kopia lustrzana https://gitlab.com/eliggett/wfview
Merge branch 'wfserver' of https://gitlab.com/eliggett/wfview into wfserver
commit
066c1b58ac
|
@ -82,224 +82,225 @@ audioConverter::~audioConverter()
|
|||
bool audioConverter::convert(audioPacket audio)
|
||||
{
|
||||
|
||||
if (audio.data.size() == 0)
|
||||
// If inFormat and outFormat are identical, just emit the data back.
|
||||
if (audio.data.size() != 0 && inFormat != outFormat)
|
||||
{
|
||||
return false;
|
||||
}
|
||||
|
||||
if (inFormat.codec() == "audio/opus")
|
||||
{
|
||||
unsigned char* in = (unsigned char*)audio.data.data();
|
||||
if (inFormat.codec() == "audio/opus")
|
||||
{
|
||||
unsigned char* in = (unsigned char*)audio.data.data();
|
||||
|
||||
//Decode the frame.
|
||||
int nSamples = opus_packet_get_nb_samples(in, audio.data.size(), inFormat.sampleRate());
|
||||
if (nSamples == -1) {
|
||||
// No opus data yet?
|
||||
return false;
|
||||
}
|
||||
QByteArray outPacket(nSamples * sizeof(float) * inFormat.channelCount(), (char)0xff); // Preset the output buffer size.
|
||||
float* out = (float*)outPacket.data();
|
||||
//Decode the frame.
|
||||
int nSamples = opus_packet_get_nb_samples(in, audio.data.size(), inFormat.sampleRate());
|
||||
if (nSamples == -1) {
|
||||
// No opus data yet?
|
||||
return false;
|
||||
}
|
||||
QByteArray outPacket(nSamples * sizeof(float) * inFormat.channelCount(), (char)0xff); // Preset the output buffer size.
|
||||
float* out = (float*)outPacket.data();
|
||||
|
||||
//if (audio.seq > lastAudioSequence + 1) {
|
||||
// nSamples = opus_decode_float(opusDecoder, Q_NULLPTR, 0, out, nSamples, 1);
|
||||
//}
|
||||
//else {
|
||||
nSamples = opus_decode_float(opusDecoder, in, audio.data.size(), out, nSamples, 0);
|
||||
//}
|
||||
//lastAudioSequence = audio.seq;
|
||||
audio.data.clear();
|
||||
audio.data = outPacket; // Replace incoming data with converted.
|
||||
}
|
||||
else if (inFormat.codec() == "audio/PCMU")
|
||||
{
|
||||
// Current packet is "technically" 8bit so need to create a new buffer that is 16bit
|
||||
QByteArray outPacket((int)audio.data.length() * 2, (char)0xff);
|
||||
qint16* out = (qint16*)outPacket.data();
|
||||
for (int f = 0; f < audio.data.length(); f++)
|
||||
{
|
||||
*out++ = ulaw_decode[(quint8)audio.data[f]];
|
||||
}
|
||||
audio.data.clear();
|
||||
audio.data = outPacket; // Replace incoming data with converted.
|
||||
// Make sure that sample size/type is set correctly
|
||||
}
|
||||
//if (audio.seq > lastAudioSequence + 1) {
|
||||
// nSamples = opus_decode_float(opusDecoder, Q_NULLPTR, 0, out, nSamples, 1);
|
||||
//}
|
||||
//else {
|
||||
nSamples = opus_decode_float(opusDecoder, in, audio.data.size(), out, nSamples, 0);
|
||||
//}
|
||||
//lastAudioSequence = audio.seq;
|
||||
audio.data.clear();
|
||||
audio.data = outPacket; // Replace incoming data with converted.
|
||||
}
|
||||
else if (inFormat.codec() == "audio/PCMU")
|
||||
{
|
||||
// Current packet is "technically" 8bit so need to create a new buffer that is 16bit
|
||||
QByteArray outPacket((int)audio.data.length() * 2, (char)0xff);
|
||||
qint16* out = (qint16*)outPacket.data();
|
||||
for (int f = 0; f < audio.data.length(); f++)
|
||||
{
|
||||
*out++ = ulaw_decode[(quint8)audio.data[f]];
|
||||
}
|
||||
audio.data.clear();
|
||||
audio.data = outPacket; // Replace incoming data with converted.
