mirror of
https://github.com/f4exb/sdrangel.git
synced 2024-11-22 16:08:39 -05:00
594 lines
17 KiB
C++
594 lines
17 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
|
|
// Copyright (C) 2012 maintech GmbH, Otto-Hahn-Str. 15, 97204 Hoechberg, Germany //
|
|
// written by Christian Daniel //
|
|
// Copyright (C) 2014 John Greb <hexameron@spam.no> //
|
|
// Copyright (C) 2015-2020, 2022-2023 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
|
|
// Copyright (C) 2016 Ziga S <ziga.svetina@gmail.com> //
|
|
// Copyright (C) 2022-2023 Jon Beniston, M7RCE <jon@beniston.com> //
|
|
// Copyright (C) 2022 Jiří Pinkava <jiri.pinkava@rossum.ai> //
|
|
// //
|
|
// This program is free software; you can redistribute it and/or modify //
|
|
// it under the terms of the GNU General Public License as published by //
|
|
// the Free Software Foundation as version 3 of the License, or //
|
|
// (at your option) any later version. //
|
|
// //
|
|
// This program is distributed in the hope that it will be useful, //
|
|
// but WITHOUT ANY WARRANTY; without even the implied warranty of //
|
|
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
|
|
// GNU General Public License V3 for more details. //
|
|
// //
|
|
// You should have received a copy of the GNU General Public License //
|
|
// along with this program. If not, see <http://www.gnu.org/licenses/>. //
|
|
///////////////////////////////////////////////////////////////////////////////////
|
|
|
|
#include <string.h>
|
|
#include <QThread>
|
|
#include <QAudioFormat>
|
|
#if QT_VERSION >= QT_VERSION_CHECK(6, 0, 0)
|
|
#include <QAudioSink>
|
|
#else
|
|
#include <QAudioOutput>
|
|
#endif
|
|
#include "audiooutputdevice.h"
|
|
#include "audiodeviceinfo.h"
|
|
#include "audiofifo.h"
|
|
#include "audionetsink.h"
|
|
#include "dsp/wavfilerecord.h"
|
|
|
|
MESSAGE_CLASS_DEFINITION(AudioOutputDevice::MsgStart, Message)
|
|
MESSAGE_CLASS_DEFINITION(AudioOutputDevice::MsgStop, Message)
|
|
MESSAGE_CLASS_DEFINITION(AudioOutputDevice::MsgReportSampleRate, Message)
|
|
|
|
AudioOutputDevice::AudioOutputDevice() :
|
|
m_audioOutput(nullptr),
|
|
m_audioNetSink(nullptr),
|
|
m_wavFileRecord(nullptr),
|
|
m_copyAudioToUdp(false),
|
|
m_udpChannelMode(UDPChannelLeft),
|
|
m_udpChannelCodec(UDPCodecL16),
|
|
m_audioUsageCount(0),
|
|
m_onExit(false),
|
|
m_volume(1.0),
|
|
m_recordToFile(false),
|
|
m_recordSilenceTime(0),
|
|
m_recordSilenceNbSamples(0),
|
|
m_recordSilenceCount(0),
|
|
m_audioFifos(),
|
|
m_managerMessageQueue(nullptr)
|
|
{
|
|
connect(&m_inputMessageQueue, SIGNAL(messageEnqueued()), this, SLOT(handleInputMessages()), Qt::QueuedConnection);
|
|
}
|
|
|
|
AudioOutputDevice::~AudioOutputDevice()
|
|
{
|
|
// stop();
|
|
//
|
|
// QMutexLocker mutexLocker(&m_mutex);
|
|
//
|
|
// for (std::list<AudioFifo*>::iterator it = m_audioFifos.begin(); it != m_audioFifos.end(); ++it)
