Result SoundSourceModPlug::tryOpen(const AudioSourceConfig& /*audioSrcCfg*/) {
ScopedTimer t("SoundSourceModPlug::open()");
// read module file to byte array
const QString fileName(getLocalFileName());
QFile modFile(fileName);
qDebug() << "[ModPlug] Loading ModPlug module " << modFile.fileName();
modFile.open(QIODevice::ReadOnly);
m_fileBuf = modFile.readAll();
modFile.close();
// get ModPlugFile descriptor for later access
m_pModFile = ModPlug::ModPlug_Load(m_fileBuf.constData(),
m_fileBuf.length());
if (m_pModFile == NULL) {
// an error occured
t.cancel();
qDebug() << "[ModPlug] Could not load module file: " << fileName;
return ERR;
}
DEBUG_ASSERT(0 == (kChunkSizeInBytes % sizeof(m_sampleBuf[0])));
const SINT chunkSizeInSamples = kChunkSizeInBytes / sizeof(m_sampleBuf[0]);
const SampleBuffer::size_type bufferSizeLimitInSamples = s_bufferSizeLimit / sizeof(m_sampleBuf[0]);
// Estimate size of sample buffer (for better performance) aligned
// with the chunk size. Beware: Module length estimation is unreliable
// due to loops!
const SampleBuffer::size_type estimateMilliseconds =
ModPlug::ModPlug_GetLength(m_pModFile);
const SampleBuffer::size_type estimateSamples =
estimateMilliseconds * kChannelCount * kSamplingRate;
const SampleBuffer::size_type estimateChunks =
(estimateSamples + (chunkSizeInSamples - 1)) / chunkSizeInSamples;
const SampleBuffer::size_type sampleBufferCapacity = math_min(
estimateChunks * chunkSizeInSamples, bufferSizeLimitInSamples);
m_sampleBuf.reserve(sampleBufferCapacity);
qDebug() << "[ModPlug] Reserved " << m_sampleBuf.capacity() << " #samples";
// decode samples into sample buffer
while (m_sampleBuf.size() < bufferSizeLimitInSamples) {
// reserve enough space in sample buffer
const SampleBuffer::size_type currentSize = m_sampleBuf.size();
m_sampleBuf.resize(currentSize + chunkSizeInSamples);
const int bytesRead = ModPlug::ModPlug_Read(m_pModFile,
&m_sampleBuf[currentSize],
kChunkSizeInBytes);
// adjust size of sample buffer after reading
if (0 < bytesRead) {
DEBUG_ASSERT(0 == (bytesRead % sizeof(m_sampleBuf[0])));
const SampleBuffer::size_type samplesRead = bytesRead / sizeof(m_sampleBuf[0]);
m_sampleBuf.resize(currentSize + samplesRead);
} else {
// nothing read -> EOF
m_sampleBuf.resize(currentSize);
break; // exit loop
}
}
qDebug() << "[ModPlug] Filled Sample buffer with " << m_sampleBuf.size()
<< " samples.";
qDebug() << "[ModPlug] Sample buffer has "
<< m_sampleBuf.capacity() - m_sampleBuf.size()
<< " samples unused capacity.";
setChannelCount(kChannelCount);
setSamplingRate(kSamplingRate);
setFrameCount(samples2frames(m_sampleBuf.size()));
m_seekPos = 0;
return OK;
}
ReadableSampleFrames SoundSourceFLAC::readSampleFramesClamped(
WritableSampleFrames writableSampleFrames) {
const SINT firstFrameIndex = writableSampleFrames.frameIndexRange().start();
if (m_curFrameIndex != firstFrameIndex) {
// Seek to the new position
SINT seekFrameIndex = firstFrameIndex;
int retryCount = 0;
// NOTE(uklotzde): This loop avoids unnecessary seek operations.
// If the file is decoded from the beginning to the end during
// continuous playback no seek operations are necessary. This
// may hide rare seek errors that we have observed in some "flaky"
// FLAC files. The retry strategy implemented by this loop tries
// to solve these issues when randomly seeking through such a file.
while ((seekFrameIndex != m_curFrameIndex) &&
(retryCount <= kSeekErrorMaxRetryCount)) {
// Discard decoded sample data before seeking
m_sampleBuffer.clear();
invalidateCurFrameIndex();
if (FLAC__stream_decoder_seek_absolute(m_decoder, seekFrameIndex)) {
// Success: Set the new position
m_curFrameIndex = seekFrameIndex;
DEBUG_ASSERT(FLAC__STREAM_DECODER_SEEK_ERROR != FLAC__stream_decoder_get_state(m_decoder));
} else {
// Failure
kLogger.warning()
<< "Seek error at" << seekFrameIndex
<< "in file" << m_file.fileName();
if (FLAC__STREAM_DECODER_SEEK_ERROR == FLAC__stream_decoder_get_state(m_decoder)) {
// Flush the input stream of the decoder according to the
// documentation of FLAC__stream_decoder_seek_absolute()
if (!FLAC__stream_decoder_flush(m_decoder)) {
kLogger.warning()
<< "Failed to flush input buffer of the FLAC decoder after seek failure"
<< "in file" << m_file.fileName();
invalidateCurFrameIndex();
// ...and abort
return ReadableSampleFrames(
IndexRange::between(
m_curFrameIndex,
m_curFrameIndex));
}
}
if (frameIndexMin() < seekFrameIndex) {
// The next seek position should start at a preceding sample block.
