qint64 audioLength(const QAudioFormat &format, qint64 microSeconds)
{
qint64 result = (format.sampleRate() * format.channelCount() * (format.sampleSize() / 8))
* microSeconds / 1000000;
result -= result % (format.channelCount() * format.sampleSize());
return result;
}
void Generator::generateData(const QAudioFormat &format, qint64 durationUs, int sampleRate)
{
const int channelBytes = format.sampleSize() / 8;
qint64 length = (format.sampleRate() * format.channelCount() * (format.sampleSize() / 8))
* durationUs / 1000000;
m_buffer.resize(length);
unsigned char *ptr = reinterpret_cast<unsigned char *>(m_buffer.data());
int sampleIndex = 0;
PHDBG() << "Type :" << format.sampleType() << " Size : " << format.sampleSize() << " Channel Count : " << format.channelCount();
while (length) {
const qreal x = qSin(2 * M_PI * sampleRate * qreal(sampleIndex % format.sampleRate()) / format.sampleRate());
for (int i = 0; i < format.channelCount(); ++i) {
qint16 value = static_cast<qint16>(x * 32767);
qToLittleEndian<qint16>(value, ptr);
ptr += channelBytes;
length -= channelBytes;
}
++sampleIndex;
}
}
bool WaveFileWriter::writeHeader(const QAudioFormat &format)
{
// check if format is supported
if (format.byteOrder() == QAudioFormat::BigEndian || format.sampleType() != QAudioFormat::SignedInt)
return false;
CombinedHeader header;
memset(&header, 0, HeaderLength);
#ifndef Q_LITTLE_ENDIAN
// only implemented for LITTLE ENDIAN
return false;
#else
// RIFF header
memcpy(header.riff.descriptor.id, "RIFF", 4);
header.riff.descriptor.size = 0; // this will be updated with correct duration:
// m_dataLength + HeaderLength - 8
// WAVE header
memcpy(header.riff.type, "WAVE", 4);
memcpy(header.wave.descriptor.id, "fmt ", 4);
header.wave.descriptor.size = quint32(16);
header.wave.audioFormat = quint16(1);
header.wave.numChannels = quint16(format.channelCount());
header.wave.sampleRate = quint32(format.sampleRate());
header.wave.byteRate = quint32(format.sampleRate() * format.channelCount() * format.sampleSize() / 8);
header.wave.blockAlign = quint16(format.channelCount() * format.sampleSize() / 8);
header.wave.bitsPerSample = quint16(format.sampleSize());
// DATA header
memcpy(header.data.descriptor.id,"data", 4);
header.data.descriptor.size = 0; // this will be updated with correct data length: m_dataLength
return (file.write(reinterpret_cast<const char *>(&header), HeaderLength) == HeaderLength);
#endif
}
// Returns position in bytes, given position in microSeconds.
qint64 audioLength(const QAudioFormat &format, qint64 microSeconds)
{
// format.sampleRate() is in Hz, format.sampleSize() in bits
qint64 result = (format.sampleRate() * format.channelCount() * (format.sampleSize() / 8))
* microSeconds / 1000000;
// Round off to start of the channel 0.
result -= result % (format.channelCount() * format.sampleSize());
return result;
}
void Generator::generateData(const QAudioFormat &format, qint64 durationUs, int sampleRate)
{
const int channelBytes = format.sampleSize() / 8;//разрядность
const int sampleBytes = format.channelCount() * channelBytes;//количество байт умнож на кол-во каналов
qint64 length = (format.sampleRate() * format.channelCount() * (format.sampleSize() / 8))
* durationUs / 1000000;
Q_ASSERT(length % sampleBytes == 0);
Q_UNUSED(sampleBytes) // suppress warning in release builds
m_buffer.resize(length);
unsigned char *ptr = reinterpret_cast<unsigned char *>(m_buffer.data());
int sampleIndex = 0;
while (length)
{
const qreal x = qSin(2 * M_PI * sampleRate * qreal(sampleIndex % format.sampleRate()) / format.sampleRate());
for (int i=0; i<format.channelCount(); ++i)
{
if (format.sampleSize() == 8 && format.sampleType() == QAudioFormat::UnSignedInt)
{
const quint8 value = static_cast<quint8>((1.0 + x) / 2 * 255);
*reinterpret_cast<quint8*>(ptr) = value;
}
else if (format.sampleSize() == 8 && format.sampleType() == QAudioFormat::SignedInt)
{
const qint8 value = static_cast<qint8>(x * 127);