|
||||
// Make sure that sample size/type is set correctly
|
||||
}
|
||||
|
||||
Eigen::VectorXf samplesF;
|
||||
Eigen::VectorXf samplesF;
|
||||
|
||||
if (inFormat.sampleType() == QAudioFormat::SignedInt && inFormat.sampleSize() == 32)
|
||||
{
|
||||
Eigen::Ref<VectorXint32> samplesI = Eigen::Map<VectorXint32>(reinterpret_cast<qint32*>(audio.data.data()), audio.data.size() / int(sizeof(qint32)));
|
||||
samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint32>::max());
|
||||
}
|
||||
else if (inFormat.sampleType() == QAudioFormat::SignedInt && inFormat.sampleSize() == 16)
|
||||
{
|
||||
Eigen::Ref<VectorXint16> samplesI = Eigen::Map<VectorXint16>(reinterpret_cast<qint16*>(audio.data.data()), audio.data.size() / int(sizeof(qint16)));
|
||||
samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint16>::max());
|
||||
}
|
||||
else if (inFormat.sampleType() == QAudioFormat::SignedInt && inFormat.sampleSize() == 8)
|
||||
{
|
||||
Eigen::Ref<VectorXint8> samplesI = Eigen::Map<VectorXint8>(reinterpret_cast<qint8*>(audio.data.data()), audio.data.size() / int(sizeof(qint8)));
|
||||
samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint8>::max());;
|
||||
}
|
||||
else if (inFormat.sampleType() == QAudioFormat::UnSignedInt && inFormat.sampleSize() == 8)
|
||||
{
|
||||
Eigen::Ref<VectorXuint8> samplesI = Eigen::Map<VectorXuint8>(reinterpret_cast<quint8*>(audio.data.data()), audio.data.size() / int(sizeof(quint8)));
|
||||
samplesF = samplesI.cast<float>() / float(std::numeric_limits<quint8>::max());;
|
||||
}
|
||||
else if (inFormat.sampleType() == QAudioFormat::Float) {
|
||||
samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(audio.data.data()), audio.data.size() / int(sizeof(float)));
|
||||
}
|
||||
else
|
||||
{
|
||||
qInfo(logAudio()) << "Unsupported Sample Type:" << inFormat.sampleType() << "Size:" << inFormat.sampleSize();
|
||||
}
|
||||
if (inFormat.sampleType() == QAudioFormat::SignedInt && inFormat.sampleSize() == 32)
|
||||
{
|
||||
Eigen::Ref<VectorXint32> samplesI = Eigen::Map<VectorXint32>(reinterpret_cast<qint32*>(audio.data.data()), audio.data.size() / int(sizeof(qint32)));
|
||||
samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint32>::max());
|
||||
}
|
||||
else if (inFormat.sampleType() == QAudioFormat::SignedInt && inFormat.sampleSize() == 16)
|
||||
{
|
||||
Eigen::Ref<VectorXint16> samplesI = Eigen::Map<VectorXint16>(reinterpret_cast<qint16*>(audio.data.data()), audio.data.size() / int(sizeof(qint16)));
|
||||
samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint16>::max());
|
||||
}
|
||||
else if (inFormat.sampleType() == QAudioFormat::SignedInt && inFormat.sampleSize() == 8)
|
||||
{
|
||||
Eigen::Ref<VectorXint8> samplesI = Eigen::Map<VectorXint8>(reinterpret_cast<qint8*>(audio.data.data()), audio.data.size() / int(sizeof(qint8)));
|
||||
samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint8>::max());;
|
||||
}
|
||||
else if (inFormat.sampleType() == QAudioFormat::UnSignedInt && inFormat.sampleSize() == 8)
|
||||
{
|
||||
Eigen::Ref<VectorXuint8> samplesI = Eigen::Map<VectorXuint8>(reinterpret_cast<quint8*>(audio.data.data()), audio.data.size() / int(sizeof(quint8)));