// By subtracting max. blocksize from the current seek position it
// is guaranteed that the targeted sample blocks of subsequent seek
// operations will differ.
DEBUG_ASSERT(0 < m_maxBlocksize);
seekFrameIndex -= m_maxBlocksize;
if (seekFrameIndex < frameIndexMin()) {
seekFrameIndex = frameIndexMin();
}
} else {
// We have already reached the beginning of the file
// and cannot move the seek position backwards any
// further!
break; // exit loop
}
}
}
// Decoding starts before the actual target position
DEBUG_ASSERT(m_curFrameIndex <= firstFrameIndex);
const auto precedingFrames =
IndexRange::between(m_curFrameIndex, firstFrameIndex);
if (!precedingFrames.empty() &&
(precedingFrames != readSampleFramesClamped(WritableSampleFrames(precedingFrames)).frameIndexRange())) {
kLogger.warning()
<< "Failed to skip preceding frames"
<< precedingFrames;
// Abort
return ReadableSampleFrames(
IndexRange::between(
m_curFrameIndex,
m_curFrameIndex));
}
}
DEBUG_ASSERT(m_curFrameIndex == firstFrameIndex);
const SINT numberOfSamplesTotal = frames2samples(writableSampleFrames.frameLength());
SINT numberOfSamplesRemaining = numberOfSamplesTotal;
SINT outputSampleOffset = 0;
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.empty()) {
// 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)) {
kLogger.warning()
<< "Failed to decode FLAC file"
<< m_file.fileName();
break; // abort
}
// After decoding we might first need to skip some samples if the
// decoder complained that it has lost sync for some malformed(?)
// files
if (m_curFrameIndex != curFrameIndexBeforeProcessing) {
if (m_curFrameIndex < curFrameIndexBeforeProcessing) {
kLogger.warning()
<< "Trying to adjust frame index"
<< m_curFrameIndex << "<" << curFrameIndexBeforeProcessing
<< "while decoding FLAC file"
<< m_file.fileName();
const auto skipFrames =
IndexRange::between(m_curFrameIndex, curFrameIndexBeforeProcessing);
if (skipFrames != readSampleFramesClamped(WritableSampleFrames(skipFrames)).frameIndexRange()) {
kLogger.warning()
<< "Failed to skip sample frames"
<< skipFrames
<< "while decoding FLAC file"
<< m_file.fileName();
break; // abort
}
} else {
kLogger.warning()
<< "Unexpected frame index"
<< m_curFrameIndex << ">" << curFrameIndexBeforeProcessing
<< "while decoding FLAC file"
<< m_file.fileName();
break; // abort
}
}
DEBUG_ASSERT(curFrameIndexBeforeProcessing == m_curFrameIndex);
}
if (m_sampleBuffer.empty()) {
break; // EOF
}
const SINT numberOfSamplesRead =
std::min(m_sampleBuffer.readableLength(), numberOfSamplesRemaining);
const SampleBuffer::ReadableSlice readableSlice(
m_sampleBuffer.shrinkForReading(numberOfSamplesRead));
DEBUG_ASSERT(readableSlice.length() == numberOfSamplesRead);
if (writableSampleFrames.writableData()) {
SampleUtil::copy(
writableSampleFrames.writableData(outputSampleOffset),
readableSlice.data(),
readableSlice.length());
outputSampleOffset += numberOfSamplesRead;
}
m_curFrameIndex += samples2frames(numberOfSamplesRead);
numberOfSamplesRemaining -= numberOfSamplesRead;
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(numberOfSamplesTotal >= numberOfSamplesRemaining);
const SINT numberOfSamples = numberOfSamplesTotal - numberOfSamplesRemaining;
return ReadableSampleFrames(
IndexRange::forward(firstFrameIndex, samples2frames(numberOfSamples)),
SampleBuffer::ReadableSlice(
writableSampleFrames.writableData(),
std::min(writableSampleFrames.writableLength(), numberOfSamples)));
}
FLAC__StreamDecoderWriteStatus SoundSourceFLAC::flacWrite(
const FLAC__Frame* frame, const FLAC__int32* const buffer[]) {
const SINT numChannels = frame->header.channels;
if (channelCount() > numChannels) {
kLogger.warning()
<< "Corrupt or unsupported FLAC file:"
<< "Invalid number of channels in FLAC frame header"
<< frame->header.channels << "<>" << channelCount();
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
if (sampleRate() != SINT(frame->header.sample_rate)) {
kLogger.warning()
<< "Corrupt or unsupported FLAC file:"
<< "Invalid sample rate in FLAC frame header"
<< frame->header.sample_rate << "<>" << sampleRate();