|
|
// {
|
|
// delete *it;
|
|
// }
|
|
//
|
|
// m_audioFifos.clear();
|
|
}
|
|
|
|
bool AudioOutputDevice::start(int deviceIndex, int sampleRate)
|
|
{
|
|
// if (m_audioOutput) {
|
|
// return true;
|
|
// }
|
|
// if (m_audioUsageCount == 0)
|
|
// {
|
|
qDebug("AudioOutputDevice::start: device: %d rate: %d thread: %p", deviceIndex, sampleRate, QThread::currentThread());
|
|
QMutexLocker mutexLocker(&m_mutex);
|
|
AudioDeviceInfo devInfo;
|
|
|
|
if (deviceIndex < 0)
|
|
{
|
|
devInfo = AudioDeviceInfo::defaultOutputDevice();
|
|
qWarning("AudioOutputDevice::start: using system default device %s", qPrintable(devInfo.defaultOutputDevice().deviceName()));
|
|
}
|
|
else
|
|
{
|
|
auto &devicesInfo = AudioDeviceInfo::availableOutputDevices();
|
|
|
|
if (deviceIndex < devicesInfo.size())
|
|
{
|
|
devInfo = devicesInfo[deviceIndex];
|
|
qWarning("AudioOutputDevice::start: using audio device #%d: %s", deviceIndex, qPrintable(devInfo.deviceName()));
|
|
}
|
|
else
|
|
{
|
|
devInfo = AudioDeviceInfo::defaultOutputDevice();
|
|
qWarning("AudioOutputDevice::start: audio device #%d does not exist. Using system default device %s", deviceIndex, qPrintable(devInfo.defaultOutputDevice().deviceName()));
|
|
deviceIndex = -1;
|
|
}
|
|
}
|
|
|
|
//QAudioDeviceInfo devInfo(QAudioDeviceInfo::defaultOutputDevice());
|
|
#if QT_VERSION >= QT_VERSION_CHECK(6, 0, 0)
|
|
// Start with a valid format
|
|
m_audioFormat = devInfo.deviceInfo().preferredFormat();
|
|
#endif
|
|
|
|
m_audioFormat.setSampleRate(sampleRate);
|
|
m_audioFormat.setChannelCount(2);
|
|
#if QT_VERSION >= QT_VERSION_CHECK(6, 0, 0)
|
|
m_audioFormat.setSampleFormat(QAudioFormat::Int16);
|
|
#else
|
|
m_audioFormat.setSampleSize(16);
|
|
m_audioFormat.setCodec("audio/pcm");
|
|
m_audioFormat.setByteOrder(QAudioFormat::LittleEndian);
|
|
m_audioFormat.setSampleType(QAudioFormat::SignedInt);
|
|
#endif
|
|
|
|
if (!devInfo.isFormatSupported(m_audioFormat))
|
|
{
|
|
#if QT_VERSION >= QT_VERSION_CHECK(6, 0, 0)
|
|
qWarning("AudioOutputDevice::start: format %d Hz 2xS16LE audio/pcm not supported.", sampleRate);
|
|
#else
|
|
m_audioFormat = devInfo.deviceInfo().nearestFormat(m_audioFormat);
|
|
std::ostringstream os;
|
|
os << " sampleRate: " << m_audioFormat.sampleRate()
|
|
<< " channelCount: " << m_audioFormat.channelCount()
|
|
<< " sampleSize: " << m_audioFormat.sampleSize()
|
|
<< " codec: " << m_audioFormat.codec().toStdString()
|
|
<< " byteOrder: " << (m_audioFormat.byteOrder() == QAudioFormat::BigEndian ? "BE" : "LE")
|
|
<< " sampleType: " << (int) m_audioFormat.sampleType();
|
|
qWarning("AudioOutputDevice::start: format %d Hz 2xS16LE audio/pcm not supported. Using: %s", sampleRate, os.str().c_str());
|
|
#endif
|
|
}
|
|
else
|
|
{
|
|
qInfo("AudioOutputDevice::start: audio format OK");
|
|
}
|
|
|
|
#if QT_VERSION >= QT_VERSION_CHECK(6, 0, 0)
