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