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
const SINT numReadableFrames = frame->header.blocksize;
if (numReadableFrames > m_maxBlocksize) {
kLogger.warning()
<< "Corrupt or unsupported FLAC file:"
<< "Block size in FLAC frame header exceeds the maximum block size"
<< frame->header.blocksize << ">" << m_maxBlocksize;
return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;
}
// According to the API docs the decoder will always report the current
// position in "FLAC samples" (= "Mixxx frames") for convenience
DEBUG_ASSERT(frame->header.number_type == FLAC__FRAME_NUMBER_TYPE_SAMPLE_NUMBER);
m_curFrameIndex = frame->header.number.sample_number;
// Decode buffer should be empty before decoding the next frame
DEBUG_ASSERT(m_sampleBuffer.empty());
const SampleBuffer::WritableSlice writableSlice(
m_sampleBuffer.growForWriting(frames2samples(numReadableFrames)));
const SINT numWritableFrames = samples2frames(writableSlice.length());
DEBUG_ASSERT(numWritableFrames <= numReadableFrames);
if (numWritableFrames < numReadableFrames) {
kLogger.warning()
<< "Sample buffer has not enough free space for all decoded FLAC samples:"
<< numWritableFrames << "<" << numReadableFrames;
}
CSAMPLE* pSampleBuffer = writableSlice.data();
DEBUG_ASSERT(channelCount() <= numChannels);
switch (channelCount()) {
case 1: {
// optimized code for 1 channel (mono)
for (SINT i = 0; i < numWritableFrames; ++i) {
*pSampleBuffer++ = convertDecodedSample(buffer[0][i], m_bitsPerSample);
}
break;
}
case 2: {
// optimized code for 2 channels (stereo)
for (SINT i = 0; i < numWritableFrames; ++i) {
*pSampleBuffer++ = convertDecodedSample(buffer[0][i], m_bitsPerSample);
*pSampleBuffer++ = convertDecodedSample(buffer[1][i], m_bitsPerSample);
}
break;
}
default: {
// generic code for multiple channels
for (SINT i = 0; i < numWritableFrames; ++i) {
for (SINT j = 0; j < channelCount(); ++j) {
*pSampleBuffer++ = convertDecodedSample(buffer[j][i], m_bitsPerSample);
}
}
}
}
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
SINT SoundSourceM4A::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
const SINT numberOfFramesTotal = math_min(
numberOfFrames, getMaxFrameIndex() - m_curFrameIndex);
const SINT numberOfSamplesTotal = frames2samples(numberOfFramesTotal);
CSAMPLE* pSampleBuffer = sampleBuffer;
SINT numberOfSamplesRemaining = numberOfSamplesTotal;
while (0 < numberOfSamplesRemaining) {
if (!m_sampleBuffer.isEmpty()) {
// Consume previously decoded sample data
const SampleBuffer::ReadableChunk readableChunk(
m_sampleBuffer.readFromHead(numberOfSamplesRemaining));
if (pSampleBuffer) {
SampleUtil::copy(pSampleBuffer, readableChunk.data(), readableChunk.size());
pSampleBuffer += readableChunk.size();
}
m_curFrameIndex += samples2frames(readableChunk.size());
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(numberOfSamplesRemaining >= readableChunk.size());
numberOfSamplesRemaining -= readableChunk.size();
if (0 == numberOfSamplesRemaining) {
break; // exit loop
}
}
// All previously decoded sample data has been consumed now
DEBUG_ASSERT(m_sampleBuffer.isEmpty());
if (0 == m_inputBufferLength) {
// Fill input buffer from file
if (isValidSampleBlockId(m_curSampleBlockId)) {
// Read data for next sample block into input buffer
u_int8_t* pInputBuffer = &m_inputBuffer[0];
u_int32_t inputBufferLength = m_inputBuffer.size(); // in/out parameter
if (!MP4ReadSample(m_hFile, m_trackId, m_curSampleBlockId,
&pInputBuffer, &inputBufferLength,
NULL, NULL, NULL, NULL)) {
qWarning()
<< "Failed to read MP4 input data for sample block"
<< m_curSampleBlockId << "(" << "min ="
<< kSampleBlockIdMin << "," << "max ="
<< m_maxSampleBlockId << ")";
break; // abort
}
++m_curSampleBlockId;
m_inputBufferLength = inputBufferLength;
m_inputBufferOffset = 0;
}
}
DEBUG_ASSERT(0 <= m_inputBufferLength);
if (0 == m_inputBufferLength) {
break; // EOF
}
// NOTE(uklotzde): The sample buffer for NeAACDecDecode2 has to
// be big enough for a whole block of decoded samples, which
// contains up to kFramesPerSampleBlock frames. Otherwise
// we need to use a temporary buffer.