|
||||
samplesF = samplesI.cast<float>() / float(std::numeric_limits<quint8>::max());;
|
||||
}
|
||||
else if (inFormat.sampleType() == QAudioFormat::Float) {
|
||||
samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(audio.data.data()), audio.data.size() / int(sizeof(float)));
|
||||
}
|
||||
else
|
||||
{
|
||||
qInfo(logAudio()) << "Unsupported Sample Type:" << inFormat.sampleType() << "Size:" << inFormat.sampleSize();
|
||||
}
|
||||
|
||||
|
||||
audio.amplitude = samplesF.array().abs().maxCoeff();
|
||||
// Set the volume
|
||||
samplesF *= audio.volume;
|
||||
audio.amplitude = samplesF.array().abs().maxCoeff();
|
||||
// Set the volume
|
||||
samplesF *= audio.volume;
|
||||
|
||||
|
||||
|
||||
|
||||
/*
|
||||
samplesF is now an Eigen Vector of the current samples in float format
|
||||
The next step is to convert to the correct number of channels in outFormat.channelCount()
|
||||
*/
|
||||
/*
|
||||
samplesF is now an Eigen Vector of the current samples in float format
|
||||
The next step is to convert to the correct number of channels in outFormat.channelCount()
|
||||
*/
|
||||
|
||||
|
||||
if (inFormat.channelCount() == 2 && outFormat.channelCount() == 1) {
|
||||
// If we need to drop one of the audio channels, do it now
|
||||
Eigen::VectorXf samplesTemp(samplesF.size() / 2);
|
||||
samplesTemp = Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesF.data(), samplesF.size() / 2);
|
||||
samplesF = samplesTemp;
|
||||
}
|
||||
else if (inFormat.channelCount() == 1 && outFormat.channelCount() == 2) {
|
||||
// Convert mono to stereo if required
|
||||
Eigen::VectorXf samplesTemp(samplesF.size() * 2);
|
||||
Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data(), samplesF.size()) = samplesF;
|
||||
Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data() + 1, samplesF.size()) = samplesF;
|
||||
samplesF = samplesTemp;
|
||||
}
|
||||
if (inFormat.channelCount() == 2 && outFormat.channelCount() == 1) {
|
||||
// If we need to drop one of the audio channels, do it now
|
||||
Eigen::VectorXf samplesTemp(samplesF.size() / 2);
|
||||
samplesTemp = Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesF.data(), samplesF.size() / 2);
|
||||
samplesF = samplesTemp;
|
||||
}
|
||||
else if (inFormat.channelCount() == 1 && outFormat.channelCount() == 2) {
|
||||
// Convert mono to stereo if required
|
||||
Eigen::VectorXf samplesTemp(samplesF.size() * 2);
|
||||
Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data(), samplesF.size()) = samplesF;
|
||||
Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesTemp.data() + 1, samplesF.size()) = samplesF;
|
||||
samplesF = samplesTemp;
|
||||
}
|
||||
|
||||
/*
|
||||
Next step is to resample (if needed)
|
||||
*/
|
||||
/*
|
||||
Next step is to resample (if needed)
|
||||
*/
|
||||
|
||||
if (resampler != Q_NULLPTR && resampleRatio != 1.0)
|
||||
{
|
||||
quint32 outFrames = ((samplesF.size() / outFormat.channelCount()) * resampleRatio);
|
||||
quint32 inFrames = (samplesF.size() / outFormat.channelCount());
|
||||
QByteArray outPacket(outFrames * outFormat.channelCount() * sizeof(float), (char)0xff); // Preset the output buffer size.
