SINT SoundSourceM4A::seekSampleFrame(SINT frameIndex) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
if (frameIndex >= getMaxFrameIndex()) {
// EOF
m_curFrameIndex = getMaxFrameIndex();
return m_curFrameIndex;
}
// NOTE(uklotzde): Resetting the decoder near to the beginning
// of the stream when seeking backwards produces invalid sample
// values! As a consequence the seeking test fails.
if (isValidSampleBlockId(m_curSampleBlockId) &&
(frameIndex <= kNumberOfPrefetchFrames)) {
// Workaround: Reset the decoder when seeking near to the beginning
// of the stream while decoding.
reopenDecoder();
skipSampleFrames(frameIndex);
}
if (frameIndex == m_curFrameIndex) {
return m_curFrameIndex;
}
MP4SampleId sampleBlockId = kSampleBlockIdMin
+ (frameIndex / m_framesPerSampleBlock);
DEBUG_ASSERT(isValidSampleBlockId(sampleBlockId));
if ((frameIndex < m_curFrameIndex) || // seeking backwards?
!isValidSampleBlockId(m_curSampleBlockId) || // invalid seek position?
(sampleBlockId
> (m_curSampleBlockId + m_numberOfPrefetchSampleBlocks))) { // jumping forward?
// Restart decoding one or more blocks of samples backwards
// from the calculated starting block to avoid audible glitches.
// Implementation note: The type MP4SampleId is unsigned so we
// need to be careful when subtracting!
if ((kSampleBlockIdMin + m_numberOfPrefetchSampleBlocks)
< sampleBlockId) {
sampleBlockId -= m_numberOfPrefetchSampleBlocks;
} else {
sampleBlockId = kSampleBlockIdMin;
}
restartDecoding(sampleBlockId);
DEBUG_ASSERT(m_curSampleBlockId == sampleBlockId);
}
// Decoding starts before the actual target position
DEBUG_ASSERT(m_curFrameIndex <= frameIndex);
// Skip (= decode and discard) all samples up to the target position
const SINT prefetchFrameCount = frameIndex - m_curFrameIndex;
const SINT skipFrameCount = skipSampleFrames(prefetchFrameCount);
DEBUG_ASSERT(skipFrameCount <= prefetchFrameCount);
if (skipFrameCount < prefetchFrameCount) {
qWarning() << "Failed to prefetch sample data while seeking"
<< skipFrameCount << "<" << prefetchFrameCount;
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
return m_curFrameIndex;
}
SINT SoundSourceFLAC::seekSampleFrame(SINT frameIndex) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(isValidFrameIndex(frameIndex));
// Avoid unnecessary seeking
// NOTE(uklotzde): Disabling this optimization might reveal rare
// seek errors on certain FLAC files were the decoder loses sync!
if (m_curFrameIndex == frameIndex) {
return m_curFrameIndex;
}
// Discard decoded sample data before seeking
m_sampleBuffer.reset();
// Seek to the new position
if (FLAC__stream_decoder_seek_absolute(m_decoder, frameIndex)) {
// Set the new position
m_curFrameIndex = frameIndex;
DEBUG_ASSERT(FLAC__STREAM_DECODER_SEEK_ERROR != FLAC__stream_decoder_get_state(m_decoder));
} else {
qWarning() << "Seek error at" << frameIndex << "in" << m_file.fileName();
// Invalidate the current position
m_curFrameIndex = getMaxFrameIndex();
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)) {
qWarning() << "Failed to flush input buffer of the FLAC decoder after seeking in"
<< m_file.fileName();
// Invalidate the current position...
m_curFrameIndex = getMaxFrameIndex();
// ...and abort
return m_curFrameIndex;
}
// Discard previously decoded sample data before decoding
// the next block of samples
m_sampleBuffer.reset();
// Trigger decoding of the next block to update the current position
if (!FLAC__stream_decoder_process_single(m_decoder)) {
qWarning() << "Failed to resync FLAC decoder after seeking in"
<< m_file.fileName();
// Invalidate the current position...