CSAMPLE* pDecodeBuffer; // in/out parameter
SINT decodeBufferCapacity;
const SINT decodeBufferCapacityMin = frames2samples(kFramesPerSampleBlock);
if (pSampleBuffer && (decodeBufferCapacityMin <= numberOfSamplesRemaining)) {
// Decode samples directly into sampleBuffer
pDecodeBuffer = pSampleBuffer;
decodeBufferCapacity = numberOfSamplesRemaining;
} else {
// Decode next sample block into temporary buffer
const SINT writeToTailCount = math_max(
numberOfSamplesRemaining, decodeBufferCapacityMin);
const SampleBuffer::WritableChunk writableChunk(
m_sampleBuffer.writeToTail(writeToTailCount));
pDecodeBuffer = writableChunk.data();
decodeBufferCapacity = writableChunk.size();
}
DEBUG_ASSERT(decodeBufferCapacityMin <= decodeBufferCapacity);
NeAACDecFrameInfo decFrameInfo;
void* pDecodeResult = NeAACDecDecode2(
m_hDecoder, &decFrameInfo,
&m_inputBuffer[m_inputBufferOffset],
m_inputBufferLength,
reinterpret_cast<void**>(&pDecodeBuffer),
decodeBufferCapacity * sizeof(*pDecodeBuffer));
// Verify the decoding result
if (0 != decFrameInfo.error) {
qWarning() << "AAC decoding error:"
<< decFrameInfo.error
<< NeAACDecGetErrorMessage(decFrameInfo.error)
<< getUrlString();
break; // abort
}
DEBUG_ASSERT(pDecodeResult == pDecodeBuffer); // verify the in/out parameter
// Verify the decoded sample data for consistency
if (getChannelCount() != decFrameInfo.channels) {
qWarning() << "Corrupt or unsupported AAC file:"
<< "Unexpected number of channels" << decFrameInfo.channels
<< "<>" << getChannelCount();
break; // abort
}
if (getFrameRate() != SINT(decFrameInfo.samplerate)) {
qWarning() << "Corrupt or unsupported AAC file:"
<< "Unexpected sample rate" << decFrameInfo.samplerate
<< "<>" << getFrameRate();
break; // abort
}
// Consume input data
m_inputBufferLength -= decFrameInfo.bytesconsumed;
m_inputBufferOffset += decFrameInfo.bytesconsumed;
// Consume decoded output data
const SINT numberOfSamplesDecoded = decFrameInfo.samples;
DEBUG_ASSERT(numberOfSamplesDecoded <= decodeBufferCapacity);
SINT numberOfSamplesRead;
if (pDecodeBuffer == pSampleBuffer) {
numberOfSamplesRead = math_min(numberOfSamplesDecoded, numberOfSamplesRemaining);
pSampleBuffer += numberOfSamplesRead;
} else {
m_sampleBuffer.readFromTail(decodeBufferCapacity - numberOfSamplesDecoded);
const SampleBuffer::ReadableChunk readableChunk(
m_sampleBuffer.readFromHead(numberOfSamplesRemaining));
numberOfSamplesRead = readableChunk.size();
if (pSampleBuffer) {
SampleUtil::copy(pSampleBuffer, readableChunk.data(), numberOfSamplesRead);
pSampleBuffer += numberOfSamplesRead;
}
}
// The decoder might decode more samples than actually needed
// at the end of the file! When the end of the file has been
// reached decoding can be restarted by seeking to a new
// position.