|
|
if (m_audioFormat.sampleFormat() != QAudioFormat::Int16)
|
|
{
|
|
qWarning("AudioOutputDevice::start: Audio device '%s' failed", qPrintable(devInfo.deviceName()));
|
|
return false;
|
|
}
|
|
#else
|
|
if (m_audioFormat.sampleSize() != 16)
|
|
{
|
|
qWarning("AudioOutputDevice::start: Audio device '%s' failed", qPrintable(devInfo.deviceName()));
|
|
return false;
|
|
}
|
|
#endif
|
|
|
|
#if QT_VERSION >= QT_VERSION_CHECK(6, 0, 0)
|
|
m_audioOutput = new QAudioSink(devInfo.deviceInfo(), m_audioFormat);
|
|
#else
|
|
m_audioOutput = new QAudioOutput(devInfo.deviceInfo(), m_audioFormat);
|
|
#endif
|
|
m_audioNetSink = new AudioNetSink(0, m_audioFormat.sampleRate(), false);
|
|
m_wavFileRecord = new WavFileRecord(m_audioFormat.sampleRate());
|
|
m_audioOutput->setVolume(m_volume);
|
|
// m_audioOutput->setBufferSize(m_audioFormat.sampleRate() / 5); FIXME: does not work generally
|
|
m_recordSilenceNbSamples = (m_recordSilenceTime * m_audioFormat.sampleRate()) / 10; // time in 100'ś ms
|
|
|
|
QIODevice::open(QIODevice::ReadOnly | QIODevice::Unbuffered);
|
|
|
|
m_audioOutput->start(this);
|
|
|
|
if (m_audioOutput->state() != QAudio::ActiveState) {
|
|
qWarning() << "AudioOutputDevice::start: cannot start - " << m_audioOutput->error();
|
|
} else {
|
|
qDebug("AudioOutputDevice::start: started buffer: %d bytes", (int)m_audioOutput->bufferSize());
|
|
}
|
|
|
|
if (m_managerMessageQueue) {
|
|
m_managerMessageQueue->push(AudioOutputDevice::MsgReportSampleRate::create(deviceIndex, devInfo.deviceName(), m_audioFormat.sampleRate()));
|
|
}
|
|
// }
|
|
//
|
|
// m_audioUsageCount++;
|
|
|
|
return true;
|
|
}
|
|
|
|
void AudioOutputDevice::stop()
|
|
{
|
|
if (!m_audioOutput) {
|
|
return;
|
|
}
|
|
|
|
qDebug("AudioOutputDevice::stop: thread: %p", QThread::currentThread());
|
|
|
|
QMutexLocker mutexLocker(&m_mutex);
|
|
m_audioOutput->stop();
|
|
QIODevice::close();
|
|
delete m_audioNetSink;
|
|
m_audioNetSink = nullptr;
|
|
delete m_wavFileRecord;
|
|
m_wavFileRecord = nullptr;
|
|
delete m_audioOutput;
|
|
m_audioOutput = nullptr;
|
|
|
|
// if (m_audioUsageCount > 0)
|
|
// {
|
|
// m_audioUsageCount--;
|
|
//
|
|
// if (m_audioUsageCount == 0)
|
|
// {
|
|
// QMutexLocker mutexLocker(&m_mutex);
|
|
// QIODevice::close();
|
|
//
|
|
// if (!m_onExit) {
|
|
// delete m_audioOutput;
|
|
// }
|
|
// }
|
|
// }
|
|
}
|
|
|
|
void AudioOutputDevice::addFifo(AudioFifo* audioFifo)
|
|
{
|
|
QMutexLocker mutexLocker(&m_mutex);
|
|
m_audioFifos.push_back(audioFifo);
|
|
}
|
|
|
|
void AudioOutputDevice::removeFifo(AudioFifo* audioFifo)
|
|
{
|
|
QMutexLocker mutexLocker(&m_mutex);
|
|
m_audioFifos.remove(audioFifo);
|
|
}
|
|
|
|
/*
|
|
bool AudioOutputDevice::open(OpenMode mode)
|
|
{
|
|
Q_UNUSED(mode);
|
|
return false;
|
|
}*/
|
|
|
|
void AudioOutputDevice::setUdpDestination(const QString& address, uint16_t port)
