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More tidying and use float resampler

merge-requests/9/merge
Phil Taylor 2022-04-05 16:47:43 +01:00
rodzic 45ac1fbe1c
commit 38fdec3da6
2 zmienionych plików z 211 dodań i 201 usunięć
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353
audiohandler.cpp
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@ -66,12 +66,23 @@ bool audioHandler::init(audioSetup setupIn)
setup.format.setSampleType(QAudioFormat::UnSignedInt);
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "audio handler starting:" << setup.name;
if (setup.port.isNull())
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "No audio device was found. You probably need to install libqt5multimedia-plugins.";
return false;
}
if (setup.codec == 0x01 || setup.codec == 0x20) {
/* Althought uLaw is 8bit unsigned, it is 16bit signed once decoded*/
setup.ulaw = true;
setup.format.setSampleSize(16);
setup.format.setSampleType(QAudioFormat::SignedInt);
}
if (setup.codec == 0x08 || setup.codec == 0x10 || setup.codec == 0x20 || setup.codec == 0x80) {
setup.format.setChannelCount(2);
}
if (setup.codec == 0x04 || setup.codec == 0x10 || setup.codec == 0x40 || setup.codec == 0x80) {
setup.format.setSampleSize(16);
setup.format.setSampleType(QAudioFormat::SignedInt);
@ -95,25 +106,9 @@ bool audioHandler::init(audioSetup setupIn)
this->setVolume(setup.localAFgain);
}
/* format.setSampleSize(16);
format.setChannelCount(2);
format.setSampleRate(INTERNAL_SAMPLE_RATE);
format.setCodec("audio/pcm");
format.setByteOrder(QAudioFormat::LittleEndian);
format.setSampleType(QAudioFormat::SignedInt); */
format = setup.port.preferredFormat();
qDebug(logAudio()) << "Preferred Format: SampleSize" << format.sampleSize() << "Channel Count" << format.channelCount() <<
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Preferred Format: SampleSize" << format.sampleSize() << "Channel Count" << format.channelCount() <<
"Sample Rate" << format.sampleRate() << "Codec" << format.codec() << "Sample Type" << format.sampleType();
if (setup.port.isNull())
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "No audio device was found. You probably need to install libqt5multimedia-plugins.";
return false;
}
else if (!setup.port.isFormatSupported(format))
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Format not supported, choosing nearest supported format - which may not work!";
format=setup.port.nearestFormat(format);
}
if (format.channelCount() > 2) {
format.setChannelCount(2);
}
@ -123,29 +118,22 @@ bool audioHandler::init(audioSetup setupIn)
return false;
}
devChannels = format.channelCount();
nativeSampleRate = format.sampleRate();
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Internal: sample rate" << format.sampleRate() << "channel count" << format.channelCount();
// We "hopefully" now have a valid format that is supported so try connecting
if (setup.isinput) {
audioInput = new QAudioInput(setup.port, format, this);
//connect(audioInput, SIGNAL(notify()), SLOT(notified()));
isInitialized = true;
}
else {
audioOutput = new QAudioOutput(setup.port, format, this);
isInitialized = true;
}
// Setup resampler and opus if they are needed.