*reinterpret_cast<quint8*>(ptr) = value;
}
else if (format.sampleSize() == 16 && format.sampleType() == QAudioFormat::UnSignedInt)
{
quint16 value = static_cast<quint16>((1.0 + x) / 2 * 65535);
if (format.byteOrder() == QAudioFormat::LittleEndian)
qToLittleEndian<quint16>(value, ptr);
else
qToBigEndian<quint16>(value, ptr);
}
else if (format.sampleSize() == 16 && format.sampleType() == QAudioFormat::SignedInt)
{
qint16 value = static_cast<qint16>(x * 32767);
if (format.byteOrder() == QAudioFormat::LittleEndian)
qToLittleEndian<qint16>(value, ptr);
else
qToBigEndian<qint16>(value, ptr);
}
ptr += channelBytes;//кол-во каналов
length -= channelBytes;
}
++sampleIndex;
}
}
QString formatToString(const QAudioFormat &format)
{
QString result;
if (QAudioFormat() != format) {
if (format.codec() == "audio/pcm") {
Q_ASSERT(format.sampleType() == QAudioFormat::SignedInt);
const QString formatEndian = (format.byteOrder() == QAudioFormat::LittleEndian)
? QString("LE") : QString("BE");
QString formatType;
switch (format.sampleType()) {
case QAudioFormat::SignedInt:
formatType = "signed";
break;
case QAudioFormat::UnSignedInt:
formatType = "unsigned";
break;
case QAudioFormat::Float:
formatType = "float";
break;
case QAudioFormat::Unknown:
formatType = "unknown";
break;
}
QString formatChannels = QString("%1 channels").arg(format.channelCount());
switch (format.channelCount()) {
case 1:
formatChannels = "mono";
break;
case 2:
formatChannels = "stereo";
break;
}
result = QString("%1 Hz %2 bit %3 %4 %5")
.arg(format.sampleRate())
.arg(format.sampleSize())
.arg(formatType)
.arg(formatEndian)
.arg(formatChannels);
} else {
result = format.codec();
}
}
return result;
}
ViAudioFormat::ViAudioFormat(const QAudioFormat &other)
{
if(other.sampleType() == QAudioFormat::SignedInt)
{
setSampleType(ViAudioFormat::SignedInt);
}
else if(other.sampleType() == QAudioFormat::UnSignedInt)
{
setSampleType(ViAudioFormat::UnSignedInt);
}
else if(other.sampleType() == QAudioFormat::Float)
{
setSampleType(ViAudioFormat::Float);
}
else
{
setSampleSize(ViAudioFormat::Unknown);
}
if(other.byteOrder() == QAudioFormat::BigEndian)
{
setByteOrder(ViAudioFormat::BigEndian);
}
else
{
setByteOrder(ViAudioFormat::LittleEndian);
}
mQuality = ViAudioFormat::Average;
mSampleSize = other.sampleSize();
mSampleRate = other.sampleRate();
mChannelCount = other.channelCount();
mCodec = NULL;
}
void WaveRenderArea::setAudioFormat(const QAudioFormat &format)
{
Q_D(WaveRenderArea);
Q_ASSERT(format.channelCount() == 1);
d->audioFormat = format;
d->maxAmplitude = AudioInputDevice::maxAmplitudeForFormat(d->audioFormat);
}
void AudioReflector::processLocalAudio(unsigned int sampleTime, const QByteArray& samples, const QAudioFormat& format) {
bool processLocalAudio = true; // Menu::getInstance()->isOptionChecked(MenuOption::AudioSpatialProcessingProcessLocalAudio)
if (processLocalAudio) {
const int NUM_CHANNELS_INPUT = 1;
const int NUM_CHANNELS_OUTPUT = 2;
const int EXPECTED_SAMPLE_RATE = 24000;
if (format.channelCount() == NUM_CHANNELS_INPUT && format.sampleRate() == EXPECTED_SAMPLE_RATE) {
QAudioFormat outputFormat = format;
outputFormat.setChannelCount(NUM_CHANNELS_OUTPUT);
QByteArray stereoInputData(samples.size() * NUM_CHANNELS_OUTPUT, 0);
int numberOfSamples = (samples.size() / sizeof(int16_t));
int16_t* monoSamples = (int16_t*)samples.data();
int16_t* stereoSamples = (int16_t*)stereoInputData.data();
for (int i = 0; i < numberOfSamples; i++) {
stereoSamples[i* NUM_CHANNELS_OUTPUT] = monoSamples[i] * _localAudioAttenuationFactor;
stereoSamples[(i * NUM_CHANNELS_OUTPUT) + 1] = monoSamples[i] * _localAudioAttenuationFactor;
}
_localAudioDelays.clear();
_localEchoesSuppressed.clear();
echoAudio(LOCAL_AUDIO, sampleTime, stereoInputData, outputFormat);
_localEchoesCount = _localAudioDelays.size();
_localEchoesSuppressedCount = _localEchoesSuppressed.size();
}
}
}
AudioStreamBasicDescription toAudioStreamBasicDescription(QAudioFormat const& audioFormat)
{
