void start_process() {
OPENSL_STREAM *p;
int samps, i, j;
float inbuffer[VECSAMPS_MONO], outbuffer[VECSAMPS_STEREO];
p = android_OpenAudioDevice(SR,1,2,BUFFERFRAMES);
if(p == NULL) return;
on = 1;
while(on) {
samps = android_AudioIn(p,inbuffer,VECSAMPS_MONO);
for(i = 0, j=0; i < samps; i++, j+=2)
outbuffer[j] = outbuffer[j+1] = inbuffer[i];
android_AudioOut(p,outbuffer,samps*2);
}
android_CloseAudioDevice(p);
}
/**
* 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;
}