|
||||
const float* in = (float*)samplesF.data();
|
||||
float* out = (float*)outPacket.data();
|
||||
if (resampler != Q_NULLPTR && resampleRatio != 1.0)
|
||||
{
|
||||
quint32 outFrames = ((samplesF.size() / outFormat.channelCount()) * resampleRatio);
|
||||
quint32 inFrames = (samplesF.size() / outFormat.channelCount());
|
||||
QByteArray outPacket(outFrames * outFormat.channelCount() * sizeof(float), (char)0xff); // Preset the output buffer size.
|
||||
const float* in = (float*)samplesF.data();
|
||||
float* out = (float*)outPacket.data();
|
||||
|
||||
int err = 0;
|
||||
if (outFormat.channelCount() == 1) {
|
||||
err = wf_resampler_process_float(resampler, 0, in, &inFrames, out, &outFrames);
|
||||
}
|
||||
else {
|
||||
err = wf_resampler_process_interleaved_float(resampler, in, &inFrames, out, &outFrames);
|
||||
}
|
||||
int err = 0;
|
||||
if (outFormat.channelCount() == 1) {
|
||||
err = wf_resampler_process_float(resampler, 0, in, &inFrames, out, &outFrames);
|
||||
}
|
||||
else {
|
||||
err = wf_resampler_process_interleaved_float(resampler, in, &inFrames, out, &outFrames);
|
||||
}
|
||||
|
||||
if (err) {
|
||||
qInfo(logAudioConverter()) << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
|
||||
}
|
||||
samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
|
||||
}
|
||||
if (err) {
|
||||
qInfo(logAudioConverter()) << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
|
||||
}
|
||||
samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
|
||||
}
|
||||
|
||||
/*
|
||||
If output is Opus so encode it now, don't do any more conversion on the output of Opus.
|
||||
*/
|
||||
/*
|
||||
If output is Opus so encode it now, don't do any more conversion on the output of Opus.
|
||||
*/
|
||||
|
||||
if (outFormat.codec() == "audio/opus")
|
||||
{
|
||||
float* in = (float*)samplesF.data();
|
||||
QByteArray outPacket(1275, (char)0xff); // Preset the output buffer size to MAXIMUM possible Opus frame size
|
||||
unsigned char* out = (unsigned char*)outPacket.data();
|
||||
if (outFormat.codec() == "audio/opus")
|
||||
{
|
||||
float* in = (float*)samplesF.data();
|
||||
QByteArray outPacket(1275, (char)0xff); // Preset the output buffer size to MAXIMUM possible Opus frame size
|
||||
unsigned char* out = (unsigned char*)outPacket.data();
|
||||
|
||||
int nbBytes = opus_encode_float(opusEncoder, in, (samplesF.size() / outFormat.channelCount()), out, outPacket.length());
|
||||
if (nbBytes < 0)
|
||||
{
|
||||
qInfo(logAudioConverter()) << "Opus encode failed:" << opus_strerror(nbBytes) << "Num Samples:" << samplesF.size();
|
||||
return false;
|
||||
}
|
||||
else {
|
||||
outPacket.resize(nbBytes);
|
||||
audio.data.clear();
|
||||
audio.data = outPacket; // Copy output packet back to input buffer.
|
||||
//samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
|
||||
}
|
||||
int nbBytes = opus_encode_float(opusEncoder, in, (samplesF.size() / outFormat.channelCount()), out, outPacket.length());
|
||||
if (nbBytes < 0)
|
||||
{
|
||||
qInfo(logAudioConverter()) << "Opus encode failed:" << opus_strerror(nbBytes) << "Num Samples:" << samplesF.size();
|
||||
return false;
|
||||
}
|
||||
else {
|
||||
outPacket.resize(nbBytes);
|
||||
audio.data.clear();
|
||||
audio.data = outPacket; // Copy output packet back to input buffer.