int resample_error = 0;
int opus_err = 0;
if (setup.isinput) {
resampler = wf_resampler_init(devChannels, nativeSampleRate, setup.format.sampleRate(), setup.resampleQuality, &resample_error);
resampler = wf_resampler_init(format.channelCount(), format.sampleRate(), setup.format.sampleRate(), setup.resampleQuality, &resample_error);
if (setup.codec == 0x40 || setup.codec == 0x80) {
// Opus codec
encoder = opus_encoder_create(setup.format.sampleRate(), setup.format.channelCount(), OPUS_APPLICATION_AUDIO, &opus_err);
@ -160,7 +148,7 @@ bool audioHandler::init(audioSetup setupIn)
//resampBufs = new r8b::CFixedBuffer<double>[format.channelCount()];
//resamps = new r8b::CPtrKeeper<r8b::CDSPResampler24*>[format.channelCount()];
resampler = wf_resampler_init(devChannels, setup.format.sampleRate(), this->nativeSampleRate, setup.resampleQuality, &resample_error);
resampler = wf_resampler_init(format.channelCount(), setup.format.sampleRate(), format.sampleRate(), setup.resampleQuality, &resample_error);
if (setup.codec == 0x40 || setup.codec == 0x80) {
// Opus codec
decoder = opus_decoder_create(setup.format.sampleRate(), setup.format.sampleRate(), &opus_err);
@ -177,7 +165,7 @@ bool audioHandler::init(audioSetup setupIn)
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "thread id" << QThread::currentThreadId();
this->start();
return isInitialized;
return true;
}
void audioHandler::start()
@ -284,8 +272,6 @@ void audioHandler::incomingAudio(audioPacket inPacket)
}
livePacket.data.clear();
livePacket.data = outPacket; // Replace incoming data with converted.
setup.format.setSampleSize(16);
setup.format.setSampleType(QAudioFormat::SignedInt);
// Buffer now contains 16bit signed samples.
}
@ -304,6 +290,10 @@ void audioHandler::incomingAudio(audioPacket inPacket)
VectorXuint8 samplesI = Eigen::Map<VectorXuint8>(reinterpret_cast<quint8*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(quint8)));
samplesF = samplesI.cast<float>() / float(std::numeric_limits<quint8>::max());;
}
else {
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Unsupported Sample Type:" << format.sampleType();
}
/* samplesF is now an Eigen Vector of the current samples in float format */
@ -322,6 +312,25 @@ void audioHandler::incomingAudio(audioPacket inPacket)
}
if (resampleRatio != 1.0) {
// We need to resample
// We have a stereo 16bit stream.
livePacket.data = QByteArray(reinterpret_cast<char*>(samplesF.data()), int(samplesF.size()) * int(sizeof(float)));
quint32 outFrames = ((livePacket.data.length() / sizeof(float) / format.channelCount()) * resampleRatio);
quint32 inFrames = (livePacket.data.length() / sizeof(float) / format.channelCount());
QByteArray outPacket(outFrames * format.channelCount() * sizeof(float), (char)0xff); // Preset the output buffer size.
const float* in = (float*)livePacket.data.data();
float* out = (float*)outPacket.data();
int err = 0;
err = wf_resampler_process_interleaved_float(resampler, in, &inFrames, out, &outFrames);
if (err) {
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
}
samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
}
if (format.sampleType() == QAudioFormat::UnSignedInt && format.sampleSize() == 8)
{
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<quint8>::max());
@ -349,25 +358,6 @@ void audioHandler::incomingAudio(audioPacket inPacket)
}
if (resampleRatio != 1.0) {
// We need to resample
// We have a stereo 16bit stream.
quint32 outFrames = ((livePacket.data.length() / 2 / devChannels) * resampleRatio);
quint32 inFrames = (livePacket.data.length() / 2 / devChannels);
QByteArray outPacket(outFrames * 4, (char)0xff); // Preset the output buffer size.
const qint16* in = (qint16*)livePacket.data.constData();
qint16* out = (qint16*)outPacket.data();
int err = 0;
err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
if (err) {
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
}
livePacket.data.clear();
livePacket.data = outPacket; // Replace incoming data with converted.