DEBUG_ASSERT(numberOfSamplesDecoded >= numberOfSamplesRead);
m_curFrameIndex += samples2frames(numberOfSamplesRead);
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(numberOfSamplesRemaining >= numberOfSamplesRead);
numberOfSamplesRemaining -= numberOfSamplesRead;
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(numberOfSamplesTotal >= numberOfSamplesRemaining);
return samples2frames(numberOfSamplesTotal - numberOfSamplesRemaining);
}
SINT SoundSourceMediaFoundation::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer) {
SINT numberOfFramesRemaining = numberOfFrames;
CSAMPLE* pSampleBuffer = sampleBuffer;
while (numberOfFramesRemaining > 0) {
SampleBuffer::ReadableChunk readableChunk(
m_sampleBuffer.readFromHead(
frames2samples(numberOfFramesRemaining)));
DEBUG_ASSERT(readableChunk.size()
<= frames2samples(numberOfFramesRemaining));
if (readableChunk.size() > 0) {
DEBUG_ASSERT(m_currentFrameIndex < getMaxFrameIndex());
if (sampleBuffer != nullptr) {
SampleUtil::copy(
pSampleBuffer,
readableChunk.data(),
readableChunk.size());
pSampleBuffer += readableChunk.size();
}
m_currentFrameIndex += samples2frames(readableChunk.size());
numberOfFramesRemaining -= samples2frames(readableChunk.size());
}
if (numberOfFramesRemaining == 0) {
break; // finished reading
}
// No more decoded sample frames available
DEBUG_ASSERT(m_sampleBuffer.isEmpty());
if (m_pSourceReader == nullptr) {
break; // abort if reader is dead
}
DWORD dwFlags = 0;
LONGLONG streamPos = 0;
IMFSample* pSample = nullptr;
HRESULT hrReadSample =
m_pSourceReader->ReadSample(
kStreamIndex, // [in] DWORD dwStreamIndex,
0, // [in] DWORD dwControlFlags,
nullptr, // [out] DWORD *pdwActualStreamIndex,
&dwFlags, // [out] DWORD *pdwStreamFlags,
&streamPos, // [out] LONGLONG *pllTimestamp,
&pSample); // [out] IMFSample **ppSample
if (FAILED(hrReadSample)) {
qWarning() << kLogPreamble
<< "IMFSourceReader::ReadSample() failed"
<< hrReadSample
<< "-> abort decoding";
DEBUG_ASSERT(pSample == nullptr);
break; // abort
}
if (dwFlags & MF_SOURCE_READERF_ERROR) {
qWarning() << kLogPreamble
<< "IMFSourceReader::ReadSample()"
<< "detected stream errors"
<< "(MF_SOURCE_READERF_ERROR)"
<< "-> abort and stop decoding";
DEBUG_ASSERT(pSample == nullptr);
safeRelease(&m_pSourceReader); // kill the reader
break; // abort
} else if (dwFlags & MF_SOURCE_READERF_ENDOFSTREAM) {
DEBUG_ASSERT(pSample == nullptr);
break; // finished reading
} else if (dwFlags & MF_SOURCE_READERF_CURRENTMEDIATYPECHANGED) {
qWarning() << kLogPreamble
<< "IMFSourceReader::ReadSample()"
<< "detected that the media type has changed"
<< "(MF_SOURCE_READERF_CURRENTMEDIATYPECHANGED)"
<< "-> abort decoding";
DEBUG_ASSERT(pSample == nullptr);
break; // abort
}
DEBUG_ASSERT(pSample != nullptr);
SINT readerFrameIndex = m_streamUnitConverter.toFrameIndex(streamPos);
DEBUG_ASSERT(
(m_currentFrameIndex == getMaxFrameIndex()) || // unknown position after seeking
(m_currentFrameIndex == readerFrameIndex));
m_currentFrameIndex = readerFrameIndex;
DWORD dwSampleBufferCount = 0;
HRESULT hrGetBufferCount =
pSample->GetBufferCount(&dwSampleBufferCount);
if (FAILED(hrGetBufferCount)) {
qWarning() << kLogPreamble
<< "IMFSample::GetBufferCount() failed"
<< hrGetBufferCount
<< "-> abort decoding";
safeRelease(&pSample);
break; // abort
}
DWORD dwSampleTotalLengthInBytes = 0;
HRESULT hrGetTotalLength = pSample->GetTotalLength(&dwSampleTotalLengthInBytes);
if (FAILED(hrGetTotalLength)) {
qWarning() << kLogPreamble
<< "IMFSample::GetTotalLength() failed"
<< hrGetTotalLength
<< "-> abort decoding";
safeRelease(&pSample);
break; // abort
}
// Enlarge temporary buffer (if necessary)
DEBUG_ASSERT((dwSampleTotalLengthInBytes % kBytesPerSample) == 0);
SINT numberOfSamplesToBuffer =
dwSampleTotalLengthInBytes / kBytesPerSample;
SINT sampleBufferCapacity = m_sampleBuffer.getCapacity();
DEBUG_ASSERT(sampleBufferCapacity > 0);
while (sampleBufferCapacity < numberOfSamplesToBuffer) {
sampleBufferCapacity *= 2;
}
if (m_sampleBuffer.getCapacity() < sampleBufferCapacity) {
qDebug() << kLogPreamble
<< "Enlarging sample buffer capacity"