m_curFrameIndex = getMaxFrameIndex();
// ...and abort
return m_curFrameIndex;
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
if (m_curFrameIndex < frameIndex) {
// Adjust the current position
skipSampleFrames(frameIndex - m_curFrameIndex);
}
}
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
return m_curFrameIndex;
}
SINT SoundSourceM4A::seekSampleFrame(SINT frameIndex) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(isValidFrameIndex(frameIndex));
// Handle trivial case
if (m_curFrameIndex == frameIndex) {
// Nothing to do
return m_curFrameIndex;
}
// Handle edge case
if (getMaxFrameIndex() <= frameIndex) {
// EOF reached
m_curFrameIndex = getMaxFrameIndex();
return m_curFrameIndex;
}
MP4SampleId sampleBlockId = kSampleBlockIdMin
+ (frameIndex / m_framesPerSampleBlock);
DEBUG_ASSERT(isValidSampleBlockId(sampleBlockId));
if ((frameIndex < m_curFrameIndex) || // seeking backwards?
!isValidSampleBlockId(m_curSampleBlockId) || // invalid seek position?
(sampleBlockId
> (m_curSampleBlockId + m_numberOfPrefetchSampleBlocks))) { // jumping forward?
// Restart decoding one or more blocks of samples backwards
// from the calculated starting block to avoid audible glitches.
// Implementation note: The type MP4SampleId is unsigned so we
// need to be careful when subtracting!
if ((kSampleBlockIdMin + m_numberOfPrefetchSampleBlocks)
< sampleBlockId) {
sampleBlockId -= m_numberOfPrefetchSampleBlocks;
} else {
sampleBlockId = kSampleBlockIdMin;
}
restartDecoding(sampleBlockId);
DEBUG_ASSERT(m_curSampleBlockId == sampleBlockId);
}
// Decoding starts before the actual target position
DEBUG_ASSERT(m_curFrameIndex <= frameIndex);
// Skip (= decode and discard) all samples up to the target position
const SINT prefetchFrameCount = frameIndex - m_curFrameIndex;
const SINT skipFrameCount = skipSampleFrames(prefetchFrameCount);
DEBUG_ASSERT(skipFrameCount <= prefetchFrameCount);
if (skipFrameCount < prefetchFrameCount) {
qWarning() << "Failed to prefetch sample data while seeking"
<< skipFrameCount << "<" << prefetchFrameCount;
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
return m_curFrameIndex;
}
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 SoundSourceOpus::readSampleFrames(
SINT numberOfFrames, CSAMPLE* sampleBuffer) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
const SINT numberOfFramesTotal = math_min(
numberOfFrames, getMaxFrameIndex() - m_curFrameIndex);
CSAMPLE* pSampleBuffer = sampleBuffer;
SINT numberOfFramesRemaining = numberOfFramesTotal;
while (0 < numberOfFramesRemaining) {
int readResult = op_read_float(m_pOggOpusFile,
pSampleBuffer,
frames2samples(numberOfFramesRemaining), NULL);
if (0 < readResult) {
m_curFrameIndex += readResult;
pSampleBuffer += frames2samples(readResult);
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 SoundSourceWV::seekSampleFrame(SINT frameIndex) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
if (frameIndex >= getMaxFrameIndex()) {
// EOF reached
m_curFrameIndex = getMaxFrameIndex();
return m_curFrameIndex;
}
if (frameIndex == m_curFrameIndex) {
return m_curFrameIndex;
}
if (WavpackSeekSample(m_wpc, frameIndex) == true) {
m_curFrameIndex = frameIndex;
return frameIndex;
} else {
kLogger.debug() << "could not seek to frame #" << frameIndex;
return WavpackGetSampleIndex(m_wpc);
}
}
SINT SoundSourceMp3::findSeekFrameIndex(
SINT frameIndex) const {
// Check preconditions
DEBUG_ASSERT(0 < m_avgSeekFrameCount);
DEBUG_ASSERT(!m_seekFrameList.empty());
DEBUG_ASSERT(getMinFrameIndex() == m_seekFrameList.front().frameIndex);
DEBUG_ASSERT(getMaxFrameIndex() == m_seekFrameList.back().frameIndex);
SINT lowerBound =
0;
SINT upperBound =