AudioStreamBasicDescription sf;
sf.mFormatFlags = kAudioFormatFlagIsPacked;
sf.mSampleRate = audioFormat.sampleRate();
sf.mFramesPerPacket = 1;
sf.mChannelsPerFrame = audioFormat.channelCount();
sf.mBitsPerChannel = audioFormat.sampleSize();
sf.mBytesPerFrame = sf.mChannelsPerFrame * (sf.mBitsPerChannel / 8);
sf.mBytesPerPacket = sf.mFramesPerPacket * sf.mBytesPerFrame;
sf.mFormatID = kAudioFormatLinearPCM;
switch (audioFormat.sampleType()) {
case QAudioFormat::SignedInt:
sf.mFormatFlags |= kAudioFormatFlagIsSignedInteger;
break;
case QAudioFormat::UnSignedInt: /* default */
break;
case QAudioFormat::Float:
sf.mFormatFlags |= kAudioFormatFlagIsFloat;
break;
case QAudioFormat::Unknown:
default:
break;
}
if (audioFormat.byteOrder() == QAudioFormat::BigEndian)
sf.mFormatFlags |= kAudioFormatFlagIsBigEndian;
return sf;
}
/* channelCount() API property test. */
void tst_QAudioFormat::checkChannelCount()
{
// channels is the old name for channelCount, so
// they should always be equal
QAudioFormat audioFormat;
QVERIFY(audioFormat.channelCount() == -1);
QVERIFY(audioFormat.channels() == -1);
audioFormat.setChannelCount(123);
QVERIFY(audioFormat.channelCount() == 123);
QVERIFY(audioFormat.channels() == 123);
audioFormat.setChannels(5);
QVERIFY(audioFormat.channelCount() == 5);
QVERIFY(audioFormat.channels() == 5);
}
pa_sample_spec audioFormatToSampleSpec(const QAudioFormat &format)
{
pa_sample_spec spec;
spec.rate = format.sampleRate();
spec.channels = format.channelCount();
if (format.sampleSize() == 8) {
spec.format = PA_SAMPLE_U8;
} else if (format.sampleSize() == 16) {
switch (format.byteOrder()) {
case QAudioFormat::BigEndian: spec.format = PA_SAMPLE_S16BE; break;
case QAudioFormat::LittleEndian: spec.format = PA_SAMPLE_S16LE; break;
}
} else if (format.sampleSize() == 24) {
switch (format.byteOrder()) {
case QAudioFormat::BigEndian: spec.format = PA_SAMPLE_S24BE; break;
case QAudioFormat::LittleEndian: spec.format = PA_SAMPLE_S24LE; break;
}
} else if (format.sampleSize() == 32) {
switch (format.byteOrder()) {
case QAudioFormat::BigEndian:
format.sampleType() == QAudioFormat::Float ? spec.format = PA_SAMPLE_FLOAT32BE : spec.format = PA_SAMPLE_S32BE;
break;
case QAudioFormat::LittleEndian:
format.sampleType() == QAudioFormat::Float ? spec.format = PA_SAMPLE_FLOAT32LE : spec.format = PA_SAMPLE_S32LE;
break;
}
} else {
spec.format = PA_SAMPLE_INVALID;
}
return spec;
}
void xmppClient::slotAudioModeChanged(QIODevice::OpenMode mode)
{
QXmppCall *call = qobject_cast<QXmppCall*>(sender());
Q_ASSERT(call);
QXmppRtpAudioChannel *channel = call->audioChannel();
// prepare audio format
QAudioFormat format;
format.setSampleRate(channel->payloadType().clockrate());
format.setChannelCount(channel->payloadType().channels());
format.setSampleSize(16);
format.setCodec("audio/pcm");
format.setByteOrder(QAudioFormat::LittleEndian);
format.setSampleType(QAudioFormat::SignedInt);
// the size in bytes of the audio buffers to/from sound devices
// 160 ms seems to be the minimum to work consistently on Linux/Mac/Windows
const int bufferSize = (format.sampleRate() * format.channelCount() * (format.sampleSize() / 8) * 160) / 1000;
if (mode & QIODevice::ReadOnly) {
// initialise audio output
QAudioOutput *audioOutput = new QAudioOutput(format, this);
audioOutput->setBufferSize(bufferSize);
audioOutput->start(channel);
}
if (mode & QIODevice::WriteOnly) {
// initialise audio input
QAudioInput *audioInput = new QAudioInput(format, this);
audioInput->setBufferSize(bufferSize);
audioInput->start(channel);
}
}
SLDataFormat_PCM QOpenSLESEngine::audioFormatToSLFormatPCM(const QAudioFormat &format)
{
SLDataFormat_PCM format_pcm;
format_pcm.formatType = SL_DATAFORMAT_PCM;
format_pcm.numChannels = format.channelCount();
format_pcm.samplesPerSec = format.sampleRate() * 1000;
format_pcm.bitsPerSample = format.sampleSize();
format_pcm.containerSize = format.sampleSize();
format_pcm.channelMask = (format.channelCount() == 1 ?