}
//qDebug(logAudio()) << "Adding packet to buffer:" << livePacket.seq << ": " << livePacket.data.length();
@ -389,6 +379,157 @@ void audioHandler::incomingAudio(audioPacket inPacket)
return;
}
void audioHandler::getNextAudioChunk(QByteArray& ret)
{
audioPacket livePacket;
livePacket.sent = 0;
if (audioDevice != Q_NULLPTR) {
livePacket.data = audioDevice->readAll();
if (livePacket.data.length() > 0)
{
Eigen::VectorXf samplesF;
if (format.sampleType() == QAudioFormat::SignedInt && format.sampleSize() == 32)
{
VectorXint32 samplesI = Eigen::Map<VectorXint32>(reinterpret_cast<qint32*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(qint32)));
samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint32>::max());
}
else if (format.sampleType() == QAudioFormat::SignedInt && format.sampleSize() == 16)
{
VectorXint16 samplesI = Eigen::Map<VectorXint16>(reinterpret_cast<qint16*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(qint16)));
samplesF = samplesI.cast<float>() / float(std::numeric_limits<qint16>::max());
}
else if (format.sampleType() == QAudioFormat::UnSignedInt && format.sampleSize() == 8)
{
VectorXuint8 samplesI = Eigen::Map<VectorXuint8>(reinterpret_cast<quint8*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(quint8)));
samplesF = samplesI.cast<float>() / float(std::numeric_limits<quint8>::max());;
}
else if (format.sampleType() == QAudioFormat::Float)
{
samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(livePacket.data.data()), livePacket.data.size() / int(sizeof(float)));
}
else {
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Unsupported Sample Type:" << format.sampleType() << format.sampleSize();
}
/* samplesF is now an Eigen Vector of the current samples in float format */
// Set the max amplitude found in the vector
amplitude = samplesF.array().abs().maxCoeff();
// Channel count should now match the device that audio is going to (rig)
if (resampleRatio != 1.0) {
// We need to resample
// We have a stereo 16bit stream.
livePacket.data = QByteArray(reinterpret_cast<char*>(samplesF.data()), int(samplesF.size()) * int(sizeof(float)));
quint32 outFrames = ((samplesF.size() / format.channelCount()) * resampleRatio);
quint32 inFrames = (samplesF.size() / format.channelCount());
QByteArray outPacket(outFrames * format.channelCount() * sizeof(float), (char)0xff); // Preset the output buffer size.
const float* in = (float*)livePacket.data.data();
float* out = (float*)outPacket.data();
int err = 0;
err = wf_resampler_process_interleaved_float(resampler, in, &inFrames, out, &outFrames);
if (err) {
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
}
samplesF = Eigen::Map<Eigen::VectorXf>(reinterpret_cast<float*>(outPacket.data()), outPacket.size() / int(sizeof(float)));
}
// If we need to drop one of the audio channels, do it now
if (format.channelCount() == 2 && setup.format.channelCount() == 1) {
Eigen::VectorXf samplesTemp(samplesF.size() / 2);
samplesTemp = Eigen::Map<Eigen::VectorXf, 0, Eigen::InnerStride<2> >(samplesF.data(), samplesF.size() / 2);
samplesF = samplesTemp;
}
if (setup.format.sampleType() == QAudioFormat::UnSignedInt && setup.format.sampleSize() == 8)
{
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<quint8>::max());
VectorXuint8 samplesI = samplesITemp.cast<quint8>();
livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(quint8)));
}
if (setup.format.sampleType() == QAudioFormat::SignedInt && setup.format.sampleSize() == 16)
{
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint16>::max());
VectorXint16 samplesI = samplesITemp.cast<qint16>();
livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint16)));
}
else if (setup.format.sampleType() == QAudioFormat::SignedInt && setup.format.sampleSize() == 32)
{
Eigen::VectorXf samplesITemp = samplesF * float(std::numeric_limits<qint32>::max());
VectorXint32 samplesI = samplesITemp.cast<qint32>();
livePacket.data = QByteArray(reinterpret_cast<char*>(samplesI.data()), int(samplesI.size()) * int(sizeof(qint32)));
}
else if (setup.format.sampleType() == QAudioFormat::Float)
{
livePacket.data = QByteArray(reinterpret_cast<char*>(samplesF.data()), int(samplesF.size()) * int(sizeof(float)));
}
else {
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Unsupported Sample Type:" << format.sampleType();
}
//qDebug(logAudio()) << "Now mono, length" << packet.data.length();
if (setup.ulaw)
{
QByteArray outPacket((int)livePacket.data.length() / 2, (char)0xff);
qint16* in = (qint16*)livePacket.data.data();
for (int f = 0; f < outPacket.length(); f++)
{
qint16 sample = *in++;
if (setup.ulaw) {
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;
}
}
livePacket.data.clear();
livePacket.data = outPacket; // Copy output packet back to input buffer.