m_seekFrameList.size();
DEBUG_ASSERT(lowerBound < upperBound);
// Initial guess based on average frame size
SINT seekFrameIndex =
frameIndex / m_avgSeekFrameCount;
if (seekFrameIndex >= upperBound) {
seekFrameIndex = upperBound - 1;
}
while ((upperBound - lowerBound) > 1) {
DEBUG_ASSERT(seekFrameIndex >= lowerBound);
DEBUG_ASSERT(seekFrameIndex < upperBound);
DEBUG_ASSERT(m_seekFrameList[lowerBound].frameIndex <= frameIndex);
if (m_seekFrameList[seekFrameIndex].frameIndex <= frameIndex) {
lowerBound = seekFrameIndex;
} else {
upperBound = seekFrameIndex;
}
// Next guess halfway between lower and upper bound
seekFrameIndex = lowerBound + (upperBound - lowerBound) / 2;
}
// Check postconditions
DEBUG_ASSERT(seekFrameIndex == lowerBound);
DEBUG_ASSERT(SINT(m_seekFrameList.size()) > seekFrameIndex);
DEBUG_ASSERT(m_seekFrameList[seekFrameIndex].frameIndex <= frameIndex);
DEBUG_ASSERT(((seekFrameIndex + 1) >= SINT(m_seekFrameList.size())) ||
(m_seekFrameList[seekFrameIndex + 1].frameIndex > frameIndex));
return seekFrameIndex;
}
SINT SoundSourceOpus::seekSampleFrame(SINT frameIndex) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(isValidFrameIndex(frameIndex));
int seekResult = op_pcm_seek(m_pOggOpusFile, frameIndex);
if (0 == seekResult) {
m_curFrameIndex = frameIndex;
} else {
qWarning() << "Failed to seek OggOpus file:" << seekResult;
const ogg_int64_t pcmOffset = op_pcm_tell(m_pOggOpusFile);
if (0 <= pcmOffset) {
m_curFrameIndex = pcmOffset;
} else {
// Reset to EOF
m_curFrameIndex = getMaxFrameIndex();
}
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
return m_curFrameIndex;
}
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);
}
Result SoundSourceM4A::tryOpen(const AudioSourceConfig& audioSrcCfg) {
DEBUG_ASSERT(MP4_INVALID_FILE_HANDLE == m_hFile);
/* open MP4 file, check for >= ver 1.9.1 */
#if MP4V2_PROJECT_version_hex <= 0x00010901
m_hFile = MP4Read(getLocalFileNameBytes().constData(), 0);
#else
m_hFile = MP4Read(getLocalFileNameBytes().constData());
#endif
if (MP4_INVALID_FILE_HANDLE == m_hFile) {
qWarning() << "Failed to open file for reading:" << getUrlString();
return ERR;
}
m_trackId = findFirstAudioTrackId(m_hFile);
if (MP4_INVALID_TRACK_ID == m_trackId) {
qWarning() << "No AAC track found:" << getUrlString();
return ERR;
}
const MP4SampleId numberOfSamples =
MP4GetTrackNumberOfSamples(m_hFile, m_trackId);
if (0 >= numberOfSamples) {
qWarning() << "Failed to read number of samples from file:" << getUrlString();
return ERR;
}
m_maxSampleBlockId = kSampleBlockIdMin + (numberOfSamples - 1);
// Determine the maximum input size (in bytes) of a
// sample block for the selected track.
const u_int32_t maxSampleBlockInputSize = MP4GetTrackMaxSampleSize(m_hFile,
m_trackId);
m_inputBuffer.resize(maxSampleBlockInputSize, 0);
DEBUG_ASSERT(NULL == m_hDecoder); // not already opened
m_hDecoder = NeAACDecOpen();
if (!m_hDecoder) {
qWarning() << "Failed to open the AAC decoder!";
return ERR;
}
NeAACDecConfigurationPtr pDecoderConfig = NeAACDecGetCurrentConfiguration(
m_hDecoder);
pDecoderConfig->outputFormat = FAAD_FMT_FLOAT;
if ((kChannelCountMono == audioSrcCfg.channelCountHint) ||
(kChannelCountStereo == audioSrcCfg.channelCountHint)) {
pDecoderConfig->downMatrix = 1;
} else {
pDecoderConfig->downMatrix = 0;
}
pDecoderConfig->defObjectType = LC;
if (!NeAACDecSetConfiguration(m_hDecoder, pDecoderConfig)) {
qWarning() << "Failed to configure AAC decoder!";
return ERR;
}
u_int8_t* configBuffer = NULL;
u_int32_t configBufferSize = 0;
if (!MP4GetTrackESConfiguration(m_hFile, m_trackId, &configBuffer,
&configBufferSize)) {
/* failed to get mpeg-4 audio config... this is ok.