SL_SPEAKER_FRONT_CENTER :
SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT);
format_pcm.endianness = (format.byteOrder() == QAudioFormat::LittleEndian ?
SL_BYTEORDER_LITTLEENDIAN :
SL_BYTEORDER_BIGENDIAN);
return format_pcm;
}
void Generator::generateData(const QAudioFormat &format, qint64 durationUs, int frequency)
{
const int channelBytes = format.sampleSize() / 8;
const int sampleBytes = format.channelCount() * channelBytes;
qint64 length = (format.sampleRate() * format.channelCount() * (format.sampleSize() / 8))
* durationUs / 100000;
Q_ASSERT(length % sampleBytes == 0);
Q_UNUSED(sampleBytes) // suppress warning in release builds
m_buffer.resize(length);
unsigned char *ptr = reinterpret_cast<unsigned char *>(m_buffer.data());
generateData(format, ptr, length);
}
void tst_QWaveDecoder::file()
{
QFETCH(QString, file);
QFETCH(tst_QWaveDecoder::Corruption, corruption);
QFETCH(int, channels);
QFETCH(int, samplesize);
QFETCH(int, samplerate);
QFETCH(QAudioFormat::Endian, byteorder);
QFile stream;
stream.setFileName(file);
stream.open(QIODevice::ReadOnly);
QVERIFY(stream.isOpen());
QWaveDecoder waveDecoder(&stream);
QSignalSpy validFormatSpy(&waveDecoder, SIGNAL(formatKnown()));
QSignalSpy parsingErrorSpy(&waveDecoder, SIGNAL(parsingError()));
if (corruption == NotAWav) {
QSKIP("Not all failures detected correctly yet");
QTRY_COMPARE(parsingErrorSpy.count(), 1);
QCOMPARE(validFormatSpy.count(), 0);
} else if (corruption == NoSampleData) {
QTRY_COMPARE(validFormatSpy.count(), 1);
QCOMPARE(parsingErrorSpy.count(), 0);
QVERIFY(waveDecoder.audioFormat().isValid());
QVERIFY(waveDecoder.size() == 0);
QVERIFY(waveDecoder.duration() == 0);
} else if (corruption == FormatDescriptor) {
QTRY_COMPARE(parsingErrorSpy.count(), 1);
QCOMPARE(validFormatSpy.count(), 0);
} else if (corruption == FormatString) {
QTRY_COMPARE(parsingErrorSpy.count(), 1);
QCOMPARE(validFormatSpy.count(), 0);
QVERIFY(!waveDecoder.audioFormat().isValid());
} else if (corruption == DataDescriptor) {
QTRY_COMPARE(parsingErrorSpy.count(), 1);
QCOMPARE(validFormatSpy.count(), 0);
QVERIFY(waveDecoder.size() == 0);
} else if (corruption == None) {
QTRY_COMPARE(validFormatSpy.count(), 1);
QCOMPARE(parsingErrorSpy.count(), 0);
QVERIFY(waveDecoder.audioFormat().isValid());
QVERIFY(waveDecoder.size() > 0);
QVERIFY(waveDecoder.duration() == 250);
QAudioFormat format = waveDecoder.audioFormat();
QVERIFY(format.isValid());
QVERIFY(format.channelCount() == channels);
QVERIFY(format.sampleSize() == samplesize);
QVERIFY(format.sampleRate() == samplerate);
if (format.sampleSize() != 8) {
QVERIFY(format.byteOrder() == byteorder);
}
}
stream.close();
}
void SpectrumAnalyser::calculate(const QByteArray &buffer,
const QAudioFormat &format)
{
// QThread::currentThread is marked 'for internal use only', but
// we're only using it for debug output here, so it's probably OK :)
SPECTRUMANALYSER_DEBUG << "SpectrumAnalyser::calculate"
<< QThread::currentThread()
<< "state" << m_state;
if (isReady()) {
Q_ASSERT(isPCMS16LE(format));
const int bytesPerSample = format.sampleSize() * format.channelCount() / 8;
#ifdef DUMP_SPECTRUMANALYSER
m_count++;
const QString pcmFileName = m_outputDir.filePath(QString("spectrum_%1.pcm").arg(m_count, 4, 10, QChar('0')));
QFile pcmFile(pcmFileName);
pcmFile.open(QIODevice::WriteOnly);
const int bufferLength = m_numSamples * bytesPerSample;
pcmFile.write(buffer, bufferLength);
m_textStream << "TimeDomain " << m_count << "\n";
const qint16* input = reinterpret_cast<const qint16*>(buffer);
for (int i=0; i<m_numSamples; ++i) {
m_textStream << i << "\t" << *input << "\n";
input += format.channels();
}
#endif
m_state = Busy;
// Invoke SpectrumAnalyserThread::calculateSpectrum using QMetaObject. If
// m_thread is in a different thread from the current thread, the
// calculation will be done in the child thread.