<< m_sampleBuffer.getCapacity()
<< "->"
<< sampleBufferCapacity;
m_sampleBuffer.resetCapacity(sampleBufferCapacity);
}
DWORD dwSampleBufferIndex = 0;
while (dwSampleBufferIndex < dwSampleBufferCount) {
IMFMediaBuffer* pMediaBuffer = nullptr;
HRESULT hrGetBufferByIndex = pSample->GetBufferByIndex(dwSampleBufferIndex, &pMediaBuffer);
if (FAILED(hrGetBufferByIndex)) {
qWarning() << kLogPreamble
<< "IMFSample::GetBufferByIndex() failed"
<< hrGetBufferByIndex
<< "-> abort decoding";
DEBUG_ASSERT(pMediaBuffer == nullptr);
break; // prematurely exit buffer loop
}
CSAMPLE* pLockedSampleBuffer = nullptr;
DWORD lockedSampleBufferLengthInBytes = 0;
HRESULT hrLock = pMediaBuffer->Lock(
reinterpret_cast<quint8**>(&pLockedSampleBuffer),
nullptr,
&lockedSampleBufferLengthInBytes);
if (FAILED(hrLock)) {
qWarning() << kLogPreamble
<< "IMFMediaBuffer::Lock() failed"
<< hrLock
<< "-> abort decoding";
safeRelease(&pMediaBuffer);
break; // prematurely exit buffer loop
}
DEBUG_ASSERT((lockedSampleBufferLengthInBytes % sizeof(pLockedSampleBuffer[0])) == 0);
SINT lockedSampleBufferCount =
lockedSampleBufferLengthInBytes / sizeof(pLockedSampleBuffer[0]);
SINT copySamplesCount = std::min(
frames2samples(numberOfFramesRemaining),
lockedSampleBufferCount);
if (copySamplesCount > 0) {
// Copy samples directly into output buffer if possible
if (pSampleBuffer != nullptr) {
SampleUtil::copy(
pSampleBuffer,
pLockedSampleBuffer,
copySamplesCount);
pSampleBuffer += copySamplesCount;
}
pLockedSampleBuffer += copySamplesCount;
lockedSampleBufferCount -= copySamplesCount;
m_currentFrameIndex += samples2frames(copySamplesCount);
numberOfFramesRemaining -= samples2frames(copySamplesCount);
}
// Buffer the remaining samples
SampleBuffer::WritableChunk writableChunk(
m_sampleBuffer.writeToTail(lockedSampleBufferCount));
// The required capacity has been calculated in advance (see above)
DEBUG_ASSERT(writableChunk.size() == lockedSampleBufferCount);
SampleUtil::copy(
writableChunk.data(),
pLockedSampleBuffer,
writableChunk.size());
HRESULT hrUnlock = pMediaBuffer->Unlock();
VERIFY_OR_DEBUG_ASSERT(SUCCEEDED(hrUnlock)) {
qWarning() << kLogPreamble
<< "IMFMediaBuffer::Unlock() failed"
<< hrUnlock;
// ignore and continue
}
safeRelease(&pMediaBuffer);
++dwSampleBufferIndex;
}
safeRelease(&pSample);
if (dwSampleBufferIndex < dwSampleBufferCount) {
// Failed to read data from all buffers -> kill the reader
qWarning() << kLogPreamble
<< "Failed to read all buffered samples"
<< "-> abort and stop decoding";
safeRelease(&m_pSourceReader);
break; // abort
}
}
return numberOfFrames - numberOfFramesRemaining;
}
SINT SoundSourceMediaFoundation::seekSampleFrame(
SINT frameIndex) {
DEBUG_ASSERT(isValidFrameIndex(m_currentFrameIndex));
if (frameIndex >= getMaxFrameIndex()) {
// EOF
m_currentFrameIndex = getMaxFrameIndex();
return m_currentFrameIndex;
}
if (frameIndex > m_currentFrameIndex) {
// seeking forward
SINT skipFramesCount = frameIndex - m_currentFrameIndex;
// When to prefer skipping over seeking:
// 1) The sample buffer would be discarded before seeking anyway and
// skipping those already decoded samples effectively costs nothing
// 2) After seeking we need to decode at least kNumberOfPrefetchFrames
// before reaching the actual target position -> Only seek if we
// need to decode more than 2 * kNumberOfPrefetchFrames frames
// while skipping
SINT skipFramesCountMax =
samples2frames(m_sampleBuffer.getSize()) +
2 * kNumberOfPrefetchFrames;
if (skipFramesCount <= skipFramesCountMax) {
skipSampleFrames(skipFramesCount);
}
}
if (frameIndex == m_currentFrameIndex) {
return m_currentFrameIndex;
}
// Discard decoded samples
m_sampleBuffer.reset();
// Invalidate current position (end of stream)
m_currentFrameIndex = getMaxFrameIndex();
if (m_pSourceReader == nullptr) {
// reader is dead
return m_currentFrameIndex;
}
// Jump to a position before the actual seeking position.
// Prefetching a certain number of frames is necessary for
// sample accurate decoding. The decoder needs to decode
// some frames in advance to produce the same result at
// each position in the stream.