* NeAACDecInit2() will simply use default values instead.
*/
qWarning() << "Failed to read the MP4 audio configuration."
<< "Continuing with default values.";
}
SAMPLERATE_TYPE sampleRate;
unsigned char channelCount;
if (0 > NeAACDecInit2(m_hDecoder, configBuffer, configBufferSize,
&sampleRate, &channelCount)) {
free(configBuffer);
qWarning() << "Failed to initialize the AAC decoder!";
return ERR;
} else {
free(configBuffer);
}
setChannelCount(channelCount);
setFrameRate(sampleRate);
setFrameCount(getFrameCountForSampleBlockId(m_maxSampleBlockId));
// Resize temporary buffer for decoded sample data
const SINT sampleBufferCapacity =
frames2samples(kFramesPerSampleBlock);
m_sampleBuffer.resetCapacity(sampleBufferCapacity);
// Invalidate current position to enforce the following
// seek operation
m_curFrameIndex = getMaxFrameIndex();
// (Re-)Start decoding at the beginning of the file
seekSampleFrame(getMinFrameIndex());
return OK;
}
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;
}
Result SoundSourceM4A::tryOpen(const AudioSourceConfig& audioSrcCfg) {
DEBUG_ASSERT(MP4_INVALID_FILE_HANDLE == m_hFile);
// open MP4 file, check for >= ver 1.9.1
// From mp4v2/file.h:
// * On Windows, this should be a UTF-8 encoded string.
// * On other platforms, it should be an 8-bit encoding that is
// * appropriate for the platform, locale, file system, etc.
// * (prefer to use UTF-8 when possible).
#if MP4V2_PROJECT_version_hex <= 0x00010901
m_hFile = MP4Read(getLocalFileName().toUtf8().constData(), 0);
#else
m_hFile = MP4Read(getLocalFileName().toUtf8().constData());
#endif
if (MP4_INVALID_FILE_HANDLE == m_hFile) {
qWarning() << "Failed to open file for reading:" << getUrlString();
return ERR;
}
m_trackId = findFirstAudioTrackId(m_hFile);
if (MP4_INVALID_TRACK_ID == m_trackId) {
qWarning() << "No AAC track found:" << getUrlString();
return ERR;
}
// Read fixed sample duration. If the sample duration is not
// fixed (that is, if the number of frames per sample block varies
// through the file), the call returns MP4_INVALID_DURATION. We
// can't currently handle these.
m_framesPerSampleBlock = MP4GetTrackFixedSampleDuration(m_hFile, m_trackId);
if (MP4_INVALID_DURATION == m_framesPerSampleBlock) {
qWarning() << "Unable to decode tracks with non-fixed sample durations: " << getUrlString();
return ERR;
}
const MP4SampleId numberOfSamples =
MP4GetTrackNumberOfSamples(m_hFile, m_trackId);
if (0 >= numberOfSamples) {
qWarning() << "Failed to read number of samples from file:" << getUrlString();
return ERR;
}
m_maxSampleBlockId = kSampleBlockIdMin + (numberOfSamples - 1);
// Determine the maximum input size (in bytes) of a
// sample block for the selected track.
const u_int32_t maxSampleBlockInputSize = MP4GetTrackMaxSampleSize(m_hFile,
m_trackId);
m_inputBuffer.resize(maxSampleBlockInputSize, 0);
DEBUG_ASSERT(nullptr == m_hDecoder); // not already opened
m_hDecoder = NeAACDecOpen();
if (!m_hDecoder) {
qWarning() << "Failed to open the AAC decoder!";
return ERR;
}
NeAACDecConfigurationPtr pDecoderConfig = NeAACDecGetCurrentConfiguration(
m_hDecoder);
pDecoderConfig->outputFormat = FAAD_FMT_FLOAT;
if ((kChannelCountMono == audioSrcCfg.channelCountHint) ||
(kChannelCountStereo == audioSrcCfg.channelCountHint)) {
pDecoderConfig->downMatrix = 1;
} else {
pDecoderConfig->downMatrix = 0;
}
pDecoderConfig->defObjectType = LC;
if (!NeAACDecSetConfiguration(m_hDecoder, pDecoderConfig)) {
qWarning() << "Failed to configure AAC decoder!";
return ERR;
}
u_int8_t* configBuffer = nullptr;
u_int32_t configBufferSize = 0;
if (!MP4GetTrackESConfiguration(m_hFile, m_trackId, &configBuffer,
&configBufferSize)) {
/* failed to get mpeg-4 audio config... this is ok.