// Once the calculation is finished, a calculationChanged signal will be
// emitted by m_thread.
const bool b = QMetaObject::invokeMethod(m_thread, "calculateSpectrum",
Qt::AutoConnection,
Q_ARG(QByteArray, buffer),
Q_ARG(int, format.sampleRate()),
Q_ARG(int, bytesPerSample));
Q_ASSERT(b);
Q_UNUSED(b) // suppress warnings in release builds
#ifdef DUMP_SPECTRUMANALYSER
m_textStream << "FrequencySpectrum " << m_count << "\n";
FrequencySpectrum::const_iterator x = m_spectrum.begin();
for (int i=0; i<m_numSamples; ++i, ++x)
m_textStream << i << "\t"
<< x->frequency << "\t"
<< x->amplitude<< "\t"
<< x->phase << "\n";
#endif
}
}
bool QAudioDeviceInfoInternal::isFormatSupported(const QAudioFormat& format) const
{
QAudioDeviceInfoInternal *self = const_cast<QAudioDeviceInfoInternal*>(this);
return format.isValid()
&& format.codec() == QString::fromLatin1("audio/pcm")
&& self->supportedSampleRates().contains(format.sampleRate())
&& self->supportedChannelCounts().contains(format.channelCount())
&& self->supportedSampleSizes().contains(format.sampleSize());
}
void TSoundOutputDeviceImp::play(const TSoundTrackP &st, TINT32 s0, TINT32 s1,
bool loop, bool scrubbing) {
if (!doSetStreamFormat(st->getFormat())) return;
MyData *myData = new MyData();
myData->imp = shared_from_this();
myData->totalPacketCount = s1 - s0;
myData->fileByteCount = (s1 - s0) * st->getSampleSize();
myData->entireFileBuffer = new char[myData->fileByteCount];
memcpy(myData->entireFileBuffer, st->getRawData() + s0 * st->getSampleSize(),
myData->fileByteCount);
m_isPlaying = true;
myData->isLooping = loop;
QAudioFormat format;
QAudioDeviceInfo info(QAudioDeviceInfo::defaultOutputDevice());
format.setSampleSize(st->getBitPerSample());
format.setCodec("audio/pcm");
format.setChannelCount(st->getChannelCount());
format.setByteOrder(QAudioFormat::LittleEndian);
format.setSampleType(st->getFormat().m_signedSample
? QAudioFormat::SignedInt
: QAudioFormat::UnSignedInt);
format.setSampleRate(st->getSampleRate());
QList<QAudioFormat::Endian> sbos = info.supportedByteOrders();
QList<int> sccs = info.supportedChannelCounts();
QList<int> ssrs = info.supportedSampleRates();
QList<QAudioFormat::SampleType> sstypes = info.supportedSampleTypes();
QList<int> ssss = info.supportedSampleSizes();
QStringList supCodes = info.supportedCodecs();
if (!info.isFormatSupported((format))) {
format = info.nearestFormat(format);
int newChannels = format.channelCount();
int newBitsPerSample = format.sampleSize();
int newSampleRate = format.sampleRate();
QAudioFormat::SampleType newSampleType = format.sampleType();
QAudioFormat::Endian newBo = format.byteOrder();
}
int test = st->getSampleCount() / st->getSampleRate();
QByteArray *data =
new QByteArray(myData->entireFileBuffer, myData->fileByteCount);
QBuffer *newBuffer = new QBuffer;
newBuffer->setBuffer(data);
newBuffer->open(QIODevice::ReadOnly);
newBuffer->seek(0);
if (m_audioOutput == NULL) {
m_audioOutput = new QAudioOutput(format, NULL);
}
m_audioOutput->start(newBuffer);
m_audioOutput->setVolume(m_volume);
}
QDebug operator<<(QDebug dbg, const QAudioFormat &f)
{
dbg.nospace() << "QAudioFormat(" << f.sampleRate();
dbg.nospace() << "Hz, " << f.sampleSize();
dbg.nospace() << "bit, channelCount=" << f.channelCount();
dbg.nospace() << ", sampleType=" << f.sampleType();
dbg.nospace() << ", byteOrder=" << f.byteOrder();
dbg.nospace() << ", codec=" << f.codec();
dbg.nospace() << ")";
return dbg.space();
}
bool Lockin2::isFormatSupported(const QAudioFormat &format)
{
