SINT SoundSourceOpus::readSampleFramesStereo(
SINT numberOfFrames, CSAMPLE* sampleBuffer,
SINT sampleBufferSize) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(getSampleBufferSize(numberOfFrames, true) <= sampleBufferSize);
const SINT numberOfFramesTotal = math_min(
numberOfFrames, getMaxFrameIndex() - m_curFrameIndex);
CSAMPLE* pSampleBuffer = sampleBuffer;
SINT numberOfFramesRemaining = numberOfFramesTotal;
while (0 < numberOfFramesRemaining) {
int readResult = op_read_float_stereo(m_pOggOpusFile,
pSampleBuffer,
numberOfFramesRemaining * 2); // stereo
if (0 < readResult) {
m_curFrameIndex += readResult;
pSampleBuffer += readResult * 2; // stereo
numberOfFramesRemaining -= readResult;
} else {
qWarning() << "Failed to read sample data from OggOpus file:"
<< readResult;
break; // abort
}
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(numberOfFramesTotal >= numberOfFramesRemaining);
return numberOfFramesTotal - numberOfFramesRemaining;
}
SINT SoundSourceOggVorbis::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer,
SINT sampleBufferSize, bool readStereoSamples) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(getSampleBufferSize(numberOfFrames, readStereoSamples) <= sampleBufferSize);
const SINT numberOfFramesTotal = math_min(
numberOfFrames, getMaxFrameIndex() - m_curFrameIndex);
CSAMPLE* pSampleBuffer = sampleBuffer;
SINT numberOfFramesRemaining = numberOfFramesTotal;
while (0 < numberOfFramesRemaining) {
float** pcmChannels;
int currentSection;
// Use 'long' here, because ov_read_float() returns this type.
// This is an exception from the rule not to any types with
// differing sizes on different platforms.
// https://bugs.launchpad.net/mixxx/+bug/1094143
const long readResult = ov_read_float(&m_vf, &pcmChannels,
numberOfFramesRemaining, ¤tSection);
if (0 < readResult) {
m_curFrameIndex += readResult;
if (kChannelCountMono == getChannelCount()) {
if (readStereoSamples) {
for (long i = 0; i < readResult; ++i) {
*pSampleBuffer++ = pcmChannels[0][i];
*pSampleBuffer++ = pcmChannels[0][i];
}
} else {
for (long i = 0; i < readResult; ++i) {
*pSampleBuffer++ = pcmChannels[0][i];
}
}
} else if (readStereoSamples || (kChannelCountStereo == getChannelCount())) {
for (long i = 0; i < readResult; ++i) {
*pSampleBuffer++ = pcmChannels[0][i];
*pSampleBuffer++ = pcmChannels[1][i];
}
} else {
for (long i = 0; i < readResult; ++i) {
for (SINT j = 0; j < getChannelCount(); ++j) {
*pSampleBuffer++ = pcmChannels[j][i];
}
}
}
numberOfFramesRemaining -= readResult;
} else {
qWarning() << "Failed to read from OggVorbis file:" << readResult;
break; // abort
}
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(numberOfFramesTotal >= numberOfFramesRemaining);
return numberOfFramesTotal - numberOfFramesRemaining;
}
SINT AudioSource::readSampleFramesStereo(
SINT numberOfFrames,
CSAMPLE* sampleBuffer,
SINT sampleBufferSize) {
DEBUG_ASSERT(getSampleBufferSize(numberOfFrames, true) <= sampleBufferSize);
switch (getChannelCount()) {
case 1: // mono channel
{
const SINT readFrameCount = readSampleFrames(
numberOfFrames, sampleBuffer);
SampleUtil::doubleMonoToDualMono(sampleBuffer, readFrameCount);
return readFrameCount;
}
case 2: // stereo channel(s)
{
return readSampleFrames(numberOfFrames, sampleBuffer);
}
default: // multiple (3 or more) channels
{
const SINT numberOfSamplesToRead = frames2samples(numberOfFrames);
if (numberOfSamplesToRead <= sampleBufferSize) {
// efficient in-place transformation
const SINT readFrameCount = readSampleFrames(
numberOfFrames, sampleBuffer);
SampleUtil::copyMultiToStereo(sampleBuffer, sampleBuffer,
readFrameCount, getChannelCount());
return readFrameCount;
} else {
// inefficient transformation through a temporary buffer
qDebug() << "Performance warning:"
<< "Allocating a temporary buffer of size"
<< numberOfSamplesToRead << "for reading stereo samples."
