( AudioEngineProps::INPUT_CHANNELS, buffer_size );
thread = 1;
// signal processors
Finalizer* limiter = new Finalizer ( 2, 500, AudioEngineProps::SAMPLE_RATE, outputChannels );
LPFHPFilter* hpf = new LPFHPFilter(( float ) AudioEngineProps::SAMPLE_RATE, 55, outputChannels );
while ( thread )
{
// erase previous buffer contents
inbuffer->silenceBuffers();
// gather the audio events by the buffer range currently being processed
int endPosition = bufferPosition + buffer_size;
channels = sequencer::getAudioEvents( channels, bufferPosition, endPosition, true );
// read pointer exceeds maximum allowed offset ? => sequencer has started its loop
// we must now also gather extra events at the start position of the seq. range
loopStarted = endPosition > max_buffer_position;
loopOffset = (( max_buffer_position + 1 ) - bufferPosition );
loopAmount = buffer_size - loopOffset;
if ( loopStarted )
{
// were we bouncing the audio ? save file and stop rendering
if ( bouncing )
{
DiskWriter::writeBufferToFile( AudioEngineProps::SAMPLE_RATE, AudioEngineProps::OUTPUT_CHANNELS, false );
// broadcast update via JNI, pass buffer identifier name to identify last recording
Observer::handleBounceComplete( 1 );
thread = 0; // stop thread, halts rendering
break;
}
else
{
endPosition -= max_buffer_position;
channels2 = sequencer::getAudioEvents( channels2, min_buffer_position, min_buffer_position + buffer_size, false );
// er? the channels are magically merged by above invocation..., performing the insert below adds the same events TWICE*POP*!?!?
//channels.insert( channels.end(), channels2.begin(), channels2.end() ); // merge the channels into one
channels2.clear(); // would clear on next "getAudioEvents"-query... but why wait ?
}
}
// record audio from Android device ?
if ( recordFromDevice && AudioEngineProps::INPUT_CHANNELS > 0 )
{
int recSamps = android_AudioIn( p, recbufferIn, AudioEngineProps::BUFFER_SIZE );
SAMPLE_TYPE* recBufferChannel = recbuffer->getBufferForChannel( 0 );
for ( int j = 0; j < recSamps; ++j )
{
recBufferChannel[ j ] = recbufferIn[ j ];//static_cast<float>( recbufferIn[ j ] );
// merge recording into current input buffer for instant monitoring
if ( monitorRecording )
{
for ( int k = 0; k < outputChannels; ++k )
inbuffer->getBufferForChannel( k )[ j ] = recBufferChannel[ j ];
}
}
}
// channel loop
int j = 0;
int channelAmount = channels.size();
for ( j; j < channelAmount; ++j )
{
AudioChannel* channel = channels[ j ];
bool isCached = channel->hasCache; // whether this channel has a fully cached buffer
bool mustCache = AudioEngineProps::CHANNEL_CACHING && channel->canCache() && !isCached; // whether to cache this channels output
bool gotBuffer = false;
int cacheReadPos = 0; // the offset we start ready from the channel buffer (when writing to cache)
SAMPLE_TYPE channelVolume = ( SAMPLE_TYPE ) channel->mixVolume;
std::vector<BaseAudioEvent*> audioEvents = channel->audioEvents;
int amount = audioEvents.size();
// clear previous channel buffer content
channelBuffer->silenceBuffers();
bool useChannelRange = channel->maxBufferPosition != 0; // channel has its own buffer range (i.e. drummachine)
int maxBufferPosition = useChannelRange ? channel->maxBufferPosition : max_buffer_position;
// we make a copy of the current buffer position indicator
int bufferPos = bufferPosition;
// ...in case the AudioChannels maxBufferPosition differs from the sequencer loop range
// note that these buffer positions are always a full bar in length (as we loop measures)
while ( bufferPos > maxBufferPosition )
bufferPos -= bytes_per_bar;
// only render sequenced events when the sequencer isn't in the paused state
// and the channel volume is actually at an audible level! ( > 0 )
if ( playing && amount > 0 && channelVolume > 0.0 )
{
if ( !isCached )
{
// write the audioEvent buffers into the main output buffer
for ( int k = 0; k < amount; ++k )
{
BaseAudioEvent* audioEvent = audioEvents[ k ];
if ( !audioEvent->isLocked()) // make sure we are allowed to query the contents
{
audioEvent->lock(); // prevent buffer mutations during this read cycle
audioEvent->mixBuffer( channelBuffer, bufferPos, min_buffer_position,