if (format.codec() != "audio/pcm") {
return false;
}
if (format.channelCount() != 2) {
return false;
}
return true;
}
CircularBuffer::CircularBuffer(const QAudioFormat& format, qint64 samples, QObject* parent)
: QIODevice(parent),
m_format(format),
m_bytesPerSample((format.sampleSize() / 8) * format.channelCount())
{
m_readPtr = 0;
m_writePtr = 0;
m_bufferSize = 0;
m_bufferCapacity = samples;
m_buffer.resize(m_bufferCapacity);
m_buffer.fill(0);
open(QIODevice::ReadWrite);
}
void AudioFileWav::addHeader(const QAudioFormat& audioFormat) {
QDataStream stream(&_file);
stream.setByteOrder(QDataStream::LittleEndian);
// RIFF
stream.writeRawData("RIFF", 4);
stream << quint32(0);
stream.writeRawData("WAVE", 4);
// Format description PCM = 16
stream.writeRawData("fmt ", 4);
stream << quint32(16);
stream << quint16(1);
stream << quint16(audioFormat.channelCount());
stream << quint32(audioFormat.sampleRate());
stream << quint32(audioFormat.sampleRate() * audioFormat.channelCount() * audioFormat.sampleSize() / 8); // bytes per second
stream << quint16(audioFormat.channelCount() * audioFormat.sampleSize() / 8); // block align
stream << quint16(audioFormat.sampleSize()); // bits Per Sample
// Init data chunck
stream.writeRawData("data", 4);
stream << quint32(0);
}
/*!
Returns the closest QAudioFormat to the supplied \a settings that the system supports.
These settings are provided by the platform/audio plugin being used.
They are also dependent on the \l {QAudio}::Mode being used.
*/
QAudioFormat QAudioDeviceInfo::nearestFormat(const QAudioFormat &settings) const
{
if (isFormatSupported(settings))
return settings;
QAudioFormat nearest = settings;
QList<QString> testCodecs = supportedCodecs();
QList<int> testChannels = supportedChannelCounts();
QList<QAudioFormat::Endian> testByteOrders = supportedByteOrders();
QList<QAudioFormat::SampleType> testSampleTypes;
QList<QAudioFormat::SampleType> sampleTypesAvailable = supportedSampleTypes();
QMap<int,int> testSampleRates;
QList<int> sampleRatesAvailable = supportedSampleRates();
QMap<int,int> testSampleSizes;
QList<int> sampleSizesAvailable = supportedSampleSizes();
// Get sorted lists for checking
if (testCodecs.contains(settings.codec())) {
testCodecs.removeAll(settings.codec());
testCodecs.insert(0, settings.codec());
}
testChannels.removeAll(settings.channelCount());
testChannels.insert(0, settings.channelCount());
testByteOrders.removeAll(settings.byteOrder());
testByteOrders.insert(0, settings.byteOrder());
if (sampleTypesAvailable.contains(settings.sampleType()))
testSampleTypes.append(settings.sampleType());
if (sampleTypesAvailable.contains(QAudioFormat::SignedInt))
testSampleTypes.append(QAudioFormat::SignedInt);
if (sampleTypesAvailable.contains(QAudioFormat::UnSignedInt))
testSampleTypes.append(QAudioFormat::UnSignedInt);
if (sampleTypesAvailable.contains(QAudioFormat::Float))
testSampleTypes.append(QAudioFormat::Float);
if (sampleSizesAvailable.contains(settings.sampleSize()))
testSampleSizes.insert(0,settings.sampleSize());
sampleSizesAvailable.removeAll(settings.sampleSize());
foreach (int size, sampleSizesAvailable) {
int larger = (size > settings.sampleSize()) ? size : settings.sampleSize();
int smaller = (size > settings.sampleSize()) ? settings.sampleSize() : size;
bool isMultiple = ( 0 == (larger % smaller));
int diff = larger - smaller;
testSampleSizes.insert((isMultiple ? diff : diff+100000), size);
}
qint64 AudioOutput::currentSerialPlayedUsec()
{
if(serial != aw->serial){
this->reset();