<< "The size of the provided sample buffer is"
<< sampleBufferSize;
SampleBuffer tempBuffer(numberOfSamplesToRead);
const SINT readFrameCount = readSampleFrames(
numberOfFrames, tempBuffer.data());
SampleUtil::copyMultiToStereo(sampleBuffer, tempBuffer.data(),
readFrameCount, getChannelCount());
return readFrameCount;
}
}
}
}
SINT SoundSourceMp3::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer,
SINT sampleBufferSize, bool readStereoSamples) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(getSampleBufferSize(numberOfFrames, readStereoSamples) <= sampleBufferSize);
const SINT numberOfFramesTotal = math_min(
numberOfFrames, getMaxFrameIndex() - m_curFrameIndex);
CSAMPLE* pSampleBuffer = sampleBuffer;
SINT numberOfFramesRemaining = numberOfFramesTotal;
while (0 < numberOfFramesRemaining) {
if (0 >= m_madSynthCount) {
// When all decoded output data has been consumed...
DEBUG_ASSERT(0 == m_madSynthCount);
// ...decode the next MP3 frame
DEBUG_ASSERT(m_madStream.buffer);
DEBUG_ASSERT(m_madStream.this_frame);
// WARNING: Correctly evaluating and handling the result
// of mad_frame_decode() has proven to be extremely tricky.
// Don't change anything at the following lines of code
// unless you know what you are doing!!!
unsigned char const* pMadThisFrame = m_madStream.this_frame;
if (mad_frame_decode(&m_madFrame, &m_madStream)) {
// Something went wrong when decoding the frame...
if (MAD_ERROR_BUFLEN == m_madStream.error) {
// Abort
break;
}
if (isUnrecoverableError(m_madStream)) {
qWarning() << "Unrecoverable MP3 frame decoding error:"
<< mad_stream_errorstr(&m_madStream);
// Abort decoding
break;
}
if (isRecoverableError(m_madStream)) {
if (pMadThisFrame != m_madStream.this_frame) {
// Ignore all recoverable errors (and especially
// "lost synchronization" warnings) while skipping
// over prefetched frames after seeking.
if (pSampleBuffer) {
qWarning() << "Recoverable MP3 frame decoding error:"
<< mad_stream_errorstr(&m_madStream);
} else {
// Decoded samples will simply be discarded
qDebug() << "Recoverable MP3 frame decoding error while skipping:"
<< mad_stream_errorstr(&m_madStream);
}
}
// Continue decoding
}
}
if (pMadThisFrame == m_madStream.this_frame) {
qDebug() << "Retry decoding MP3 frame @" << m_curFrameIndex;
// Retry decoding
continue;
}
DEBUG_ASSERT(isStreamValid(m_madStream));
#ifndef QT_NO_DEBUG_OUTPUT
const SINT madFrameChannelCount = MAD_NCHANNELS(&m_madFrame.header);
if (madFrameChannelCount != getChannelCount()) {
qDebug() << "MP3 frame header with mismatching number of channels"
<< madFrameChannelCount << "<>" << getChannelCount();
}
#endif
// Once decoded the frame is synthesized to PCM samples
mad_synth_frame(&m_madSynth, &m_madFrame);
#ifndef QT_NO_DEBUG_OUTPUT
const SINT madSynthSampleRate = m_madSynth.pcm.samplerate;
if (madSynthSampleRate != getSamplingRate()) {
qDebug() << "Reading MP3 data with different sampling rate"
<< madSynthSampleRate << "<>" << getSamplingRate();
}
#endif
m_madSynthCount = m_madSynth.pcm.length;
DEBUG_ASSERT(0 < m_madSynthCount);
}
const SINT synthReadCount = math_min(
m_madSynthCount, numberOfFramesRemaining);
if (pSampleBuffer) {
DEBUG_ASSERT(m_madSynthCount <= m_madSynth.pcm.length);
const SINT madSynthOffset =
m_madSynth.pcm.length - m_madSynthCount;
DEBUG_ASSERT(madSynthOffset < m_madSynth.pcm.length);
const SINT madSynthChannelCount = m_madSynth.pcm.channels;
DEBUG_ASSERT(0 < madSynthChannelCount);
DEBUG_ASSERT(madSynthChannelCount <= getChannelCount());
#ifndef QT_NO_DEBUG_OUTPUT
if (madSynthChannelCount != getChannelCount()) {
qDebug() << "Reading MP3 data with different number of channels"
<< madSynthChannelCount << "<>" << getChannelCount();
}
#endif
if (kChannelCountMono == madSynthChannelCount) {
// MP3 frame contains a mono signal
if (readStereoSamples || (kChannelCountStereo == getChannelCount())) {
// The reader explicitly requested a stereo signal
// or the AudioSource itself provides a stereo signal.