SINT seekIndex = std::max(SINT(frameIndex - kNumberOfPrefetchFrames), AudioSource::getMinFrameIndex());
LONGLONG seekPos = m_streamUnitConverter.fromFrameIndex(seekIndex);
DEBUG_ASSERT(seekPos >= 0);
PROPVARIANT prop;
HRESULT hrInitPropVariantFromInt64 =
InitPropVariantFromInt64(seekPos, &prop);
DEBUG_ASSERT(SUCCEEDED(hrInitPropVariantFromInt64)); // never fails
HRESULT hrSetCurrentPosition =
m_pSourceReader->SetCurrentPosition(GUID_NULL, prop);
PropVariantClear(&prop);
if (SUCCEEDED(hrSetCurrentPosition)) {
// NOTE(uklotzde): After SetCurrentPosition() the actual position
// of the stream is unknown until reading the next samples from
// the reader. Please note that the first sample decoded after
// SetCurrentPosition() may start BEFORE the actual target position.
// See also: https://msdn.microsoft.com/en-us/library/windows/desktop/dd374668(v=vs.85).aspx
// "The SetCurrentPosition method does not guarantee exact seeking." ...
// "After seeking, the application should call IMFSourceReader::ReadSample
// and advance to the desired position.
SINT skipFramesCount = frameIndex - seekIndex;
if (skipFramesCount > 0) {
// We need to fetch at least 1 sample from the reader to obtain the
// current position!
skipSampleFrames(skipFramesCount);
// Now m_currentFrameIndex reflects the actual position of the reader
if (m_currentFrameIndex < frameIndex) {
// Skip more samples if frameIndex has not yet been reached
skipSampleFrames(frameIndex - m_currentFrameIndex);
}
if (m_currentFrameIndex != frameIndex) {
qWarning() << kLogPreamble
<< "Seek to frame"
<< frameIndex
<< "failed";
// Jump to end of stream (= invalidate current position)
m_currentFrameIndex = getMaxFrameIndex();
}
} else {
// We are at the beginning of the stream and don't need
// to skip any frames. Calling IMFSourceReader::ReadSample
// is not necessary in this special case.
DEBUG_ASSERT(frameIndex == AudioSource::getMinFrameIndex());
m_currentFrameIndex = frameIndex;
}
} else {
qWarning() << kLogPreamble
<< "IMFSourceReader::SetCurrentPosition() failed"
<< hrSetCurrentPosition;
safeRelease(&m_pSourceReader); // kill the reader
}
return m_currentFrameIndex;
}
SINT SoundSourceMediaFoundation::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer) {
if (sDebug) {
qDebug() << "read()" << numberOfFrames;
}
SINT framesNeeded(numberOfFrames);
// first, copy frames from leftover buffer IF the leftover buffer is at
// the correct frame
if (m_leftoverBufferLength > 0 && m_leftoverBufferPosition == m_nextFrame) {
copyFrames(sampleBuffer, &framesNeeded, m_leftoverBuffer,
m_leftoverBufferLength);
if (m_leftoverBufferLength > 0) {
if (framesNeeded != 0) {
qWarning() << __FILE__ << __LINE__
<< "WARNING: Expected frames needed to be 0. Abandoning this file.";
m_dead = true;
}
m_leftoverBufferPosition += numberOfFrames;
}
} else {
// leftoverBuffer already empty or in the wrong position, clear it
m_leftoverBufferLength = 0;
}
while (!m_dead && framesNeeded > 0) {
HRESULT hr(S_OK);
DWORD dwFlags(0);
qint64 timestamp(0);
IMFSample *pSample(nullptr);
bool error(false); // set to true to break after releasing
hr = m_pReader->ReadSample(MF_SOURCE_READER_FIRST_AUDIO_STREAM, // [in] DWORD dwStreamIndex,
0, // [in] DWORD dwControlFlags,
nullptr, // [out] DWORD *pdwActualStreamIndex,
&dwFlags, // [out] DWORD *pdwStreamFlags,
×tamp, // [out] LONGLONG *pllTimestamp,
&pSample); // [out] IMFSample **ppSample
if (FAILED(hr)) {
qWarning() << "ReadSample failed!";
break; // abort
}
if (sDebug) {
qDebug() << "ReadSample timestamp:" << timestamp << "frame:"
<< frameFromMF(timestamp, getSamplingRate()) << "dwflags:" << dwFlags;
}
if (dwFlags & MF_SOURCE_READERF_ERROR) {
// our source reader is now dead, according to the docs
qWarning()
<< "SSMF: ReadSample set ERROR, SourceReader is now dead";
m_dead = true;
break;
} else if (dwFlags & MF_SOURCE_READERF_ENDOFSTREAM) {
qDebug() << "SSMF: End of input file.";
break;
} else if (dwFlags & MF_SOURCE_READERF_CURRENTMEDIATYPECHANGED) {