//debug
audioDevice = this->start();
serial = aw->serial;
return 0;
}
qint64 currentSerialProcessedUsec = this->processedUSecs();
int bytesInBuffer = this->bufferSize() - this->bytesFree();
QAudioFormat format = this->format();
int sampleSizes = format.sampleSize();
int channels = format.channelCount();
int sampleRates = format.sampleRate();
qint64 playedUsec = currentSerialProcessedUsec - (qint64)(1000000) * 8 * bytesInBuffer/sampleSizes/channels/sampleRates;
return playedUsec;
}
void SoundEngine::sweepInputFile(const qint64 &audioLength, const QAudioFormat &format)
{
emit reset();
int channelCount=format.channelCount();
QByteArray startBuffer(4410*channelCount,0);
m_backingTrack->m_inputFile->seek(m_backingTrack->m_inputFile->headerLength());
while(!m_backingTrack->m_inputFile->atEnd())
{
m_backingTrack->m_inputFile->read(startBuffer.data(),3528*channelCount);
m_backingTrack->calculateLevel(startBuffer,3528*channelCount);
emit drawingBackEnabled(m_backingTrack->m_level);
}
emit timeLinePosition(0);
}
void Generator::generateData(const QAudioFormat& format, unsigned char* ptr, qint64 length)
{
const int channelBytes = format.sampleSize() / 8;
if (length <= 0)
return;
QMutexLocker lock(m_mutex);
QVector<qreal> channels;
channels.resize(2);
// LOG_INFO() << "generating Sounds" << QDateTime::currentDateTime() << length;
initializeSounds();
while (length) {
generateTone(channels[0], channels[1], m_sampleIndex);
//const qreal x = soundFunc(2 * M_PI * frequency * qreal(sampleIndex ) / format.sampleRate());
for (int i=0; i<format.channelCount(); ++i) {
if (format.sampleSize() == 8 && format.sampleType() == QAudioFormat::UnSignedInt) {
const quint8 value = static_cast<quint8>((1.0 + channels[i]) / 2 * 255);
*reinterpret_cast<quint8*>(ptr) = value;
} else if (format.sampleSize() == 8 && format.sampleType() == QAudioFormat::SignedInt) {
const qint8 value = static_cast<qint8>(channels[i] * 127);
*reinterpret_cast<quint8*>(ptr) = value;
} else if (format.sampleSize() == 16 && format.sampleType() == QAudioFormat::UnSignedInt) {
quint16 value = static_cast<quint16>((1.0 + channels[i]) / 2 * 65535);
if (format.byteOrder() == QAudioFormat::LittleEndian)
qToLittleEndian<quint16>(value, ptr);
else
qToBigEndian<quint16>(value, ptr);
} else if (format.sampleSize() == 16 && format.sampleType() == QAudioFormat::SignedInt) {
qint16 value = static_cast<qint16>(channels[i] * 32767);
if (format.byteOrder() == QAudioFormat::LittleEndian)
qToLittleEndian<qint16>(value, ptr);
else
qToBigEndian<qint16>(value, ptr);
}
ptr += channelBytes;
length -= channelBytes;
}
++m_sampleIndex;
}
}
void TaudioOUT::createOutputDevice() {
m_deviceInfo = QAudioDeviceInfo::defaultOutputDevice();
QList<QAudioDeviceInfo> devList = QAudioDeviceInfo::availableDevices(QAudio::AudioOutput);
for (int i = 0; i < devList.size(); ++i) { // find device with name or keep default one
if (devList[i].deviceName() == m_audioParams->OUTdevName) {
m_deviceInfo = devList[i];
break;
}
}
m_devName = m_deviceInfo.deviceName();
QAudioFormat format;
format.setChannelCount(1);
format.setSampleRate(m_sampleRate);
format.setSampleType(QAudioFormat::SignedInt);
format.setSampleSize(16);
format.setCodec("audio/pcm");
format.setByteOrder(QAudioFormat::LittleEndian);
if (!m_deviceInfo.isFormatSupported(format)) {
qDebug() << "Output Format 44100/16 stereo is not supported";
format = m_deviceInfo.nearestFormat(format);
qDebug() << "Format is" << format.sampleRate() << format.channelCount() << format.sampleSize();
}