// Mono -> Stereo: Copy 1st channel twice
for (SINT i = 0; i < synthReadCount; ++i) {
const CSAMPLE sampleValue = madScaleSampleValue(
m_madSynth.pcm.samples[0][madSynthOffset + i]);
*pSampleBuffer++ = sampleValue;
*pSampleBuffer++ = sampleValue;
}
} else {
// Mono -> Mono: Copy 1st channel
for (SINT i = 0; i < synthReadCount; ++i) {
const CSAMPLE sampleValue = madScaleSampleValue(
m_madSynth.pcm.samples[0][madSynthOffset + i]);
*pSampleBuffer++ = sampleValue;
}
}
} else {
// MP3 frame contains a stereo signal
DEBUG_ASSERT(kChannelCountStereo == madSynthChannelCount);
// If the MP3 frame contains a stereo signal then the whole
// AudioSource must also provide 2 channels, because the
// maximum channel count of all MP3 frames is used.
DEBUG_ASSERT(kChannelCountStereo == getChannelCount());
// Stereo -> Stereo: Copy 1st + 2nd channel
for (SINT i = 0; i < synthReadCount; ++i) {
*pSampleBuffer++ = madScaleSampleValue(
m_madSynth.pcm.samples[0][madSynthOffset + i]);
*pSampleBuffer++ = madScaleSampleValue(
m_madSynth.pcm.samples[1][madSynthOffset + i]);
}
}
}
// consume decoded output data
m_madSynthCount -= synthReadCount;
m_curFrameIndex += synthReadCount;
numberOfFramesRemaining -= synthReadCount;
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(numberOfFramesTotal >= numberOfFramesRemaining);
return numberOfFramesTotal - numberOfFramesRemaining;
}
SINT SoundSourceFLAC::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer,
SINT sampleBufferSize, bool readStereoSamples) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(getSampleBufferSize(numberOfFrames, readStereoSamples) <= sampleBufferSize);
const SINT numberOfFramesTotal =
math_min(numberOfFrames, getMaxFrameIndex() - m_curFrameIndex);
const SINT numberOfSamplesTotal = frames2samples(numberOfFramesTotal);
CSAMPLE* outBuffer = sampleBuffer;
SINT numberOfSamplesRemaining = numberOfSamplesTotal;
while (0 < numberOfSamplesRemaining) {
// If our buffer from libflac is empty (either because we explicitly cleared
// it or because we've simply used all the samples), ask for a new buffer
if (m_sampleBuffer.isEmpty()) {
// Save the current frame index
const SINT curFrameIndexBeforeProcessing = m_curFrameIndex;
// Documentation of FLAC__stream_decoder_process_single():
// "Depending on what was decoded, the metadata or write callback
// will be called with the decoded metadata block or audio frame."
// See also: https://xiph.org/flac/api/group__flac__stream__decoder.html#ga9d6df4a39892c05955122cf7f987f856
if (!FLAC__stream_decoder_process_single(m_decoder)) {
qWarning() << "Failed to decode FLAC file"
<< m_file.fileName();
break; // abort
}
// After seeking we might need to skip some samples if the decoder
// complained that it has lost sync for some malformed(?) files
if (curFrameIndexBeforeProcessing != m_curFrameIndex) {
if (curFrameIndexBeforeProcessing > m_curFrameIndex) {
qWarning() << "Trying to adjust frame index"
<< m_curFrameIndex << "<>" << curFrameIndexBeforeProcessing
<< "while decoding FLAC file"
<< m_file.fileName();
skipSampleFrames(curFrameIndexBeforeProcessing - m_curFrameIndex);
} else {
qWarning() << "Unexpected frame index"
<< m_curFrameIndex << "<>" << curFrameIndexBeforeProcessing
<< "while decoding FLAC file"
<< m_file.fileName();
break; // abort
}
}
DEBUG_ASSERT(curFrameIndexBeforeProcessing == m_curFrameIndex);
}
if (m_sampleBuffer.isEmpty()) {
break; // EOF
}
const SampleBuffer::ReadableChunk readableChunk(
m_sampleBuffer.readFromHead(numberOfSamplesRemaining));
const SINT framesToCopy = samples2frames(readableChunk.size());
if (outBuffer) {
if (readStereoSamples && (kChannelCountStereo != getChannelCount())) {
if (kChannelCountMono == getChannelCount()) {
SampleUtil::copyMonoToDualMono(outBuffer,
readableChunk.data(),
framesToCopy);
} else {
SampleUtil::copyMultiToStereo(outBuffer,
readableChunk.data(),
framesToCopy, getChannelCount());
}
outBuffer += framesToCopy * kChannelCountStereo;
} else {
SampleUtil::copy(outBuffer, readableChunk.data(), readableChunk.size());
outBuffer += readableChunk.size();
}
}
m_curFrameIndex += framesToCopy;
numberOfSamplesRemaining -= readableChunk.size();
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(numberOfSamplesTotal >= numberOfSamplesRemaining);
return samples2frames(numberOfSamplesTotal - numberOfSamplesRemaining);
}