|
||||
//samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
|
||||
}
|
||||
|
||||
}
|
||||
else {
|
||||
}
|
||||
else {
|
||||
|
||||
|
||||
/*
|
||||
Now convert back into the output format required
|
||||
*/
|
||||
audio.data.clear();
|
||||
/*
|
||||
Now convert back into the output format required
|
||||
*/
|
||||
audio.data.clear();
|
||||
|
||||
if (outFormat.sampleType() == QAudioFormat::UnSignedInt && outFormat.sampleSize() == 8)
|
||||
{
|
||||
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<quint8>::max());
|
||||
VectorXuint8 samplesI = samplesITemp.cast<quint8>();
|
||||
audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(quint8)));
|
||||
}
|
||||
if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 8)
|
||||
{
|
||||
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint8>::max());
|
||||
VectorXint8 samplesI = samplesITemp.cast<qint8>();
|
||||
audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint8)));
|
||||
}
|
||||
if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 16)
|
||||
{
|
||||
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint16>::max());
|
||||
VectorXint16 samplesI = samplesITemp.cast<qint16>();
|
||||
audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint16)));
|
||||
}
|
||||
else if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 32)
|
||||
{
|
||||
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint32>::max());
|
||||
VectorXint32 samplesI = samplesITemp.cast<qint32>();
|
||||
audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint32)));
|
||||
}
|
||||
else if (outFormat.sampleType() == QAudioFormat::Float)
|
||||
{
|
||||
audio.data = QByteArray(reinterpret_cast<char*>(samplesF.data()), int(samplesF.size()) * int(sizeof(float)));
|
||||
}
|
||||
else {
|
||||
qInfo(logAudio()) << "Unsupported Sample Type:" << outFormat.sampleType();
|
||||
}
|
||||
if (outFormat.sampleType() == QAudioFormat::UnSignedInt && outFormat.sampleSize() == 8)
|
||||
{
|
||||
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<quint8>::max());
|
||||
VectorXuint8 samplesI = samplesITemp.cast<quint8>();
|
||||
audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(quint8)));
|
||||
}
|
||||
if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 8)
|
||||
{
|
||||
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint8>::max());
|
||||
VectorXint8 samplesI = samplesITemp.cast<qint8>();
|
||||
audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint8)));
|
||||
}
|
||||
if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 16)
|
||||
{
|
||||
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint16>::max());
|
||||
VectorXint16 samplesI = samplesITemp.cast<qint16>();
|
||||
audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint16)));
|
||||
}
|
||||
else if (outFormat.sampleType() == QAudioFormat::SignedInt && outFormat.sampleSize() == 32)
|
||||
{
|
||||
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint32>::max());
|
||||
VectorXint32 samplesI = samplesITemp.cast<qint32>();
|
||||
audio.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint32)));
|
||||
}
|
||||
else if (outFormat.sampleType() == QAudioFormat::Float)
|
||||
{
|
||||
audio.data = QByteArray(reinterpret_cast<char*>(samplesF.data()), int(samplesF.size()) * int(sizeof(float)));
|
||||
}
|
||||
else {
|
||||
qInfo(logAudio()) << "Unsupported Sample Type:" << outFormat.sampleType();
|
||||
}
|
||||
|
||||
/*
|
||||
As we currently don't have a float based uLaw encoder, this must be done
|
||||
after all other conversion has taken place.
|
||||
*/
|
||||
/*
|
||||
As we currently don't have a float based uLaw encoder, this must be done
|
||||
after all other conversion has taken place.
|
||||
*/
|
||||
|
||||
if (outFormat.codec() == "audio/PCMU")
|
||||
{
|
||||
QByteArray outPacket((int)audio.data.length() / 2, (char)0xff);
|
||||
qint16* in = (qint16*)audio.data.data();
|
||||
for (int f = 0; f < outPacket.length(); f++)
|
||||
{
|
||||
qint16 sample = *in++;
|
||||
int sign = (sample >> 8) & 0x80;
|
||||
if (sign)
|
||||
sample = (short)-sample;
|
||||
if (sample > cClip)
|
||||
sample = cClip;
|
||||
sample = (short)(sample + cBias);
|
||||
int exponent = (int)MuLawCompressTable[(sample >> 7) & 0xFF];
|
||||
int mantissa = (sample >> (exponent + 3)) & 0x0F;
|
||||
int compressedByte = ~(sign | (exponent << 4) | mantissa);
|
||||
outPacket[f] = (quint8)compressedByte;
|
||||
}
|
||||
audio.data.clear();
|
||||
audio.data = outPacket; // Copy output packet back to input buffer.
