TEST( BaseAudioEvent, MixBufferLoopeableEvent )
{
BaseAudioEvent* audioEvent = new BaseAudioEvent();
int sourceSize = 16;
AudioBuffer* sourceBuffer = new AudioBuffer( 1, sourceSize );
SAMPLE_TYPE* rawBuffer = sourceBuffer->getBufferForChannel( 0 );
fillAudioBuffer( sourceBuffer );
audioEvent->setBuffer( sourceBuffer, false );
audioEvent->setLoopeable( true );
audioEvent->setSampleLength( 16 * 4 ); // thus will loop 4 times
audioEvent->positionEvent ( 0, 16, 0 );
// create an output buffer at a size smaller than the source buffer length
int outputSize = ( int )(( double ) sourceSize * .4 );
AudioBuffer* targetBuffer = new AudioBuffer( sourceBuffer->amountOfChannels, outputSize );
int minBufferPos = audioEvent->getSampleStart();
int bufferPos = minBufferPos;
int maxBufferPos = audioEvent->getSampleEnd();
// test the seamless mixing over multiple iterations
for ( ; bufferPos < maxBufferPos; bufferPos += outputSize )
{
// mix buffer contents
targetBuffer->silenceBuffers();
bool loopStarted = bufferPos + ( outputSize - 1 ) > maxBufferPos;
int loopOffset = ( maxBufferPos - bufferPos ) + 1;
audioEvent->mixBuffer( targetBuffer, bufferPos, minBufferPos, maxBufferPos, loopStarted, loopOffset, false );
// assert results
SAMPLE_TYPE* mixedBuffer = targetBuffer->getBufferForChannel( 0 );
for ( int i = 0; i < outputSize; ++i )
{
int compareOffset = ( bufferPos + i ) % sourceSize;
EXPECT_EQ( rawBuffer[ compareOffset ], mixedBuffer[ i ] )
<< "expected mixed buffer contents to equal the source contents at mixed offset " << i << " for source offset " << compareOffset;
}
}
delete targetBuffer;
delete sourceBuffer;
delete audioEvent;
}
void JavaUtilities::createSampleFromBuffer( jstring aKey, jint aBufferLength, jint aChannelAmount, jdoubleArray aBuffer, jdoubleArray aOptRightBuffer )
{
AudioBuffer* sampleBuffer = new AudioBuffer( aChannelAmount, aBufferLength );
int i = 0;
// get a pointer to the Java array
jdouble* c_array;
c_array = JavaBridge::getEnvironment()->GetDoubleArrayElements( aBuffer, 0 );
// exception checking
if ( c_array == NULL )
return;
// copy buffer contents
SAMPLE_TYPE* channelBuffer = sampleBuffer->getBufferForChannel( 0 );
for ( i = 0; i < aBufferLength; i++ )
channelBuffer[ i ] = ( SAMPLE_TYPE ) c_array[ i ];
// release the memory so Java can have it again
JavaBridge::getEnvironment()->ReleaseDoubleArrayElements( aBuffer, c_array, 0 );
// stereo ?
if ( aChannelAmount == 2 )
{
c_array = JavaBridge::getEnvironment()->GetDoubleArrayElements( aOptRightBuffer, 0 );
// exception checking
if ( c_array == NULL )
return;
// copy buffer contents
channelBuffer = sampleBuffer->getBufferForChannel( 1 );
for ( i = 0; i < aBufferLength; i++ )
channelBuffer[ i ] = ( SAMPLE_TYPE ) c_array[ i ];
// release the memory so Java can have it again
JavaBridge::getEnvironment()->ReleaseDoubleArrayElements( aOptRightBuffer, c_array, 0 );
}
// convert jstring to std::string
const char* s = JavaBridge::getEnvironment()->GetStringUTFChars( aKey, NULL );
std::string theKey = s;
JavaBridge::getEnvironment()->ReleaseStringUTFChars( aKey, s );
SampleManager::setSample( theKey, sampleBuffer );
}
AudioBuffer* AudioBuffer::clone()
{
AudioBuffer* output = new AudioBuffer( amountOfChannels, bufferSize );
for ( int i = 0; i < amountOfChannels; ++i )
{
SAMPLE_TYPE* sourceBuffer = getBufferForChannel( i );
SAMPLE_TYPE* targetBuffer = output->getBufferForChannel( i );
memcpy( targetBuffer, sourceBuffer, bufferSize * sizeof( SAMPLE_TYPE ));
}
return output;
}
/**
* write the contents of the write buffer into
* an output file, this will only write content
* up until the point if was written to in case
* the buffer wasn't full yet
*/
void writeBufferToFile( int aSampleRate, int aNumChannels, bool broadcastUpdate )
{
// quick assertion
if ( cachedBuffer == 0 )
return;
// copy string contents for appending of filename
std::string outputFile = std::string( outputDirectory.c_str());
int bufferSize = outputBufferSize;
// recorded less than maximum available in buffer ? cut silence
// by writing recording into temporary buffers
if ( outputWriterIndex < bufferSize )
{
bufferSize = outputWriterIndex;
AudioBuffer* tempBuffer = new AudioBuffer( aNumChannels, bufferSize );
for ( int i = 0; i < bufferSize; ++i )