|
|
{
|
|
if (m_audioNetSink) {
|
|
m_audioNetSink->setDestination(address, port);
|
|
}
|
|
}
|
|
|
|
void AudioOutputDevice::setUdpCopyToUDP(bool copyToUDP)
|
|
{
|
|
m_copyAudioToUdp = copyToUDP;
|
|
}
|
|
|
|
void AudioOutputDevice::setUdpUseRTP(bool useRTP)
|
|
{
|
|
if (m_audioNetSink) {
|
|
m_audioNetSink->selectType(useRTP ? AudioNetSink::SinkRTP : AudioNetSink::SinkUDP);
|
|
}
|
|
}
|
|
|
|
void AudioOutputDevice::setUdpChannelMode(UDPChannelMode udpChannelMode)
|
|
{
|
|
m_udpChannelMode = udpChannelMode;
|
|
}
|
|
|
|
void AudioOutputDevice::setUdpChannelFormat(UDPChannelCodec udpChannelCodec, bool stereo, int sampleRate)
|
|
{
|
|
m_udpChannelCodec = udpChannelCodec;
|
|
|
|
if (m_audioNetSink) {
|
|
m_audioNetSink->setParameters((AudioNetSink::Codec) m_udpChannelCodec, stereo, sampleRate);
|
|
}
|
|
|
|
if (m_wavFileRecord)
|
|
{
|
|
if (m_wavFileRecord->isRecording()) {
|
|
m_wavFileRecord->stopRecording();
|
|
}
|
|
|
|
m_wavFileRecord->setMono(!stereo);
|
|
}
|
|
}
|
|
|
|
void AudioOutputDevice::setUdpDecimation(uint32_t decimation)
|
|
{
|
|
if (m_audioNetSink) {
|
|
m_audioNetSink->setDecimation(decimation);
|
|
}
|
|
}
|
|
|
|
void AudioOutputDevice::setFileRecordName(const QString& fileRecordName)
|
|
{
|
|
if (!m_wavFileRecord) {
|
|
return;
|
|
}
|
|
|
|
QStringList dotBreakout = fileRecordName.split(QLatin1Char('.'));
|
|
|
|
if (dotBreakout.size() > 1) {
|
|
QString extension = dotBreakout.last();
|
|
|
|
if (extension != "wav") {
|
|
dotBreakout.last() = "wav";
|
|
}
|
|
}
|
|
else
|
|
{
|
|
dotBreakout.append("wav");
|
|
}
|
|
|
|
QString newFileRecordName = dotBreakout.join(QLatin1Char('.'));
|
|
QString fileBase;
|
|
FileRecordInterface::guessTypeFromFileName(newFileRecordName, fileBase);
|
|
qDebug("AudioOutputDevice::setFileRecordName: newFileRecordName: %s fileBase: %s", qPrintable(newFileRecordName), qPrintable(fileBase));
|
|
m_wavFileRecord->setFileName(fileBase);
|
|
}
|
|
|
|
void AudioOutputDevice::setRecordToFile(bool recordToFile)
|
|
{
|
|
if (!m_wavFileRecord) {
|
|
return;
|
|
}
|
|
|
|
if (recordToFile)
|
|
{
|
|
if (!m_wavFileRecord->isRecording()) {
|
|
m_wavFileRecord->startRecording();
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if (m_wavFileRecord->isRecording()) {
|
|
m_wavFileRecord->stopRecording();
|
|
}
|
|
}
|
|
|
|
m_recordToFile = recordToFile;
|
|
m_recordSilenceCount = 0;
|
|
}
|
|
|
|
void AudioOutputDevice::setRecordSilenceTime(int recordSilenceTime)
|
|
{
|
|
m_recordSilenceNbSamples = (recordSilenceTime * m_audioFormat.sampleRate()) / 10; // time in 100'ś ms
|
|
m_recordSilenceCount = 0;
|
|
m_recordSilenceTime = recordSilenceTime;
|
|
}
|
|
|
|
qint64 AudioOutputDevice::readData(char* data, qint64 maxLen)
|
|
{
|
|
// Study this mutex on OSX, for now deadlocks possible
|
|
// Removed as it may indeed cause lockups and is in fact useless.
