C++ QAudioFormat::sampleSize方法代码示例

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
}

void qMultiplySamples(qreal factor, const QAudioFormat &format, const void* src, void* dest, int len)
{
    int samplesCount = len / (format.sampleSize()/8);

    switch ( format.sampleSize() ) {
    case 8:
        if (format.sampleType() == QAudioFormat::SignedInt)
            QAudioHelperInternal::adjustSamples<qint8>(factor,src,dest,samplesCount);
        else if (format.sampleType() == QAudioFormat::UnSignedInt)
            QAudioHelperInternal::adjustUnsignedSamples<quint8>(factor,src,dest,samplesCount);
        break;
    case 16:
        if (format.sampleType() == QAudioFormat::SignedInt)
            QAudioHelperInternal::adjustSamples<qint16>(factor,src,dest,samplesCount);
        else if (format.sampleType() == QAudioFormat::UnSignedInt)
            QAudioHelperInternal::adjustUnsignedSamples<quint16>(factor,src,dest,samplesCount);
        break;
    default:
        if (format.sampleType() == QAudioFormat::SignedInt)
            QAudioHelperInternal::adjustSamples<qint32>(factor,src,dest,samplesCount);
        else if (format.sampleType() == QAudioFormat::UnSignedInt)
            QAudioHelperInternal::adjustUnsignedSamples<quint32>(factor,src,dest,samplesCount);
        else if (format.sampleType() == QAudioFormat::Float)
            QAudioHelperInternal::adjustSamples<float>(factor,src,dest,samplesCount);
    }
}

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;
}

bool QAudioDeviceInfoInternal::testSettings(const QAudioFormat& format) const
{
    // Set nearest to closest settings that do work.
    // See if what is in settings will work (return value).

    bool failed = false;

    // For now, just accept only audio/pcm codec
    if(!format.codec().startsWith(QLatin1String("audio/pcm")))
        failed = true;

    if(!failed && !(format.channels() == 1 || format.channels() == 2))
        failed = true;

    if(!failed) {
        if(!(format.frequency() == 8000 || format.frequency() == 11025 || format.frequency() == 22050 ||
	   format.frequency() == 44100 || format.frequency() == 48000 || format.frequency() == 96000))
	    failed = true;
    }

    if(!failed && !(format.sampleSize() == 8 || format.sampleSize() == 16))
        failed = true;

    if(!failed) {
        // settings work
        return true;
    }
    return false;
}

pa_sample_spec audioFormatToSampleSpec(const QAudioFormat &format)
{
    pa_sample_spec  spec;

    spec.rate = format.frequency();
    spec.channels = format.channels();

    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 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();
}

// 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 xmppClient::slotConnected()
{
    QXmppCall *call = qobject_cast<QXmppCall*>(sender());
    Q_ASSERT(call);

    qDebug() << "Call connected";
    QXmppRtpChannel *channel = call->audioChannel();

    // prepare audio format
    QAudioFormat format;
    format.setFrequency(channel->payloadType().clockrate());
    format.setChannels(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.frequency() * format.channels() * (format.sampleSize() / 8) * 160) / 1000;

    // initialise audio output
    QAudioOutput *audioOutput = new QAudioOutput(format, this);
    audioOutput->setBufferSize(bufferSize);
    audioOutput->start(channel);

    // initialise audio input
    QAudioInput *audioInput = new QAudioInput(format, this);
    audioInput->setBufferSize(bufferSize);
    audioInput->start(channel);
}

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);
    }
}

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;
}

AudioStreamBasicDescription toAudioStreamBasicDescription(QAudioFormat const& audioFormat)
{
    AudioStreamBasicDescription sf;

    sf.mFormatFlags         = kAudioFormatFlagIsPacked;
    sf.mSampleRate          = audioFormat.frequency();
    sf.mFramesPerPacket     = 1;
    sf.mChannelsPerFrame    = audioFormat.channels();
    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;
}

bool isPCMS16LE(const QAudioFormat &format)
{
    return isPCM(format) &&
           format.sampleType() == QAudioFormat::SignedInt &&
           format.sampleSize() == 16 &&
           format.byteOrder() == QAudioFormat::LittleEndian;
}

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 SpectrumAnalyser::calculate(const QByteArray &buffer,
                                 const QAudioFormat &format, bool isSample,char phoneme)
{
    // 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.channels() / 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.frequency()),
                       Q_ARG(int, bytesPerSample),
                       Q_ARG(bool,isSample),
                       Q_ARG(char,phoneme));
        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
    }
}