m_sampleRate = format.sampleRate();
if (m_audioOUT) {
delete m_audioOUT;
delete m_buffer;
}
m_audioOUT = new QAudioOutput(m_deviceInfo, format, this);
m_audioOUT->setBufferSize(m_bufferFrames * 2);
m_buffer = new TaudioBuffer(this);
m_buffer->open(QIODevice::ReadOnly);
m_buffer->setBufferSize(m_audioOUT->bufferSize());
m_maxSamples = m_sampleRate * 1.5;
qDebug() << "OUT:" << m_deviceInfo.deviceName() << m_audioOUT->format().sampleRate() << m_maxSamples;
connect(m_buffer, &TaudioBuffer::feedAudio, this, &TaudioOUT::outCallBack, Qt::DirectConnection);
connect(m_audioOUT, &QAudioOutput::stateChanged, this, &TaudioOUT::stateChangedSlot);
}
void SpectrumAnalyser::calculate(const QByteArray &buffer,
const QAudioFormat &format)
{
// QThread::currentThread is marked 'for internal use only', but
// we're only using it for debug output here, so it's probably OK :)
SPECTRUMANALYSER_DEBUG << "SpectrumAnalyser::calculate"
<< QThread::currentThread()
<< "state" << m_state;
SPECTRUMANALYSER_DEBUG << "buffer size =" << buffer.size();
if (isReady()) {
Q_ASSERT(isPCMS16LE(format));
const int bytesPerSample = format.sampleSize() * format.channelCount() / 8;
m_state = Busy;
// Invoke SpectrumAnalyserThread::calculateSpectrum using QMetaObject. If
// m_thread is in a different thread from the current thread, the
// calculation will be done in the child thread.
// Once the calculation is finished, a calculationChanged signal will be
// emitted by m_thread.
const bool b = QMetaObject::invokeMethod(m_thread, "calculateSpectrum",
Qt::AutoConnection,
Q_ARG(QByteArray, buffer),
Q_ARG(int, format.sampleRate()),
Q_ARG(int, bytesPerSample));
Q_ASSERT(b);
Q_UNUSED(b) // suppress warnings in release builds
#ifdef DUMP_SPECTRUMANALYSER
m_textStream << "FrequencySpectrum " << m_count << "\n";
FrequencySpectrum::const_iterator x = m_spectrum.begin();
for (int i=0; i<m_numSamples; ++i, ++x)
m_textStream << i << "\t"
<< x->frequency << "\t"
<< x->amplitude<< "\t"
<< x->phase << "\n";
#endif
}
}
TaudioIN::TaudioIN(QObject *parent) :
QObject(parent),
m_inDevice(0),
m_buffer(0),
m_pitchFinder(0)
{
QList<QAudioDeviceInfo> devList = QAudioDeviceInfo::availableDevices(QAudio::AudioInput);
for (int i = 0; i < devList.count(); ++i)
qDebug() << i << devList[i].deviceName();
QAudioDeviceInfo defaultIn = QAudioDeviceInfo::defaultInputDevice();
QAudioFormat format;
format.setChannelCount(1);
format.setSampleRate(48000);
format.setSampleType(QAudioFormat::SignedInt);
format.setSampleSize(16);
format.setCodec("audio/pcm");
format.setByteOrder(QAudioFormat::LittleEndian);
if (!defaultIn.isFormatSupported(format)) {
qDebug() << "Format 48000/16 mono is not suported";
format = defaultIn.nearestFormat(format);
qDebug() << "Format is" << format.sampleRate() << format.channelCount() << format.sampleSize();
}
m_lastNote = new Tnote();
m_audioIN = new QAudioInput(defaultIn, format, this);
m_audioIN->setBufferSize(2048);
m_pitchFinder = new TpitchFinder();
m_pitchFinder->setMinimalDuration(0.1f);
m_pitchFinder->setSplitByVolChange(false);
m_pitchFinder->setSplitVolume(0.0);
m_pitchFinder->setSkipStillerVal(0.0);
m_pitchFinder->aGl()->equalLoudness = true;
m_pitchFinder->setSampleRate(m_audioIN->format().sampleRate()); // framesPerChunk is determined here
connect(m_pitchFinder, &TpitchFinder::pitchInChunk, this, &TaudioIN::pitchDetected);
connect(m_pitchFinder, &TpitchFinder::noteStarted, this, &TaudioIN::noteStartedSlot);
connect(m_pitchFinder, &TpitchFinder::noteFinished, this, &TaudioIN::noteFinishedSlot);
}