qWarning() << "SSMF: Type change";
break;
} else if (pSample == nullptr) {
// generally this will happen when dwFlags contains ENDOFSTREAM,
// so it'll be caught before now -bkgood
qWarning() << "SSMF: No sample";
continue;
} // we now own a ref to the instance at pSample
IMFMediaBuffer *pMBuffer(nullptr);
// I know this does at least a memcopy and maybe a malloc, if we have
// xrun issues with this we might want to look into using
// IMFSample::GetBufferByIndex (although MS doesn't recommend this)
if (FAILED(hr = pSample->ConvertToContiguousBuffer(&pMBuffer))) {
error = true;
goto releaseSample;
}
CSAMPLE *buffer(nullptr);
DWORD bufferLengthInBytes(0);
hr = pMBuffer->Lock(reinterpret_cast<quint8**>(&buffer), nullptr, &bufferLengthInBytes);
if (FAILED(hr)) {
error = true;
goto releaseMBuffer;
}
SINT bufferLength = samples2frames(bufferLengthInBytes / sizeof(buffer[0]));
if (m_seeking) {
qint64 bufferPosition(frameFromMF(timestamp, getSamplingRate()));
if (sDebug) {
qDebug() << "While seeking to " << m_nextFrame
<< "WMF put us at" << bufferPosition;
}
if (m_nextFrame < bufferPosition) {
// Uh oh. We are farther forward than our seek target. Emit
// silence? We can't seek backwards here.
CSAMPLE* pBufferCurpos = sampleBuffer
+ frames2samples(numberOfFrames - framesNeeded);
qint64 offshootFrames = bufferPosition - m_nextFrame;
// If we can correct this immediately, write zeros and adjust
// m_nextFrame to pretend it never happened.
if (offshootFrames <= framesNeeded) {
qWarning() << __FILE__ << __LINE__
<< "Working around inaccurate seeking. Writing silence for"
<< offshootFrames << "frames";
// Set offshootFrames samples to zero.
memset(pBufferCurpos, 0,
sizeof(*pBufferCurpos) * frames2samples(offshootFrames));
// Now m_nextFrame == bufferPosition
m_nextFrame += offshootFrames;
framesNeeded -= offshootFrames;
} else {
// It's more complicated. The buffer we have just decoded is
// more than framesNeeded frames away from us. It's too hard
// for us to handle this correctly currently, so let's just
// try to get on with our lives.
m_seeking = false;
m_nextFrame = bufferPosition;
qWarning() << __FILE__ << __LINE__
<< "Seek offshoot is too drastic. Cutting losses and pretending the current decoded audio buffer is the right seek point.";
}
}
if (m_nextFrame >= bufferPosition
&& m_nextFrame < bufferPosition + bufferLength) {
// m_nextFrame is in this buffer.
buffer += frames2samples(m_nextFrame - bufferPosition);
bufferLength -= m_nextFrame - bufferPosition;
m_seeking = false;
} else {
// we need to keep going forward
goto releaseRawBuffer;
}
}
// If the bufferLength is larger than the leftover buffer, re-allocate
// it with 2x the space.
if (frames2samples(bufferLength) > m_leftoverBufferSize) {
SINT newSize = m_leftoverBufferSize;
while (newSize < frames2samples(bufferLength)) {
newSize *= 2;
}
CSAMPLE* newBuffer = new CSAMPLE[newSize];
memcpy(newBuffer, m_leftoverBuffer,
sizeof(m_leftoverBuffer[0]) * m_leftoverBufferSize);
delete[] m_leftoverBuffer;
m_leftoverBuffer = newBuffer;
m_leftoverBufferSize = newSize;
}
copyFrames(
sampleBuffer + frames2samples(numberOfFrames - framesNeeded),
&framesNeeded,
buffer, bufferLength);
releaseRawBuffer: hr = pMBuffer->Unlock();
// I'm ignoring this, MSDN for IMFMediaBuffer::Unlock stipulates
// nothing about the state of the instance if this fails so might as
// well just let it be released.
//if (FAILED(hr)) break;
releaseMBuffer: safeRelease(&pMBuffer);
releaseSample: safeRelease(&pSample);
if (error)
break;
}
SINT framesRead = numberOfFrames - framesNeeded;
m_iCurrentPosition += framesRead;
m_nextFrame += framesRead;
if (m_leftoverBufferLength > 0) {
if (framesNeeded != 0) {
qWarning() << __FILE__ << __LINE__
<< "WARNING: Expected frames needed to be 0. Abandoning this file.";
m_dead = true;
}
m_leftoverBufferPosition = m_nextFrame;
}
if (sDebug) {
qDebug() << "read()" << numberOfFrames << "returning" << framesRead;
}
return framesRead;
}
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);
}