|
||||
if (outFormat.codec() == "audio/PCMU")
|
||||
{
|
||||
QByteArray outPacket((int)audio.data.length() / 2, (char)0xff);
|
||||
qint16* in = (qint16*)audio.data.data();
|
||||
for (int f = 0; f < outPacket.length(); f++)
|
||||
{
|
||||
qint16 sample = *in++;
|
||||
int sign = (sample >> 8) & 0x80;
|
||||
if (sign)
|
||||
sample = (short)-sample;
|
||||
if (sample > cClip)
|
||||
sample = cClip;
|
||||
sample = (short)(sample + cBias);
|
||||
int exponent = (int)MuLawCompressTable[(sample >> 7) & 0xFF];
|
||||
int mantissa = (sample >> (exponent + 3)) & 0x0F;
|
||||
int compressedByte = ~(sign | (exponent << 4) | mantissa);
|
||||
outPacket[f] = (quint8)compressedByte;
|
||||
}
|
||||
audio.data.clear();
|
||||
audio.data = outPacket; // Copy output packet back to input buffer.
|
||||
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
}
|
||||
}
|
||||
emit converted(audio);
|
||||
return true;
|
||||
}
|
||||
|
|
|
@ -85,6 +85,7 @@ static inline QAudioFormat toQAudioFormat(quint8 codec, quint32 sampleRate)
|
|||
format.setSampleType(QAudioFormat::UnSignedInt);
|
||||
format.setByteOrder(QAudioFormat::LittleEndian);
|
||||
format.setCodec("audio/pcm");
|
||||
format.setSampleRate(sampleRate);
|
||||
|
||||
if (codec == 0x01 || codec == 0x20) {
|
||||
/* Set sample to be what is expected by the encoder and the output of the decoder */
|
||||
|
@ -108,7 +109,6 @@ static inline QAudioFormat toQAudioFormat(quint8 codec, quint32 sampleRate)
|
|||
format.setCodec("audio/opus");
|
||||
}
|
||||
|
||||
format.setSampleRate(sampleRate);
|
||||
return format;
|
||||
}
|
||||
|
||||
|
|
|
@ -209,8 +209,8 @@ void audioHandler::setVolume(unsigned char volume)
|
|||
|
||||
void audioHandler::incomingAudio(audioPacket packet)
|
||||
{
|
||||
//QTime startProcessing = QTime::currentTime();
|
||||
if (audioDevice != Q_NULLPTR) {
|
||||
|
||||
if (audioDevice != Q_NULLPTR && packet.data.size() > 0) {
|
||||
packet.volume = volume;
|
||||
|
||||
emit sendToConverter(packet);
|
||||
|
@ -221,32 +221,35 @@ void audioHandler::incomingAudio(audioPacket packet)
|
|||
|
||||
void audioHandler::convertedOutput(audioPacket packet) {
|
||||
|
||||
currentLatency = packet.time.msecsTo(QTime::currentTime()) + (outFormat.durationForBytes(audioOutput->bufferSize() - audioOutput->bytesFree()) / 1000);
|
||||
if (audioDevice != Q_NULLPTR) {
|
||||
audioDevice->write(packet.data);
|
||||
if (lastReceived.msecsTo(QTime::currentTime()) > 100) {
|
||||
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Time since last audio packet" << lastReceived.msecsTo(QTime::currentTime()) << "Expected around" << setup.blockSize;
|
||||
}
|
||||
lastReceived = QTime::currentTime();
|
||||
}
|
||||
/*if ((packet.seq > lastSentSeq + 1) && (setup.codec == 0x40 || setup.codec == 0x80)) {
|
||||
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Attempting FEC on packet" << packet.seq << "as last is" << lastSentSeq;
|
||||
lastSentSeq = packet.seq;
|
||||
incomingAudio(packet); // Call myself again to run the packet a second time (FEC)
|
||||
}
|
||||
*/
|
||||
lastSentSeq = packet.seq;
|
||||
emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun);
|
||||
