static void
validateChannels(const AudioSourceConfig &aConfig, const SenderConfig &rConfig)
{
using boost::lexical_cast;
using std::string;
if (rConfig.codec() == "mp3")
{
if (aConfig.numChannels() > 2)
throw std::runtime_error("MP3 only accepts 1 or 2 channels, not " +
lexical_cast<string>(aConfig.numChannels()));
}
}
SoundSource::OpenResult SoundSourceWV::tryOpen(const AudioSourceConfig& audioSrcCfg) {
DEBUG_ASSERT(!m_wpc);
char msg[80]; // hold possible error message
int openFlags = OPEN_WVC | OPEN_NORMALIZE;
if ((kChannelCountMono == audioSrcCfg.getChannelCount()) ||
(kChannelCountStereo == audioSrcCfg.getChannelCount())) {
openFlags |= OPEN_2CH_MAX;
}
// We use WavpackOpenFileInputEx to support Unicode paths on windows
// http://www.wavpack.com/lib_use.txt
QString wavPackFileName = getLocalFileName();
m_pWVFile = new QFile(wavPackFileName);
m_pWVFile->open(QFile::ReadOnly);
QString correctionFileName(wavPackFileName + "c");
if (QFile::exists(correctionFileName)) {
// If there is a correction file, open it as well
m_pWVCFile = new QFile(correctionFileName);
m_pWVCFile->open(QFile::ReadOnly);
}
m_wpc = WavpackOpenFileInputEx(&s_streamReader, m_pWVFile, m_pWVCFile,
msg, openFlags, 0);
if (!m_wpc) {
qDebug() << "SSWV::open: failed to open file : " << msg;
return OpenResult::FAILED;
}
setChannelCount(WavpackGetReducedChannels(m_wpc));
setSamplingRate(WavpackGetSampleRate(m_wpc));
setFrameCount(WavpackGetNumSamples(m_wpc));
if (WavpackGetMode(m_wpc) & MODE_FLOAT) {
m_sampleScaleFactor = CSAMPLE_PEAK;
} else {
const int bitsPerSample = WavpackGetBitsPerSample(m_wpc);
const uint32_t wavpackPeakSampleValue = 1u
<< (bitsPerSample - 1);
m_sampleScaleFactor = CSAMPLE_PEAK / wavpackPeakSampleValue;
}
return OpenResult::SUCCEEDED;
}
/** Cobbled together from:
http://msdn.microsoft.com/en-us/library/dd757929(v=vs.85).aspx
and http://msdn.microsoft.com/en-us/library/dd317928(VS.85).aspx
-- Albert
If anything in here fails, just bail. I'm not going to decode HRESULTS.
-- Bill
*/
bool SoundSourceMediaFoundation::configureAudioStream(const AudioSourceConfig& audioSrcCfg) {
HRESULT hr;
// deselect all streams, we only want the first
hr = m_pSourceReader->SetStreamSelection(
MF_SOURCE_READER_ALL_STREAMS, false);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to deselect all streams";
return false;
}
hr = m_pSourceReader->SetStreamSelection(
kStreamIndex, true);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to select first audio stream";
return false;
}
IMFMediaType* pAudioType = nullptr;
hr = m_pSourceReader->GetCurrentMediaType(
kStreamIndex, &pAudioType);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to get current media type from stream";
return false;
}
//------ Get bitrate from the file, before we change it to get uncompressed audio
UINT32 avgBytesPerSecond;
hr = pAudioType->GetUINT32(MF_MT_AUDIO_AVG_BYTES_PER_SECOND, &avgBytesPerSecond);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "error getting MF_MT_AUDIO_AVG_BYTES_PER_SECOND";
return false;
}
setBitrate( (avgBytesPerSecond * 8) / 1000);
//------
hr = pAudioType->SetGUID(MF_MT_MAJOR_TYPE, MFMediaType_Audio);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to set major type to audio";
safeRelease(&pAudioType);
return false;
}
hr = pAudioType->SetGUID(MF_MT_SUBTYPE, MFAudioFormat_Float);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to set subtype format to float";
safeRelease(&pAudioType);
return false;
}
hr = pAudioType->SetUINT32(MF_MT_ALL_SAMPLES_INDEPENDENT, true);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to set all samples independent";
safeRelease(&pAudioType);
return false;
}
hr = pAudioType->SetUINT32(MF_MT_FIXED_SIZE_SAMPLES, true);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to set fixed size samples";