|
|
//#ifndef __APPLE__
|
|
// QMutexLocker mutexLocker(&m_mutex);
|
|
//#endif
|
|
// qDebug("AudioOutputDevice::readData: thread %p (%s)", (void *) QThread::currentThread(), qPrintable(m_deviceName));
|
|
|
|
unsigned int samplesPerBuffer = maxLen / 4;
|
|
|
|
if (samplesPerBuffer == 0)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
if (m_mixBuffer.size() < samplesPerBuffer * 2)
|
|
{
|
|
m_mixBuffer.resize(samplesPerBuffer * 2); // allocate 2 qint32 per sample (stereo)
|
|
|
|
if (m_mixBuffer.size() != samplesPerBuffer * 2)
|
|
{
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
// See how much data we have available
|
|
// If we have less than the requested amount, we only output what we have
|
|
// If we have no data, then we output some zeros to avoid underflow
|
|
// (bytesAvailable() returns this amount when none available)
|
|
unsigned int samplesAvailable = bytesAvailable() / 4;
|
|
samplesPerBuffer = std::min(samplesAvailable, samplesPerBuffer);
|
|
|
|
memset(&m_mixBuffer[0], 0x00, 2 * samplesPerBuffer * sizeof(m_mixBuffer[0])); // start with silence
|
|
|
|
// sum up a block from all fifos
|
|
|
|
for (std::list<AudioFifo*>::iterator it = m_audioFifos.begin(); it != m_audioFifos.end(); ++it)
|
|
{
|
|
// use outputBuffer as temp - yes, one memcpy could be saved
|
|
unsigned int samples = (*it)->read((quint8*) data, samplesPerBuffer);
|
|
const qint16* src = (const qint16*) data;
|
|
std::vector<qint32>::iterator dst = m_mixBuffer.begin();
|
|
|
|
if (samples != samplesPerBuffer)
|
|
{
|
|
//qDebug("AudioOutputDevice::readData: read %d samples vs %d requested", samples, samplesPerBuffer);
|
|
emit (*it)->underflow();
|
|
}
|
|
|
|
for (unsigned int i = 0; i < samples; i++)
|
|
{
|
|
*dst += *src;
|
|
++src;
|
|
++dst;
|
|
*dst += *src;
|
|
++src;
|
|
++dst;
|
|
}
|
|
}
|
|
// convert to int16
|
|
|
|
//std::vector<qint32>::const_iterator src = m_mixBuffer.begin(); // Valgrind optim
|
|
qint16* dst = (qint16*) data;
|
|
qint32 sl, sr;
|
|
|
|
for (unsigned int i = 0; i < samplesPerBuffer; i++)
|
|
{
|
|
// left channel
|
|
|
|
//s = *src++; // Valgrind optim
|
|
sl = m_mixBuffer[2*i];
|
|
|
|
if(sl < -32768)
|
|
{
|
|
sl = -32768;
|
|
}
|
|
else if (sl > 32767)
|
|
{
|
|
sl = 32767;
|
|
}
|
|
|
|
*dst++ = sl;
|
|
|
|
// right channel
|
|
|
|
//s = *src++; // Valgrind optim
|
|
sr = m_mixBuffer[2*i + 1];
|
|
|
|
if(sr < -32768)
|
|
{
|
|
sr = -32768;
|
|
}
|
|
else if (sr > 32767)
|
|
{
|
|
sr = 32767;
|
|
}
|
|
|
|
*dst++ = sr;
|
|
|
|
if ((m_copyAudioToUdp) && (m_audioNetSink))
|
|
{
|
|
switch (m_udpChannelMode)
|
|
{
|
|
case UDPChannelStereo:
|
|
m_audioNetSink->write(sl, sr);
|
|
break;
|
|
case UDPChannelMixed:
|
|
m_audioNetSink->write((sl+sr)/2);
|
|
break;
|
|
case UDPChannelRight:
|
|
m_audioNetSink->write(sr);
|
|
break;
|
|
case UDPChannelLeft:
|
|
default:
|
|
m_audioNetSink->write(sl);
|
|
break;
|
|
}
|
|
}
