kopia lustrzana https://gitlab.com/eliggett/wfview
Fix if 0 samples detected
Phil Taylor
rodzic
7098564121
commit
3e0c008144
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@ -84,7 +84,7 @@ bool audioConverter::convert(audioPacket audio)
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{
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// If inFormat and outFormat are identical, just emit the data back.
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if (audio.data.size() != 0 && inFormat != outFormat)
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if (audio.data.size() > 0 && inFormat != outFormat)
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{
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if (inFormat.codec() == "audio/opus")
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@ -154,150 +154,150 @@ bool audioConverter::convert(audioPacket audio)
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qInfo(logAudio()) << "Unsupported Sample Type:" << inFormat.sampleType() << "Size:" << inFormat.sampleSize();
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}
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if (samplesF.size() > 0)
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audio.amplitude = samplesF.array().abs().maxCoeff();
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// Set the volume
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samplesF *= audio.volume;
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/*
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samplesF is now an Eigen Vector of the current samples in float format
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The next step is to convert to the correct number of channels in outFormat.channelCount()
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*/
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if (inFormat.channelCount() == 2 && outFormat.channelCount() == 1) {
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// If we need to drop one of the audio channels, do it now
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Eigen::VectorXf samplesTemp(samplesF.size() / 2);
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samplesTemp = Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesF.data(), samplesF.size() / 2);
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samplesF = samplesTemp;
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}
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else if (inFormat.channelCount() == 1 && outFormat.channelCount() == 2) {
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// Convert mono to stereo if required
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Eigen::VectorXf samplesTemp(samplesF.size() * 2);
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Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data(), samplesF.size()) = samplesF;
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Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data() + 1, samplesF.size()) = samplesF;
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samplesF = samplesTemp;
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}
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/*
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Next step is to resample (if needed)
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*/
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if (resampler != Q_NULLPTR && resampleRatio != 1.0)
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{
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quint32 outFrames = ((samplesF.size() / outFormat.channelCount()) * resampleRatio);
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quint32 inFrames = (samplesF.size() / outFormat.channelCount());
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QByteArray outPacket(outFrames * outFormat.channelCount() * sizeof(float), (char)0xff); // Preset the output buffer size.
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const float* in = (float*)samplesF.data();
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float* out = (float*)outPacket.data();
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int err = 0;
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if (outFormat.channelCount() == 1) {
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err = wf_resampler_process_float(resampler, 0, in, &inFrames, out, &outFrames);
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}
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else {
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err = wf_resampler_process_interleaved_float(resampler, in, &inFrames, out, &outFrames);
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}
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if (err) {
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qInfo(logAudioConverter()) << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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}
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samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
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}
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/*
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If output is Opus so encode it now, don't do any more conversion on the output of Opus.
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*/
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if (outFormat.codec() == "audio/opus")
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{
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float* in = (float*)samplesF.data();
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QByteArray outPacket(1275, (char)0xff); // Preset the output buffer size to MAXIMUM possible Opus frame size
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unsigned char* out = (unsigned char*)outPacket.data();
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int nbBytes = opus_encode_float(opusEncoder, in, (samplesF.size() / outFormat.channelCount()), out, outPacket.length());
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if (nbBytes < 0)
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{
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qInfo(logAudioConverter()) << "Opus encode failed:" << opus_strerror(nbBytes) << "Num Samples:" << samplesF.size();
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return false;
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}
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else {
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outPacket.resize(nbBytes);
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audio.data.clear();
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audio.data = outPacket; // Copy output packet back to input buffer.
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//samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
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}
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}
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else {
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audio.amplitude = samplesF.array().abs().maxCoeff();
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// Set the volume
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samplesF *= audio.volume;
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/*
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Now convert back into the output format required
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samplesF is now an Eigen Vector of the current samples in float format
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The next step is to convert to the correct number of channels in outFormat.channelCount()
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*/
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audio.data.clear();
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if (outFormat.sampleType() == QAudioFormat::UnSignedInt && outFormat.sampleSize() == 8)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<quint8>::max());
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VectorXuint8 samplesI = samplesITemp.cast<quint8>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(quint8)));
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if (inFormat.channelCount() == 2 && outFormat.channelCount() == 1) {
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// If we need to drop one of the audio channels, do it now
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Eigen::VectorXf samplesTemp(samplesF.size() / 2);
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samplesTemp = Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesF.data(), samplesF.size() / 2);
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samplesF = samplesTemp;
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}
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if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 8)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint8>::max());
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VectorXint8 samplesI = samplesITemp.cast<qint8>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint8)));
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}
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if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 16)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint16>::max());
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VectorXint16 samplesI = samplesITemp.cast<qint16>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint16)));
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}
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else if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 32)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint32>::max());
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VectorXint32 samplesI = samplesITemp.cast<qint32>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint32)));
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}
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else if (outFormat.sampleType() == QAudioFormat::Float)
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{
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audio.data = QByteArray(reinterpret_cast<char*>(samplesF.data()), int(samplesF.size()) * int(sizeof(float)));
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}
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else {
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qInfo(logAudio()) << "Unsupported Sample Type:" << outFormat.sampleType();
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else if (inFormat.channelCount() == 1 && outFormat.channelCount() == 2) {
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// Convert mono to stereo if required
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Eigen::VectorXf samplesTemp(samplesF.size() * 2);
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Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data(), samplesF.size()) = samplesF;
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Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data() + 1, samplesF.size()) = samplesF;
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samplesF = samplesTemp;
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}
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/*
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As we currently don't have a float based uLaw encoder, this must be done
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after all other conversion has taken place.