if (packet.data.size() > 0 ) {
|
||||
|
||||
amplitude = packet.amplitude;
|
||||
currentLatency = packet.time.msecsTo(QTime::currentTime()) + (outFormat.durationForBytes(audioOutput->bufferSize() - audioOutput->bytesFree()) / 1000);
|
||||
if (audioDevice != Q_NULLPTR) {
|
||||
audioDevice->write(packet.data);
|
||||
if (lastReceived.msecsTo(QTime::currentTime()) > 100) {
|
||||
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Time since last audio packet" << lastReceived.msecsTo(QTime::currentTime()) << "Expected around" << setup.blockSize;
|
||||
}
|
||||
lastReceived = QTime::currentTime();
|
||||
}
|
||||
/*if ((packet.seq > lastSentSeq + 1) && (setup.codec == 0x40 || setup.codec == 0x80)) {
|
||||
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Attempting FEC on packet" << packet.seq << "as last is" << lastSentSeq;
|
||||
lastSentSeq = packet.seq;
|
||||
incomingAudio(packet); // Call myself again to run the packet a second time (FEC)
|
||||
}
|
||||
*/
|
||||
lastSentSeq = packet.seq;
|
||||
emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun);
|
||||
|
||||
amplitude = packet.amplitude;
|
||||
}
|
||||
}
|
||||
|
||||
void audioHandler::getNextAudioChunk()
|
||||
{
|
||||
tempBuf.data.append(audioDevice->readAll());
|
||||
|
||||
while (tempBuf.data.length() >= outFormat.bytesForDuration(setup.blockSize * 1000)) {
|
||||
if (audioDevice) {
|
||||
tempBuf.data.append(audioDevice->readAll());
|
||||
}
|
||||
if (tempBuf.data.length() >= outFormat.bytesForDuration(setup.blockSize * 1000)) {
|
||||
audioPacket packet;
|
||||
packet.time = QTime::currentTime();
|
||||
packet.sent = 0;
|
||||
|
@ -259,20 +262,27 @@ void audioHandler::getNextAudioChunk()
|
|||
|
||||
emit sendToConverter(packet);
|
||||
}
|
||||
return;
|
||||
|
||||
/* If there is still enough data in the buffer, call myself again in 20ms */
|
||||
if (tempBuf.data.length() >= outFormat.bytesForDuration(setup.blockSize * 1000)) {
|
||||
QTimer::singleShot(setup.blockSize, this, &audioHandler::getNextAudioChunk);
|
||||
}
|
||||
|
||||
return;
|
||||
}
|
||||
|
||||
|
||||
void audioHandler::convertedInput(audioPacket audio)
|
||||
{
|
||||
emit haveAudioData(audio);
|
||||
if (lastReceived.msecsTo(QTime::currentTime()) > 100) {
|
||||
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Time since last audio packet" << lastReceived.msecsTo(QTime::currentTime()) << "Expected around" << setup.blockSize ;
|
||||
}
|
||||
lastReceived = QTime::currentTime();
|
||||
amplitude = audio.amplitude;
|
||||
emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun);
|
||||
if (audio.data.size() > 0) {
|
||||
emit haveAudioData(audio);
|
||||
if (lastReceived.msecsTo(QTime::currentTime()) > 100) {
|
||||
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Time since last audio packet" << lastReceived.msecsTo(QTime::currentTime()) << "Expected around" << setup.blockSize ;
|
||||
}
|
||||
lastReceived = QTime::currentTime();
|
||||
amplitude = audio.amplitude;
|
||||
emit haveLevels(getAmplitude(), setup.latency, currentLatency, isUnderrun);
|
||||
}
|
||||
}
|
||||
|
||||
void audioHandler::changeLatency(const quint16 newSize)
|
||||
|
|
Ładowanie…
Reference in New Issue