{
for ( int c = 0; c < aNumChannels; ++c )
tempBuffer->getBufferForChannel( c )[ i ] = cachedBuffer->getBufferForChannel( c )[ i ];
}
WaveWriter::bufferToFile( outputFile.append( SSTR( AudioEngine::recordingFileId )),
tempBuffer, aSampleRate );
// free memory allocated by temporary buffer
delete tempBuffer;
}
else
{
WaveWriter::bufferToFile( outputFile.append( SSTR( AudioEngine::recordingFileId )),
cachedBuffer, aSampleRate );
}
flushOutput(); // free memory
// broadcast update, pass buffer identifier to identify last recording
if ( broadcastUpdate )
Notifier::broadcast( Notifications::RECORDING_STATE_UPDATED, AudioEngine::recordingFileId );
}
TEST( AudioEngine, Output )
{
AudioEngine::test_program = 2; // help mocked OpenSL IO identify which test is running
AudioEngine::test_successful = false;
// prepare engine environment
SequencerController* controller = new SequencerController();
controller->prepare( 16, 48000, 130.0f, 4, 4 ); // 130 BPM in 4/4 time at 48 kHz sample rate w/buffer size of 16 samples
controller->setTempoNow( 130.0f, 4, 4 );
controller->rewind();
AudioEngine::volume = 1; // QQQ : later on we test mix volume ;)
// create an AudioEvent that holds a simple waveform
// the resulting 16 sample mono buffer contains the following samples:
//
// -1,-1,-1,-1,0,0,0,0,1,1,1,1,0,0,0,0
//
// the event will last for an entire measure in duration
BaseInstrument* instrument = new BaseInstrument();
BaseAudioEvent* event = new BaseAudioEvent( instrument );
AudioBuffer* buffer = new AudioBuffer( 1, 16 );
SAMPLE_TYPE* rawBuffer = buffer->getBufferForChannel( 0 );
for ( int i = 0; i < 4; ++i )
rawBuffer[ i ] = ( SAMPLE_TYPE ) -MAX_PHASE;
for ( int i = 4; i < 8; ++i )
rawBuffer[ i ] = ( SAMPLE_TYPE ) 0;
for ( int i = 8; i < 12; ++i )
rawBuffer[ i ] = ( SAMPLE_TYPE ) MAX_PHASE;
for ( int i = 12; i < 16; ++i )
rawBuffer[ i ] = ( SAMPLE_TYPE ) 0;
event->setBuffer( buffer, false );
event->setLoopeable( true );
event->setSampleLength( AudioEngine::samples_per_bar );
event->positionEvent( 0, 16, 0 );
event->addToSequencer();
// start the engine
controller->setPlaying( true );
AudioEngine::start();
// evaluate results (assertions are made in mock_opensl_io.cpp)
ASSERT_TRUE( AudioEngine::test_successful )
<< "expected test to be successful";
EXPECT_EQ( 3, AudioEngine::test_program )
<< "expected test program to have incremented";
// clean up
controller->setPlaying( false );
AudioEngine::render_iterations = 0;
delete controller;
delete instrument;
delete event;
delete buffer;
}
/**
* 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;
}
TEST( BaseAudioEvent, MixBuffer )
{
BaseAudioEvent* audioEvent = new BaseAudioEvent();
int sampleLength = randomInt( 8, 24 );
int sampleStart = randomInt( 0, ( int )( sampleLength / 2 ));
audioEvent->setSampleStart ( sampleStart );
audioEvent->setSampleLength( sampleLength );
int sampleEnd = audioEvent->getSampleEnd();
AudioBuffer* buffer = fillAudioBuffer( new AudioBuffer( randomInt( 1, 4 ), sampleLength ));
audioEvent->setBuffer( buffer, true );
float volume = randomFloat();
audioEvent->setVolume( volume );
//std::cout << " ss: " << sampleStart << " se: " << sampleEnd << " sl: " << sampleLength << " ch: " << buffer->amountOfChannels;
// create a temporary buffer to write output in, ensure it is smaller than the event buffer
AudioBuffer* targetBuffer = new AudioBuffer( buffer->amountOfChannels, randomInt( 2, 4 ));
int buffersToWrite = targetBuffer->bufferSize;
ASSERT_FALSE( bufferHasContent( targetBuffer ))
<< "expected target buffer to be silent after creation, but it has content";
// test 1. mix without loopable range
int maxBufferPos = sampleLength * 2; // use a "loop range" larger than the size of the events length
int minBufferPos = randomInt( 0, maxBufferPos / 2 );
int bufferPos = randomInt( minBufferPos, maxBufferPos - 1 );
bool loopStarted = false;
int loopOffset = 0;
// if the random bufferPosition wasn't within the events sampleStart and sampleEnd range, we expect no content
bool expectContent = ( bufferPos >= sampleStart && bufferPos <= sampleEnd ) ||
(( bufferPos + buffersToWrite ) >= sampleStart && ( bufferPos + buffersToWrite ) <= sampleEnd );
//std::cout << " expected content: " << expectContent << " for buffer size: " << buffersToWrite;
