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 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);
}