safeRelease(&pAudioType);
return false;
}
hr = pAudioType->SetUINT32(
MF_MT_AUDIO_BITS_PER_SAMPLE, kBitsPerSample);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to set bits per sample:"
<< kBitsPerSample;
safeRelease(&pAudioType);
return false;
}
const UINT sampleSize = kLeftoverSize * kBytesPerSample;
hr = pAudioType->SetUINT32(
MF_MT_SAMPLE_SIZE, sampleSize);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to set sample size:"
<< sampleSize;
safeRelease(&pAudioType);
return false;
}
UINT32 numChannels;
hr = pAudioType->GetUINT32(
MF_MT_AUDIO_NUM_CHANNELS, &numChannels);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to get actual number of channels";
return false;
} else {
qDebug() << "Number of channels in input stream" << numChannels;
}
if (audioSrcCfg.hasValidChannelCount()) {
numChannels = audioSrcCfg.getChannelCount();
hr = pAudioType->SetUINT32(
MF_MT_AUDIO_NUM_CHANNELS, numChannels);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to set number of channels:"
<< numChannels;
safeRelease(&pAudioType);
return false;
}
qDebug() << "Requested number of channels" << numChannels;
}
UINT32 samplesPerSecond;
hr = pAudioType->GetUINT32(
MF_MT_AUDIO_SAMPLES_PER_SECOND, &samplesPerSecond);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to get samples per second";
return false;
} else {
qDebug() << "Samples per second in input stream" << samplesPerSecond;
}
if (audioSrcCfg.hasValidSamplingRate()) {
samplesPerSecond = audioSrcCfg.getSamplingRate();
hr = pAudioType->SetUINT32(
MF_MT_AUDIO_SAMPLES_PER_SECOND, samplesPerSecond);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to set samples per second:"
<< samplesPerSecond;
safeRelease(&pAudioType);
return false;
}
qDebug() << "Requested samples per second" << samplesPerSecond;
}
// Set this type on the source reader. The source reader will
// load the necessary decoder.
hr = m_pSourceReader->SetCurrentMediaType(
kStreamIndex, nullptr, pAudioType);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to set media type";
safeRelease(&pAudioType);
return false;
}
// Finally release the reference before reusing the pointer
safeRelease(&pAudioType);
// Get the resulting output format.
hr = m_pSourceReader->GetCurrentMediaType(
kStreamIndex, &pAudioType);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to retrieve completed media type";
return false;
}
// Ensure the stream is selected.
hr = m_pSourceReader->SetStreamSelection(
kStreamIndex, true);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to select first audio stream (again)";
return false;
}
hr = pAudioType->GetUINT32(
MF_MT_AUDIO_NUM_CHANNELS, &numChannels);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to get actual number of channels";
return false;
}
setChannelCount(numChannels);
hr = pAudioType->GetUINT32(
MF_MT_AUDIO_SAMPLES_PER_SECOND, &samplesPerSecond);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to get the actual sample rate";
return false;
}
setSamplingRate(samplesPerSecond);
UINT32 leftoverBufferSizeInBytes = 0;
hr = pAudioType->GetUINT32(MF_MT_SAMPLE_SIZE, &leftoverBufferSizeInBytes);
if (FAILED(hr)) {
qWarning() << kLogPreamble << hr
<< "failed to get sample buffer size (in bytes)";
return false;
}
DEBUG_ASSERT((leftoverBufferSizeInBytes % kBytesPerSample) == 0);
m_sampleBuffer.resetCapacity(leftoverBufferSizeInBytes / kBytesPerSample);
DEBUG_ASSERT(m_sampleBuffer.getCapacity() > 0);
qDebug() << kLogPreamble
<< "Sample buffer capacity"
<< m_sampleBuffer.getCapacity();
// Finally release the reference
safeRelease(&pAudioType);
return true;
}
// soundsource overrides
Result SoundSourceCoreAudio::tryOpen(const AudioSourceConfig& audioSrcCfg) {
const QString fileName(getLocalFileName());
//Open the audio file.