//std::cout << " min: " << minBufferPos << " max: " << maxBufferPos << " cur: " << bufferPos;
audioEvent->mixBuffer( targetBuffer, bufferPos, minBufferPos, maxBufferPos, loopStarted, loopOffset, false );
// validate buffer contents after mixing
if ( expectContent )
{
for ( int c = 0, ca = targetBuffer->amountOfChannels; c < ca; ++c )
{
SAMPLE_TYPE* buffer = targetBuffer->getBufferForChannel( c );
SAMPLE_TYPE* sourceBuffer = audioEvent->getBuffer()->getBufferForChannel( c );
SAMPLE_TYPE expectedSample;
for ( int i = 0; i < buffersToWrite; ++i )
{
int r = i + bufferPos; // read pointer for the source buffer
if ( r >= maxBufferPos && !loopStarted )
r -= ( maxBufferPos - minBufferPos );
if ( r >= sampleStart && r <= sampleEnd )
{
r -= sampleStart; // substract audioEvent start position
expectedSample = sourceBuffer[ r ] * volume;
}
else {
expectedSample = 0.0;
}
SAMPLE_TYPE sample = buffer[ i ];
EXPECT_EQ( expectedSample, sample )
<< "expected mixed sample at " << i << " to be equal to the calculated expected sample at read offset " << r;
}
}
}
else {
ASSERT_FALSE( bufferHasContent( targetBuffer ))
<< "expected target buffer to contain no content after mixing for an out-of-range buffer position";
}
// test 2. mixing within a loopable range (implying sequencer is starting a loop)
targetBuffer->silenceBuffers();
ASSERT_FALSE( bufferHasContent( targetBuffer ))
<< "expected target buffer to be silent after silencing, but it still has content";
bufferPos = randomInt( minBufferPos, maxBufferPos - 1 );
loopStarted = true;
loopOffset = ( maxBufferPos - bufferPos ) + 1;
// pre calculate at which buffer iterator the looping will commence
// loopStartIteratorPosition describes at which sequencer position the loop starts
// loopStartWritePointer describes at which position in the targetBuffer the loop is written to
// amountOfLoopedWrites is the amount of samples written in the loop
// loopStartReadPointer describes at which position the samples from the source audioEvent will be read when loop starts
// loopStartReadPointerEnd describes the last position the samples from the source audioEvent will be read for the amount of loop writes
int loopStartIteratorPosition = maxBufferPos + 1;
int loopStartWritePointer = loopOffset;
int loopStartReadPointer = minBufferPos;
int amountOfLoopedWrites = ( bufferPos + buffersToWrite ) - loopStartIteratorPosition;
int loopStartReadPointerEnd = ( loopStartReadPointer + amountOfLoopedWrites ) - 1;
expectContent = ( bufferPos >= sampleStart && bufferPos <= sampleEnd ) ||
(( bufferPos + buffersToWrite ) >= sampleStart && ( bufferPos + buffersToWrite ) <= sampleEnd ) ||
( loopStartIteratorPosition > maxBufferPos && (
( loopStartReadPointer >= sampleStart && loopStartReadPointer <= sampleEnd ) ||
( loopStartReadPointerEnd >= sampleStart && loopStartReadPointerEnd <= sampleEnd )));
audioEvent->mixBuffer( targetBuffer, bufferPos, minBufferPos, maxBufferPos, loopStarted, loopOffset, false );
//std::cout << " expected content: " << expectContent << " for buffer size: " << buffersToWrite;
//std::cout << " min: " << minBufferPos << " max: " << maxBufferPos << " cur: " << bufferPos << " loop offset: " << loopOffset;
if ( expectContent )
{
for ( int c = 0, ca = targetBuffer->amountOfChannels; c < ca; ++c )
{
SAMPLE_TYPE* buffer = targetBuffer->getBufferForChannel( c );
SAMPLE_TYPE* sourceBuffer = audioEvent->getBuffer()->getBufferForChannel( c );
for ( int i = 0; i < buffersToWrite; ++i )
{
SAMPLE_TYPE expectedSample = 0.0;
int r = i + bufferPos; // read pointer for the source buffer
if ( i >= loopOffset )
r = minBufferPos + ( i - loopOffset );
if ( r >= sampleStart && r <= sampleEnd )
{
r -= sampleStart; // substract audioEvent start position
expectedSample = sourceBuffer[ r ] * volume;
}
SAMPLE_TYPE sample = buffer[ i ];
EXPECT_EQ( expectedSample, sample )
<< "expected mixed sample at " << i << " to be equal to the calculated expected sample at read "
<< "offset " << r << " ( sanitized from " << ( i + bufferPos ) << " )";
}
}
}
else {
ASSERT_FALSE( bufferHasContent( targetBuffer ))
<< "expected output buffer to contain no content after mixing for an out-of-range buffer position";
}
delete audioEvent;
delete targetBuffer;
delete buffer;
}