* NeAACDecInit2() will simply use default values instead.
*/
qWarning() << "Failed to read the MP4 audio configuration."
<< "Continuing with default values.";
}
SAMPLERATE_TYPE sampleRate;
unsigned char channelCount;
if (0 > NeAACDecInit2(m_hDecoder, configBuffer, configBufferSize,
&sampleRate, &channelCount)) {
free(configBuffer);
qWarning() << "Failed to initialize the AAC decoder!";
return ERR;
} else {
free(configBuffer);
}
// Calculate how many sample blocks we need to decode in advance
// of a random seek in order to get the recommended number of
// prefetch frames
m_numberOfPrefetchSampleBlocks = (kNumberOfPrefetchFrames +
(m_framesPerSampleBlock - 1)) / m_framesPerSampleBlock;
setChannelCount(channelCount);
setFrameRate(sampleRate);
setFrameCount(((m_maxSampleBlockId - kSampleBlockIdMin) + 1) * m_framesPerSampleBlock);
// Resize temporary buffer for decoded sample data
const SINT sampleBufferCapacity =
frames2samples(m_framesPerSampleBlock);
m_sampleBuffer.resetCapacity(sampleBufferCapacity);
// Invalidate current position to enforce the following
// seek operation
m_curFrameIndex = getMaxFrameIndex();
// (Re-)Start decoding at the beginning of the file
seekSampleFrame(getMinFrameIndex());
return OK;
}
SINT SoundSourceFLAC::seekSampleFrame(SINT frameIndex) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(isValidFrameIndex(frameIndex));
// Seek to the new position
SINT seekFrameIndex = frameIndex;
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.reset();
// Invalidate the current position
m_curFrameIndex = getMaxFrameIndex();
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
qWarning() << "Seek error at" << seekFrameIndex << "in" << 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)) {
qWarning() << "Failed to flush input buffer of the FLAC decoder after seek failure in"
<< m_file.fileName();
// Invalidate the current position again...
m_curFrameIndex = getMaxFrameIndex();
// ...and abort
return m_curFrameIndex;
}
}
if (getMinFrameIndex() < 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 < getMinFrameIndex()) {
seekFrameIndex = getMinFrameIndex();
}
} else {
// We have already reached the beginning of the file
// and cannot move the seek position backward any
// further!
break; // exit loop
}
}
} while (m_curFrameIndex != seekFrameIndex);
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
if (frameIndex > m_curFrameIndex) {
// Adjust the current position
skipSampleFrames(frameIndex - m_curFrameIndex);
}
return m_curFrameIndex;
}
bool SoundSourceM4A::openDecoder() {
DEBUG_ASSERT(m_hDecoder == nullptr); // not already opened
m_hDecoder = NeAACDecOpen();
if (m_hDecoder == nullptr) {
qWarning() << "Failed to open the AAC decoder!";
return false;
}
NeAACDecConfigurationPtr pDecoderConfig = NeAACDecGetCurrentConfiguration(
m_hDecoder);
pDecoderConfig->outputFormat = FAAD_FMT_FLOAT;
if ((kChannelCountMono == m_audioSrcCfg.getChannelCount()) ||
(kChannelCountStereo == m_audioSrcCfg.getChannelCount())) {
pDecoderConfig->downMatrix = 1;
} else {
pDecoderConfig->downMatrix = 0;
}
pDecoderConfig->defObjectType = LC;
if (!NeAACDecSetConfiguration(m_hDecoder, pDecoderConfig)) {
qWarning() << "Failed to configure AAC decoder!";
return false;
}
u_int8_t* configBuffer = nullptr;
u_int32_t configBufferSize = 0;
if (!MP4GetTrackESConfiguration(m_hFile, m_trackId, &configBuffer,
&configBufferSize)) {
// Failed to get mpeg-4 audio config... this is ok.
// NeAACDecInit2() will simply use default values instead.
qWarning() << "Failed to read the MP4 audio configuration."
