Add support for WAV assets with arbitrary sample rate, not just 48khz. Fast path when already 24khz.

This commit is contained in:
Ken Cooke 2016-11-30 14:56:11 -08:00
parent 7ad384d731
commit 452f081366
2 changed files with 41 additions and 33 deletions

View file

@ -62,8 +62,10 @@ void Sound::downloadFinished(const QByteArray& data) {
QByteArray outputAudioByteArray;
interpretAsWav(rawAudioByteArray, outputAudioByteArray);
downSample(outputAudioByteArray);
int sampleRate = interpretAsWav(rawAudioByteArray, outputAudioByteArray);
if (sampleRate != 0) {
downSample(outputAudioByteArray, sampleRate);
}
} else if (fileName.endsWith(RAW_EXTENSION)) {
// check if this was a stereo raw file
// since it's raw the only way for us to know that is if the file was called .stereo.raw
@ -73,7 +75,7 @@ void Sound::downloadFinished(const QByteArray& data) {
}
// Process as RAW file
downSample(rawAudioByteArray);
downSample(rawAudioByteArray, 48000);
} else {
qCDebug(audio) << "Unknown sound file type";
}
@ -84,29 +86,35 @@ void Sound::downloadFinished(const QByteArray& data) {
emit ready();
}
void Sound::downSample(const QByteArray& rawAudioByteArray) {
// assume that this was a RAW file and is now an array of samples that are
// signed, 16-bit, 48Khz
void Sound::downSample(const QByteArray& rawAudioByteArray, int sampleRate) {
// we want to convert it to the format that the audio-mixer wants
// which is signed, 16-bit, 24Khz
int numChannels = _isStereo ? 2 : 1;
AudioSRC resampler(48000, AudioConstants::SAMPLE_RATE, numChannels);
if (sampleRate == AudioConstants::SAMPLE_RATE) {
// resize to max possible output
int numSourceFrames = rawAudioByteArray.size() / (numChannels * sizeof(AudioConstants::AudioSample));
int maxDestinationFrames = resampler.getMaxOutput(numSourceFrames);
int maxDestinationBytes = maxDestinationFrames * numChannels * sizeof(AudioConstants::AudioSample);
_byteArray.resize(maxDestinationBytes);
// no resampling needed
_byteArray = rawAudioByteArray;
int numDestinationFrames = resampler.render((int16_t*)rawAudioByteArray.data(),
(int16_t*)_byteArray.data(),
numSourceFrames);
} else {
// truncate to actual output
int numDestinationBytes = numDestinationFrames * numChannels * sizeof(AudioConstants::AudioSample);
_byteArray.resize(numDestinationBytes);
int numChannels = _isStereo ? 2 : 1;
AudioSRC resampler(sampleRate, AudioConstants::SAMPLE_RATE, numChannels);
// resize to max possible output
int numSourceFrames = rawAudioByteArray.size() / (numChannels * sizeof(AudioConstants::AudioSample));
int maxDestinationFrames = resampler.getMaxOutput(numSourceFrames);
int maxDestinationBytes = maxDestinationFrames * numChannels * sizeof(AudioConstants::AudioSample);
_byteArray.resize(maxDestinationBytes);
int numDestinationFrames = resampler.render((int16_t*)rawAudioByteArray.data(),
(int16_t*)_byteArray.data(),
numSourceFrames);
// truncate to actual output
int numDestinationBytes = numDestinationFrames * numChannels * sizeof(AudioConstants::AudioSample);
_byteArray.resize(numDestinationBytes);
}
}
//
@ -160,7 +168,8 @@ struct CombinedHeader {
WAVEHeader wave;
};
void Sound::interpretAsWav(const QByteArray& inputAudioByteArray, QByteArray& outputAudioByteArray) {
// returns wavfile sample rate, used for resampling
int Sound::interpretAsWav(const QByteArray& inputAudioByteArray, QByteArray& outputAudioByteArray) {
CombinedHeader fileHeader;
@ -174,36 +183,33 @@ void Sound::interpretAsWav(const QByteArray& inputAudioByteArray, QByteArray& ou
// descriptor.id == "RIFX" also signifies BigEndian file
// waveStream.setByteOrder(QDataStream::BigEndian);
qCDebug(audio) << "Currently not supporting big-endian audio files.";
return;
return 0;
}
if (strncmp(fileHeader.riff.type, "WAVE", 4) != 0
|| strncmp(fileHeader.wave.descriptor.id, "fmt", 3) != 0) {
qCDebug(audio) << "Not a WAVE Audio file.";
return;
return 0;
}
// added the endianess check as an extra level of security
if (qFromLittleEndian<quint16>(fileHeader.wave.audioFormat) != 1) {
qCDebug(audio) << "Currently not supporting non PCM audio files.";
return;
return 0;
}
if (qFromLittleEndian<quint16>(fileHeader.wave.numChannels) == 2) {
_isStereo = true;
} else if (qFromLittleEndian<quint16>(fileHeader.wave.numChannels) > 2) {
qCDebug(audio) << "Currently not support audio files with more than 2 channels.";
return 0;
}
if (qFromLittleEndian<quint16>(fileHeader.wave.bitsPerSample) != 16) {
qCDebug(audio) << "Currently not supporting non 16bit audio files.";
return;
return 0;
}
if (qFromLittleEndian<quint32>(fileHeader.wave.sampleRate) != 48000) {
qCDebug(audio) << "Currently not supporting non 48KHz audio files.";
return;
}
// Skip any extra data in the WAVE chunk
waveStream.skipRawData(fileHeader.wave.descriptor.size - (sizeof(WAVEHeader) - sizeof(chunk)));
@ -218,7 +224,7 @@ void Sound::interpretAsWav(const QByteArray& inputAudioByteArray, QByteArray& ou
waveStream.skipRawData(dataHeader.descriptor.size);
} else {
qCDebug(audio) << "Could not read wav audio data header.";
return;
return 0;
}
}
@ -227,12 +233,14 @@ void Sound::interpretAsWav(const QByteArray& inputAudioByteArray, QByteArray& ou
outputAudioByteArray.resize(outputAudioByteArraySize);
if (waveStream.readRawData(outputAudioByteArray.data(), outputAudioByteArraySize) != (int)outputAudioByteArraySize) {
qCDebug(audio) << "Error reading WAV file";
return 0;
}
_duration = (float) (outputAudioByteArraySize / (fileHeader.wave.sampleRate * fileHeader.wave.numChannels * fileHeader.wave.bitsPerSample / 8.0f));
return fileHeader.wave.sampleRate;
} else {
qCDebug(audio) << "Could not read wav audio file header.";
return;
return 0;
}
}

View file

@ -40,8 +40,8 @@ private:
bool _isReady;
float _duration; // In seconds
void downSample(const QByteArray& rawAudioByteArray);
void interpretAsWav(const QByteArray& inputAudioByteArray, QByteArray& outputAudioByteArray);
void downSample(const QByteArray& rawAudioByteArray, int sampleRate);
int interpretAsWav(const QByteArray& inputAudioByteArray, QByteArray& outputAudioByteArray);
virtual void downloadFinished(const QByteArray& data) override;
};