// static
int SourceFileOggVorbis::seekFn( void *datasource, ogg_int64_t offset, int whence )
{
auto sourceFile = (SourceFileOggVorbis *)datasource;
switch( whence ) {
case SEEK_SET:
sourceFile->mStream->seekAbsolute( (off_t)offset );
break;
case SEEK_CUR:
sourceFile->mStream->seekRelative( (off_t)offset );
break;
case SEEK_END:
sourceFile->mStream->seekAbsolute( sourceFile->mStream->size() );
break;
default:
CI_ASSERT_NOT_REACHABLE();
return -1;
}
return 0;
}
void TargetFileMediaFoundation::performWrite( const Buffer *buffer, size_t numFrames, size_t frameOffset )
{
// create media sample
::IMFSample *mediaSample;
HRESULT hr = ::MFCreateSample( &mediaSample );
CI_ASSERT( hr == S_OK );
auto samplePtr = ci::msw::makeComUnique( mediaSample );
double lengthSeconds = (double)numFrames / (double)mSampleRate;
const LONGLONG sampleDuration = secondsToNanoSeconds( lengthSeconds );
hr = mediaSample->SetSampleDuration( sampleDuration );
CI_ASSERT( hr == S_OK );
double currentTimeSeconds = (double)frameOffset / (double)mSampleRate;
const LONGLONG sampleTime = secondsToNanoSeconds( currentTimeSeconds );
hr = mediaSample->SetSampleTime( sampleTime );
CI_ASSERT( hr == S_OK );
// create media buffer and fill with audio samples.
DWORD bufferSizeBytes = numFrames * buffer->getNumChannels() * mSampleSize;
::IMFMediaBuffer *mediaBuffer;
hr = ::MFCreateMemoryBuffer( bufferSizeBytes, &mediaBuffer );
CI_ASSERT( hr == S_OK );
hr = mediaBuffer->SetCurrentLength( bufferSizeBytes );
CI_ASSERT( hr == S_OK );
hr = mediaSample->AddBuffer( mediaBuffer );
CI_ASSERT( hr == S_OK );
BYTE *audioData;
hr = mediaBuffer->Lock( &audioData, NULL, NULL );
CI_ASSERT( hr == S_OK );
if( mSampleType == SampleType::FLOAT_32 ) {
float *destFloatSamples = (float *)audioData;
if( mNumChannels == 1 )
memcpy( destFloatSamples, buffer->getData(), numFrames * mSampleSize );
else
dsp::interleave( buffer->getData(), destFloatSamples, buffer->getNumFrames(), mNumChannels, numFrames );
}
else if( mSampleType == SampleType::INT_16 ) {
int16_t *destInt16Samples = (int16_t *)audioData;
dsp::interleave( buffer->getData(), destInt16Samples, buffer->getNumFrames(), mNumChannels, numFrames );
}
else if( mSampleType == SampleType::INT_24 ) {
char *destInt24Samples = (char *)audioData;
if( mNumChannels == 1 )
dsp::convertFloatToInt24( buffer->getData(), destInt24Samples, numFrames );
else {
if( mBitConverterBuffer.getNumFrames() != numFrames || mBitConverterBuffer.getNumChannels() != mNumChannels )
mBitConverterBuffer.setSize( numFrames, mNumChannels );
dsp::interleaveBuffer( buffer, &mBitConverterBuffer );
dsp::convertFloatToInt24( mBitConverterBuffer.getData(), destInt24Samples, numFrames * mNumChannels );
}
}
else
CI_ASSERT_NOT_REACHABLE();
hr = mediaBuffer->Unlock();
CI_ASSERT( hr == S_OK );
hr = mSinkWriter->WriteSample( mStreamIndex, mediaSample );
CI_ASSERT( hr == S_OK );
}
size_t SourceFileMediaFoundation::processNextReadSample()
{
::IMFSample *mediaSample;
DWORD streamFlags = 0;
LONGLONG timeStamp;
HRESULT hr = mSourceReader->ReadSample( MF_SOURCE_READER_FIRST_AUDIO_STREAM, 0, NULL, &streamFlags, &timeStamp, &mediaSample );
CI_ASSERT( hr == S_OK );
if( streamFlags & MF_SOURCE_READERF_CURRENTMEDIATYPECHANGED ) {
CI_LOG_W( "type change unhandled" );
return 0;
}
if( streamFlags & MF_SOURCE_READERF_ENDOFSTREAM ) {
// end of file
return 0;
}
if( ! mediaSample ) {
// out of samples
mediaSample->Release();
return 0;
}
auto samplePtr = ci::msw::makeComUnique( mediaSample );
DWORD bufferCount;
hr = samplePtr->GetBufferCount( &bufferCount );
CI_ASSERT( hr == S_OK );
CI_ASSERT( bufferCount == 1 ); // just looking out for a file type with more than one buffer.. haven't seen one yet.
// get the buffer
::IMFMediaBuffer *mediaBuffer;
BYTE *audioData = NULL;
DWORD audioDataLength;
hr = samplePtr->ConvertToContiguousBuffer( &mediaBuffer );
hr = mediaBuffer->Lock( &audioData, NULL, &audioDataLength );
size_t numChannels = mNumChannels;
size_t numFramesRead = audioDataLength / ( mBytesPerSample * numChannels );
mReadBuffer.setNumFrames( numFramesRead );
if( mSampleType == SampleType::FLOAT_32 ) {
float *sourceFloatSamples = (float *)audioData;
if( numChannels == 1 )
memcpy( mReadBuffer.getData(), sourceFloatSamples, numFramesRead * sizeof( float ) );
else
dsp::deinterleave( sourceFloatSamples, mReadBuffer.getData(), mReadBuffer.getNumFrames(), numChannels, numFramesRead );
}
else if( mSampleType == SampleType::INT_16 ) {
int16_t *sourceInt16Samples = (int16_t *)audioData;
dsp::deinterleave( sourceInt16Samples, mReadBuffer.getData(), mReadBuffer.getNumFrames(), numChannels, numFramesRead );
}
else if( mSampleType == SampleType::INT_24 ) {
const char *sourceInt24Samples = (const char *)audioData;
if( numChannels == 1 )
dsp::convertInt24ToFloat( sourceInt24Samples, mReadBuffer.getData(), numFramesRead );
else {
if( mBitConverterBuffer.getNumFrames() != numFramesRead )
mBitConverterBuffer.setNumFrames( numFramesRead );
dsp::convertInt24ToFloat( sourceInt24Samples, mBitConverterBuffer.getData(), numFramesRead * numChannels );
dsp::deinterleave( mBitConverterBuffer.getData(), mReadBuffer.getData(), mReadBuffer.getNumFrames(), numChannels, numFramesRead );
}
}
else
CI_ASSERT_NOT_REACHABLE();
hr = mediaBuffer->Unlock();
CI_ASSERT( hr == S_OK );
mediaBuffer->Release();
return numFramesRead;
}