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Next step is to resample (if needed)
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*/
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if (outFormat.codec() == "audio/PCMU")
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if (resampler != Q_NULLPTR && resampleRatio != 1.0)
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{
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QByteArray outPacket((int)audio.data.length() / 2, (char)0xff);
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qint16* in = (qint16*)audio.data.data();
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for (int f = 0; f < outPacket.length(); f++)
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{
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qint16 sample = *in++;
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int sign = (sample >> 8) & 0x80;
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if (sign)
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sample = (short)-sample;
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if (sample > cClip)
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sample = cClip;
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sample = (short)(sample + cBias);
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int exponent = (int)MuLawCompressTable[(sample >> 7) & 0xFF];
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int mantissa = (sample >> (exponent + 3)) & 0x0F;
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int compressedByte = ~(sign | (exponent << 4) | mantissa);
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outPacket[f] = (quint8)compressedByte;
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quint32 outFrames = ((samplesF.size() / outFormat.channelCount()) * resampleRatio);
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quint32 inFrames = (samplesF.size() / outFormat.channelCount());
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QByteArray outPacket(outFrames * outFormat.channelCount() * sizeof(float), (char)0xff); // Preset the output buffer size.
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const float* in = (float*)samplesF.data();
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float* out = (float*)outPacket.data();
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int err = 0;
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if (outFormat.channelCount() == 1) {
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err = wf_resampler_process_float(resampler, 0, in, &inFrames, out, &outFrames);
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}
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else {
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err = wf_resampler_process_interleaved_float(resampler, in, &inFrames, out, &outFrames);
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}
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audio.data.clear();
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audio.data = outPacket; // Copy output packet back to input buffer.
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if (err) {
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qInfo(logAudioConverter()) << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
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}
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samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
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}
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/*
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If output is Opus so encode it now, don't do any more conversion on the output of Opus.
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*/
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if (outFormat.codec() == "audio/opus")
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{
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float* in = (float*)samplesF.data();
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QByteArray outPacket(1275, (char)0xff); // Preset the output buffer size to MAXIMUM possible Opus frame size
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unsigned char* out = (unsigned char*)outPacket.data();
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int nbBytes = opus_encode_float(opusEncoder, in, (samplesF.size() / outFormat.channelCount()), out, outPacket.length());
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if (nbBytes < 0)
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{
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qInfo(logAudioConverter()) << "Opus encode failed:" << opus_strerror(nbBytes) << "Num Samples:" << samplesF.size();
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return false;
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}
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else {
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outPacket.resize(nbBytes);
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audio.data.clear();
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audio.data = outPacket; // Copy output packet back to input buffer.
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//samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
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}
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}
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else {
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/*
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Now convert back into the output format required
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*/
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audio.data.clear();
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if (outFormat.sampleType() == QAudioFormat::UnSignedInt && outFormat.sampleSize() == 8)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<quint8>::max());
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VectorXuint8 samplesI = samplesITemp.cast<quint8>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(quint8)));
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}
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if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 8)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint8>::max());
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VectorXint8 samplesI = samplesITemp.cast<qint8>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint8)));
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}
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if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 16)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint16>::max());
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VectorXint16 samplesI = samplesITemp.cast<qint16>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint16)));
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}
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else if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 32)
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{
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Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint32>::max());
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VectorXint32 samplesI = samplesITemp.cast<qint32>();
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audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint32)));
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}
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else if (outFormat.sampleType() == QAudioFormat::Float)
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{
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audio.data = QByteArray(reinterpret_cast<char*>(samplesF.data()), int(samplesF.size()) * int(sizeof(float)));
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}
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else {
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qInfo(logAudio()) << "Unsupported Sample Type:" << outFormat.sampleType();
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}
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/*
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As we currently don't have a float based uLaw encoder, this must be done
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after all other conversion has taken place.
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*/
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if (outFormat.codec() == "audio/PCMU")
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{
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QByteArray outPacket((int)audio.data.length() / 2, (char)0xff);
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qint16* in = (qint16*)audio.data.data();
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for (int f = 0; f < outPacket.length(); f++)
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{
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qint16 sample = *in++;
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int sign = (sample >> 8) & 0x80;
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if (sign)
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sample = (short)-sample;
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if (sample > cClip)
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sample = cClip;
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sample = (short)(sample + cBias);
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int exponent = (int)MuLawCompressTable[(sample >> 7) & 0xFF];
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int mantissa = (sample >> (exponent + 3)) & 0x0F;
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int compressedByte = ~(sign | (exponent << 4) | mantissa);
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outPacket[f] = (quint8)compressedByte;
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}
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audio.data.clear();
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audio.data = outPacket; // Copy output packet back to input buffer.
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}
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}
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}
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}
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