<< "Continuing with default values.";
}
SAMPLERATE_TYPE samplingRate;
unsigned char channelCount;
if (0 > NeAACDecInit2(m_hDecoder, configBuffer, configBufferSize,
&samplingRate, &channelCount)) {
free(configBuffer);
qWarning() << "Failed to initialize the AAC decoder!";
return false;
} else {
free(configBuffer);
}
// Calculate how many sample blocks we need to decode in advance
// of a random seek in order to get the recommended number of
// prefetch frames
m_numberOfPrefetchSampleBlocks =
(kNumberOfPrefetchFrames + (m_framesPerSampleBlock - 1)) /
m_framesPerSampleBlock;
setChannelCount(channelCount);
setSamplingRate(samplingRate);
setFrameCount(((m_maxSampleBlockId - kSampleBlockIdMin) + 1) * m_framesPerSampleBlock);
// Resize temporary buffer for decoded sample data
const SINT sampleBufferCapacity =
frames2samples(m_framesPerSampleBlock);
m_sampleBuffer.resetCapacity(sampleBufferCapacity);
// Discard all buffered samples
m_inputBufferLength = 0;
// Invalidate current position(s) for the following seek operation
m_curSampleBlockId = MP4_INVALID_SAMPLE_ID;
m_curFrameIndex = getMaxFrameIndex();
// (Re-)Start decoding at the beginning of the file
seekSampleFrame(getMinFrameIndex());
return m_curFrameIndex == getMinFrameIndex();
}
SINT SoundSourceMp3::seekSampleFrame(SINT frameIndex) {
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
DEBUG_ASSERT(isValidFrameIndex(frameIndex));
// Handle trivial case
if (m_curFrameIndex == frameIndex) {
// Nothing to do
return m_curFrameIndex;
}
// Handle edge case
if (getMaxFrameIndex() <= frameIndex) {
// EOF reached
m_curFrameIndex = getMaxFrameIndex();
return m_curFrameIndex;
}
SINT seekFrameIndex = findSeekFrameIndex(
frameIndex);
DEBUG_ASSERT(SINT(m_seekFrameList.size()) > seekFrameIndex);
const SINT curSeekFrameIndex = findSeekFrameIndex(
m_curFrameIndex);
DEBUG_ASSERT(SINT(m_seekFrameList.size()) > curSeekFrameIndex);
// some consistency checks
DEBUG_ASSERT((curSeekFrameIndex >= seekFrameIndex) || (m_curFrameIndex < frameIndex));
DEBUG_ASSERT((curSeekFrameIndex <= seekFrameIndex) || (m_curFrameIndex > frameIndex));
if ((getMaxFrameIndex() <= m_curFrameIndex) || // out of range
(frameIndex < m_curFrameIndex) || // seek backward
(seekFrameIndex > (curSeekFrameIndex + kMp3SeekFramePrefetchCount))) { // jump forward
// Adjust the seek frame index for prefetching
// Implementation note: The type SINT is unsigned so
// need to be careful when subtracting!
if (kMp3SeekFramePrefetchCount < seekFrameIndex) {
// Restart decoding kMp3SeekFramePrefetchCount seek frames
// before the expected sync position
seekFrameIndex -= kMp3SeekFramePrefetchCount;
} else {
// Restart decoding at the beginning of the audio stream
seekFrameIndex = 0;
}
m_curFrameIndex = restartDecoding(m_seekFrameList[seekFrameIndex]);
if (getMaxFrameIndex() <= m_curFrameIndex) {
// out of range -> abort
return m_curFrameIndex;
}
DEBUG_ASSERT(findSeekFrameIndex(m_curFrameIndex) == seekFrameIndex);
}
// Decoding starts before the actual target position
DEBUG_ASSERT(m_curFrameIndex <= frameIndex);
// Skip (= decode and discard) all samples up to the target position
const SINT prefetchFrameCount = frameIndex - m_curFrameIndex;
const SINT skipFrameCount = skipSampleFrames(prefetchFrameCount);
DEBUG_ASSERT(skipFrameCount <= prefetchFrameCount);
if (skipFrameCount < prefetchFrameCount) {
qWarning() << "Failed to prefetch sample data while seeking"
<< skipFrameCount << "<" << prefetchFrameCount;
}
DEBUG_ASSERT(isValidFrameIndex(m_curFrameIndex));
return m_curFrameIndex;
}
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);
}
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;
}
int LayerImage::getMaxFramePosition()
{
return getFramePositionAt(getMaxFrameIndex());
}