OSStatus err;
/** This code blocks works with OS X 10.5+ only. DO NOT DELETE IT for now. */
CFStringRef urlStr = CFStringCreateWithCharacters(0,
reinterpret_cast<const UniChar *>(fileName.unicode()),
fileName.size());
CFURLRef urlRef = CFURLCreateWithFileSystemPath(NULL, urlStr,
kCFURLPOSIXPathStyle, false);
err = ExtAudioFileOpenURL(urlRef, &m_audioFile);
CFRelease(urlStr);
CFRelease(urlRef);
/** TODO: Use FSRef for compatibility with 10.4 Tiger.
Note that ExtAudioFileOpen() is deprecated above Tiger, so we must maintain
both code paths if someone finishes this part of the code.
FSRef fsRef;
CFURLGetFSRef(reinterpret_cast<CFURLRef>(url.get()), &fsRef);
err = ExtAudioFileOpen(&fsRef, &m_audioFile);
*/
if (err != noErr) {
qDebug() << "SSCA: Error opening file " << fileName;
return ERR;
}
// get the input file format
UInt32 inputFormatSize = sizeof(m_inputFormat);
err = ExtAudioFileGetProperty(m_audioFile,
kExtAudioFileProperty_FileDataFormat, &inputFormatSize,
&m_inputFormat);
if (err != noErr) {
qDebug() << "SSCA: Error getting file format (" << fileName << ")";
return ERR;
}
m_bFileIsMp3 = m_inputFormat.mFormatID == kAudioFormatMPEGLayer3;
// create the output format
const UInt32 numChannels =
audioSrcCfg.hasChannelCount() ? audioSrcCfg.getChannelCount() : 2;
m_outputFormat = CAStreamBasicDescription(m_inputFormat.mSampleRate,
numChannels, CAStreamBasicDescription::kPCMFormatFloat32, true);
// set the client format
err = ExtAudioFileSetProperty(m_audioFile,
kExtAudioFileProperty_ClientDataFormat, sizeof(m_outputFormat),
&m_outputFormat);
if (err != noErr) {
qDebug() << "SSCA: Error setting file property";
return ERR;
}
//get the total length in frames of the audio file - copypasta: http://discussions.apple.com/thread.jspa?threadID=2364583&tstart=47
SInt64 totalFrameCount;
UInt32 totalFrameCountSize = sizeof(totalFrameCount);
err = ExtAudioFileGetProperty(m_audioFile,
kExtAudioFileProperty_FileLengthFrames, &totalFrameCountSize,
&totalFrameCount);
if (err != noErr) {
qDebug() << "SSCA: Error getting number of frames";
return ERR;
}
//
// WORKAROUND for bug in ExtFileAudio
//
AudioConverterRef acRef;
UInt32 acrsize = sizeof(AudioConverterRef);
err = ExtAudioFileGetProperty(m_audioFile,
kExtAudioFileProperty_AudioConverter, &acrsize, &acRef);
//_ThrowExceptionIfErr(@"kExtAudioFileProperty_AudioConverter", err);
AudioConverterPrimeInfo primeInfo;
UInt32 piSize = sizeof(AudioConverterPrimeInfo);
memset(&primeInfo, 0, piSize);
err = AudioConverterGetProperty(acRef, kAudioConverterPrimeInfo, &piSize,
&primeInfo);
if (err != kAudioConverterErr_PropertyNotSupported) { // Only if decompressing
//_ThrowExceptionIfErr(@"kAudioConverterPrimeInfo", err);
m_headerFrames = primeInfo.leadingFrames;
} else {
m_headerFrames = 0;
}
setChannelCount(m_outputFormat.NumberChannels());
setSamplingRate(m_inputFormat.mSampleRate);
// NOTE(uklotzde): This is what I found when migrating
// the code from SoundSource (sample-oriented) to the new
// AudioSource (frame-oriented) API. It is not documented
// when m_headerFrames > 0 and what the consequences are.
setFrameCount(totalFrameCount/* - m_headerFrames*/);
//Seek to position 0, which forces us to skip over all the header frames.
//This makes sure we're ready to just let the Analyser rip and it'll
//get the number of samples it expects (ie. no header frames).
seekSampleFrame(0);
return OK;
}