maxBufferPosition, loopStarted, loopOffset, useChannelRange );
audioEvent->unlock(); // release lock
}
}
}
else
{
channel->readCachedBuffer( channelBuffer, bufferPos );
}
}
// perform live rendering for this instrument
if ( channel->hasLiveEvents )
{
int lAmount = channel->liveEvents.size();
// the volume of the live events is divided by the channel mix as a live event
// is played on the same instrument, but just as a different voice (note the
// events can have their own mix level)
float lAmp = channel->mixVolume > 0.0 ? MAX_PHASE / channel->mixVolume : MAX_PHASE;
for ( int k = 0; k < lAmount; ++k )
{
BaseAudioEvent* vo = channel->liveEvents[ k ];
channelBuffer->mergeBuffers( vo->synthesize( buffer_size ), 0, 0, lAmp );
}
}
// apply the processing chains processors / modulators
ProcessingChain* chain = channel->processingChain;
std::vector<BaseProcessor*> processors = chain->getActiveProcessors();
for ( int k = 0; k < processors.size(); k++ )
{
BaseProcessor* processor = processors[ k ];
bool canCacheProcessor = processor->isCacheable();
// only apply processor when we're not caching or cannot cache its output
if ( !isCached || !canCacheProcessor )
{
// cannot cache this processor and we're caching ? write all contents
// of the channelBuffer into the channels cache
if ( mustCache && !canCacheProcessor )
mustCache = !writeChannelCache( channel, channelBuffer, cacheReadPos );
processors[ k ]->process( channelBuffer, channel->isMono );
}
}
// write cache if it didn't happen yet ;) (bus processors are (currently) non-cacheable)
if ( mustCache )
mustCache = !writeChannelCache( channel, channelBuffer, cacheReadPos );
// write the channel buffer into the combined output buffer, apply channel volume
// note live events are always audible as their volume is relative to the instrument
if ( channel->hasLiveEvents && channelVolume == 0.0 ) channelVolume = MAX_PHASE;
inbuffer->mergeBuffers( channelBuffer, 0, 0, channelVolume );
}
// TODO: create bus processors for these ?
// apply high pass filtering to prevent extreme low rumbling and nasty filter offsets
hpf->process( inbuffer, buffer_size );
// limit the audio to prevent clipping
limiter->process( inbuffer, isMono );
// write the accumulated buffers into the output buffer
for ( i = 0, c = 0; i < buffer_size; i++, c += outputChannels )
{
for ( ci = 0; ci < outputChannels; ci++ )
{
float sample = ( float ) inbuffer->getBufferForChannel( ci )[ i ] * volume; // apply master volume
// extreme limiting (still above the thresholds?)
if ( sample < -MAX_PHASE )
sample = -MAX_PHASE;
else if ( sample > +MAX_PHASE )
sample = +MAX_PHASE;
outbuffer[ c + ci ] = sample;
}
// update the buffer pointers and sequencer position
if ( playing )
{
if ( ++bufferPosition % bytes_per_tick == 0 )
handleSequencerPositionUpdate( android_GetTimestamp( p ));
if ( bufferPosition > max_buffer_position )
bufferPosition = min_buffer_position;
}
}
// render the buffer in the audio hardware (unless we're bouncing as writing the output
// makes it both unnecessarily audible and stalls this thread's execution
if ( !bouncing )
android_AudioOut( p, outbuffer, buffer_size * AudioEngineProps::OUTPUT_CHANNELS );
// record the output if recording state is active
if ( playing && ( recordOutput || recordFromDevice ))
{
if ( recordFromDevice ) // recording from device input ? > write the record buffer
DiskWriter::appendBuffer( recbuffer );
else // recording global output ? > write the combined buffer
DiskWriter::appendBuffer( inbuffer );
// exceeded maximum recording buffer amount ? > write current recording
if ( DiskWriter::bufferFull() || haltRecording )
{
int amountOfChannels = recordFromDevice ? AudioEngineProps::INPUT_CHANNELS : outputChannels;
DiskWriter::writeBufferToFile( AudioEngineProps::SAMPLE_RATE, amountOfChannels, true );
if ( !haltRecording )
{
DiskWriter::generateOutputBuffer(); // allocate new buffer for next iteration
++recordingFileId;
}
else {
haltRecording = false;
}
}
}
// tempo update queued ?
if ( queuedTempo != tempo )
handleTempoUpdate( queuedTempo, true );
}
android_CloseAudioDevice( p );
// clear heap memory allocated before thread loop
delete inbuffer;
delete channelBuffer;
delete limiter;
delete hpf;
}