}
else if (setup.codec == 0x40 || setup.codec == 0x80)
{
//Are we using the opus codec?
qint16* in = (qint16*)livePacket.data.data();
/* Encode the frame. */
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(encoder, in, (setup.format.sampleRate() / 50), out, outPacket.length());
if (nbBytes < 0)
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus encode failed:" << opus_strerror(nbBytes);
return;
}
else {
outPacket.resize(nbBytes);
livePacket.data.clear();
livePacket.data = outPacket; // Replace incoming data with converted.
}
}
ret = livePacket.data;
}
}
return;
}
void audioHandler::changeLatency(const quint16 newSize)
{
@ -398,7 +539,7 @@ void audioHandler::changeLatency(const quint16 newSize)
stop();
start();
}
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Configured latency: " << setup.latency << "Buffer Duration:" << getAudioDuration(audioOutput->bufferSize(),format) <<"ms";
qDebug(logAudio()) << (setup.isinput ? "Input" : "Output") << "Configured latency: " << setup.latency << "Buffer Duration:" << getAudioDuration(audioOutput->bufferSize(), format) << "ms";
}
@ -409,120 +550,6 @@ int audioHandler::getLatency()
void audioHandler::getNextAudioChunk(QByteArray& ret)
{
audioPacket packet;
packet.sent = 0;
if (audioDevice != Q_NULLPTR) {
packet.data = audioDevice->readAll();
}
if (packet.data.length() > 0)
{
// Packet will arrive as stereo interleaved 16bit 48K
if (resampleRatio != 1.0)
{
quint32 outFrames = ((packet.data.length() / 2 / devChannels) * resampleRatio);
quint32 inFrames = (packet.data.length() / 2 / devChannels);
QByteArray outPacket((int)outFrames * 2 * devChannels, (char)0xff);
const qint16* in = (qint16*)packet.data.constData();
qint16* out = (qint16*)outPacket.data();
int err = 0;
err = wf_resampler_process_interleaved_int(resampler, in, &inFrames, out, &outFrames);
if (err) {
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Resampler error " << err << " inFrames:" << inFrames << " outFrames:" << outFrames;
}
packet.data.clear();
packet.data = outPacket; // Copy output packet back to input buffer.
}
//qDebug(logAudio()) << "Now resampled, length" << packet.data.length();
int tempAmplitude = 0;
// Do we need to convert mono to stereo?
if (setup.format.channelCount() == 1 && devChannels > 1)
{
// Strip out right channel?
QByteArray outPacket(packet.data.length()/2, (char)0xff);
const qint16* in = (qint16*)packet.data.constData();
qint16* out = (qint16*)outPacket.data();
for (int f = 0; f < outPacket.length()/2; f++)
{
tempAmplitude = qMax(tempAmplitude, (int)(abs(*in) / 256));
*out++ = *in++;
in++; // Skip each even channel.
}
packet.data.clear();
packet.data = outPacket; // Copy output packet back to input buffer.
}
//qDebug(logAudio()) << "Now mono, length" << packet.data.length();
if (setup.codec == 0x40 || setup.codec == 0x80)
{
//Are we using the opus codec?
qint16* in = (qint16*)packet.data.data();
/* Encode the frame. */
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(encoder, in, (setup.format.sampleRate() / 50), out, outPacket.length());
if (nbBytes < 0)
{
qInfo(logAudio()) << (setup.isinput ? "Input" : "Output") << "Opus encode failed:" << opus_strerror(nbBytes);
return;
}
else {
outPacket.resize(nbBytes);
packet.data.clear();
packet.data = outPacket; // Replace incoming data with converted.