|
|
|
|
if ((m_recordToFile) && (m_wavFileRecord))
|
|
{
|
|
if ((sr == 0) && (sl == 0))
|
|
{
|
|
if (m_recordSilenceNbSamples <= 0)
|
|
{
|
|
writeSampleToFile(sl, sr);
|
|
m_recordSilenceCount = 0;
|
|
}
|
|
else if (m_recordSilenceCount < m_recordSilenceNbSamples)
|
|
{
|
|
writeSampleToFile(sl, sr);
|
|
m_recordSilenceCount++;
|
|
}
|
|
else
|
|
{
|
|
m_wavFileRecord->stopRecording();
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if (!m_wavFileRecord->isRecording()) {
|
|
m_wavFileRecord->startRecording();
|
|
}
|
|
|
|
writeSampleToFile(sl, sr);
|
|
m_recordSilenceCount = 0;
|
|
}
|
|
}
|
|
}
|
|
|
|
return samplesPerBuffer * 4;
|
|
}
|
|
|
|
void AudioOutputDevice::writeSampleToFile(qint16 lSample, qint16 rSample)
|
|
{
|
|
switch (m_udpChannelMode)
|
|
{
|
|
case UDPChannelStereo:
|
|
m_wavFileRecord->write(lSample, rSample);
|
|
break;
|
|
case UDPChannelMixed:
|
|
m_wavFileRecord->writeMono((lSample+rSample)/2);
|
|
break;
|
|
case UDPChannelRight:
|
|
m_wavFileRecord->writeMono(rSample);
|
|
break;
|
|
case UDPChannelLeft:
|
|
default:
|
|
m_wavFileRecord->writeMono(lSample);
|
|
break;
|
|
}
|
|
}
|
|
|
|
qint64 AudioOutputDevice::writeData(const char* data, qint64 len)
|
|
{
|
|
Q_UNUSED(data);
|
|
Q_UNUSED(len);
|
|
return 0;
|
|
}
|
|
|
|
void AudioOutputDevice::setVolume(float volume)
|
|
{
|
|
m_volume = volume;
|
|
|
|
if (m_audioOutput) {
|
|
m_audioOutput->setVolume(m_volume);
|
|
}
|
|
}
|
|
|
|
// Qt6 requires bytesAvailable to be implemented. Not needed for Qt5.
|
|
qint64 AudioOutputDevice::bytesAvailable() const
|
|
{
|
|
qint64 available = 0;
|
|
for (std::list<AudioFifo*>::const_iterator it = m_audioFifos.begin(); it != m_audioFifos.end(); ++it)
|
|
{
|
|
qint64 fill = (*it)->fill();
|
|
if (available == 0) {
|
|
available = fill;
|
|
} else {
|
|
available = std::min(fill, available);
|
|
}
|
|
}
|
|
// If we return 0 from this twice in a row, audio will stop.
|
|
// So we always return a value, and if we don't have enough data in the FIFOs
|
|
// when readData is called, that will output silence
|
|
if (available == 0)
|
|
{
|
|
// Use a small value, so padding is minimized, but not too small, we get underflow again straight away
|
|
// Could make this a function of sample rate
|
|
available = 512;
|
|
}
|
|
return available * 2 * 2; // 2 Channels of 16-bit data
|
|
}
|
|
|
|
bool AudioOutputDevice::handleMessage(const Message& cmd)
|
|
{
|
|
if (MsgStart::match(cmd))
|
|
{
|
|
MsgStart ctl = (MsgStart&) cmd;
|
|
start(ctl.getDeviceIndex(), ctl.getSampleRate());
|
|
return true;
|
|
}
|
|
else if (MsgStop::match(cmd))
|
|
{
|
|
stop();
|
|
return true;
|
|
}
|
|
|
|
return false;
|
|
}
|
|
|
|
void AudioOutputDevice::handleInputMessages()
|
|
{
|
|
Message* message;
|
|
|
|
while ((message = m_inputMessageQueue.pop()) != nullptr)
|
|
{
|
|
if (handleMessage(*message)) {
|
|
delete message;
|
|
}
|
|
}
|
|
}
|