}
}
else if (setup.format.sampleSize() == 8)
{
// Do we need to convert 16-bit to 8-bit?
QByteArray outPacket((int)packet.data.length() / 2, (char)0xff);
qint16* in = (qint16*)packet.data.data();
for (int f = 0; f < outPacket.length(); f++)
{
qint16 sample = *in++;
if (setup.ulaw) {
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;
}
else {
int compressedByte = (((sample + 32768) >> 8) & 0xff);
outPacket[f] = (quint8)compressedByte;
}
tempAmplitude = qMax(tempAmplitude, abs(outPacket[f]));
}
packet.data.clear();
packet.data = outPacket; // Copy output packet back to input buffer.
}
amplitude = tempAmplitude;
ret = packet.data;
}
return;
}
quint16 audioHandler::getAmplitude()
{
return static_cast<quint16>(amplitude * 255.0);

59
audiohandler.h
Wyświetl plik

@ -1,63 +1,52 @@
#ifndef AUDIOHANDLER_H
#define AUDIOHANDLER_H
/* QT Headers */
#include <QObject>
#include <QByteArray>
#include <QMutex>
#include <QtEndian>
#include <QtMath>
#include <QThread>
#include <QTimer>
#include <QTime>
#include <QMap>
#include <QDebug>
#if defined(RTAUDIO)
#ifdef Q_OS_WIN
#include "RtAudio.h"
#else
#include "rtaudio/RtAudio.h"
#endif
#elif defined (PORTAUDIO)
#include "portaudio.h"
//#error "PORTAUDIO is not currently supported"
#else
/* QT Audio Headers */
#include <QAudioOutput>
#include <QAudioFormat>
#include <QAudioDeviceInfo>
#include <QAudioInput>
#include <QIODevice>
#endif
#include "packettypes.h"
typedef signed short MY_TYPE;
#define FORMAT RTAUDIO_SINT16
#include <QThread>
#include <QTimer>
#include <QTime>
#include <QMap>
/* Current resampler code */
#include "resampler/speex_resampler.h"
/* Potential new resampler */
//#include <r8bbase.h>
//#include <CDSPResampler.h>
/* Opus */
#ifdef Q_OS_WIN
#include "opus.h"
#else
#include "opus/opus.h"
#endif
#include "audiotaper.h"
/* Eigen */
#ifdef Q_OS_LINUX
#include <eigen3/Eigen/Eigen>
#else
#include <Eigen/Eigen>
#endif
//#include <r8bbase.h>
//#include <CDSPResampler.h>
/* wfview Packet types */
#include "packettypes.h"
#include <QDebug>
//#define BUFFER_SIZE (32*1024)
#define INTERNAL_SAMPLE_RATE 48000
/* Logarithmic taper for volume control */
#include "audiotaper.h"
#define MULAW_BIAS 33
#define MULAW_MAX 0x1fff
@ -132,20 +121,13 @@ private:
//r8b::CPtrKeeper<r8b::CDSPResampler24*>* resamps;
quint16 audioLatency;
bool chunkAvailable;
quint32 lastSeq;
quint32 lastSentSeq=0;
qint64 elapsedMs = 0;
quint16 nativeSampleRate=0;
quint8 radioSampleBits;
quint8 radioChannels;
int delayedPackets=0;
QMap<quint32, audioPacket>audioBuffer;
double resampleRatio;
volatile bool ready = false;
@ -153,8 +135,9 @@ private:
quint16 currentLatency;
float amplitude;
qreal volume=1.0;
int devChannels;
audioSetup setup;
OpusEncoder* encoder=Q_NULLPTR;
OpusDecoder* decoder=Q_NULLPTR;
};