C++ QAudioFormat::setSampleSize方法代码示例

void S60AudioEncoderControl::setAudioSettings(const QAudioEncoderSettings &settings)
{
    QAudioFormat fmt = m_session->format();
    if (settings.encodingMode() == QtMultimediaKit::ConstantQualityEncoding) {
        fmt.setCodec(settings.codec());
        setQuality(settings.quality(), fmt);
        if (settings.sampleRate() > 0) {
            fmt.setFrequency(settings.sampleRate());
        }
        if (settings.channelCount() > 0)
            fmt.setChannels(settings.channelCount());
    }else {
        if (settings.sampleRate() == 8000) {
            fmt.setSampleType(QAudioFormat::UnSignedInt);
            fmt.setSampleSize(8);
        } else {
            fmt.setSampleType(QAudioFormat::SignedInt);
            fmt.setSampleSize(16);
        }
        fmt.setCodec(settings.codec());
        fmt.setFrequency(settings.sampleRate());
        fmt.setChannels(settings.channelCount());
    }
    m_session->setFormat(fmt);
    m_settings = settings;
}

void S60AudioEncoderControl::setQuality(QtMultimediaKit::EncodingQuality value, QAudioFormat &fmt)
{
    switch (value) {
    case QtMultimediaKit::VeryLowQuality:
    case QtMultimediaKit::LowQuality:
        // low, 8000Hz mono U8
        fmt.setSampleType(QAudioFormat::UnSignedInt);
        fmt.setSampleSize(8);
        fmt.setFrequency(8000);
        fmt.setChannels(1);
        break;
    case QtMultimediaKit::NormalQuality:
        // medium, 22050Hz mono S16
        fmt.setSampleType(QAudioFormat::SignedInt);
        fmt.setSampleSize(16);
        fmt.setFrequency(22050);
        fmt.setChannels(1);
        break;
    case QtMultimediaKit::HighQuality:
    case QtMultimediaKit::VeryHighQuality:
        // high, 44100Hz mono S16
        fmt.setSampleType(QAudioFormat::SignedInt);
        fmt.setSampleSize(16);
        fmt.setFrequency(44100);
        fmt.setChannels(2);
        break;
    default:
        break;
    }
}

QAudioFormat audioFormat( const int channels, const int sampleRate, AVSampleFormat sampleFormat )
{
    QAudioFormat format;
    format.setChannelCount( channels );
    format.setCodec( "audio/pcm" );
    switch (sampleFormat)
    {
    case AV_SAMPLE_FMT_U8:          ///< unsigned 8 bits
        format.setSampleSize(8);
        format.setSampleType( QAudioFormat::UnSignedInt  );
        break;
    case AV_SAMPLE_FMT_S16:         ///< signed 16 bits
        format.setSampleSize(16);
        format.setSampleType( QAudioFormat::SignedInt  );
        break;

    case AV_SAMPLE_FMT_S32:         ///< signed 32 bits
        format.setSampleSize(32);
        format.setSampleType( QAudioFormat::SignedInt  );
        break;
    case AV_SAMPLE_FMT_FLT:         ///< float
        format.setSampleSize(16);
        format.setSampleType( QAudioFormat::Float  );
        break;
    case AV_SAMPLE_FMT_DBL:         ///< double
        format.setSampleSize(32);
        format.setSampleType( QAudioFormat::Float  );
        break;

    case AV_SAMPLE_FMT_U8P:         ///< unsigned 8 bits: planar
        format.setSampleSize(8);
        format.setSampleType( QAudioFormat::UnSignedInt  );
        break;
    case AV_SAMPLE_FMT_S16P:        ///< signed 16 bits: planar
        format.setSampleSize(16);
        format.setSampleType( QAudioFormat::SignedInt  );
        break;
    case AV_SAMPLE_FMT_S32P:        ///< signed 32 bits: planar
        format.setSampleSize(32);
        format.setSampleType( QAudioFormat::SignedInt  );
        break;
    case AV_SAMPLE_FMT_FLTP:        ///< float: planar
        format.setSampleSize(16);
        format.setSampleType( QAudioFormat::SignedInt  );
        break;
    case AV_SAMPLE_FMT_DBLP:        ///< double, planar
        format.setSampleSize(32);
        format.setSampleType( QAudioFormat::SignedInt  );
        break;
    default:
        qWarning() << "codec format: " << sampleFormat << AV_SAMPLE_FMT_NONE;
        return QAudioFormat();
    }

    format.setSampleRate( sampleRate );
    return format;
}

void Airplay :: start()
{
	// read the airplay pipe
	file.setFileName("/tmp/shairport-sync-pipe");
	file.open(QIODevice::ReadOnly);

	// set the audio format
	// https://github.com/mikebrady/shairport-sync/issues/126
	// Playing raw data 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
	QAudioFormat format;
	format.setSampleRate(44100); // rate
	format.setChannelCount(2); // stereo
	format.setCodec("audio/pcm"); // raw
	format.setSampleSize(16); // 16 bit
	format.setByteOrder(QAudioFormat::LittleEndian);
	format.setSampleType(QAudioFormat::SignedInt);

	// check format is supported
	if (!deviceInfo.isFormatSupported(format))
		qWarning() << "Output format not supported";

	// connect signals
	audio = new QAudioOutput(deviceInfo, format, this);
	connect(audio, SIGNAL(stateChanged(QAudio::State)), this, SLOT(handleStateChange(QAudio::State)));

	// start playing
	audio->start(&file);
}

BeatController::BeatController(QAudioDeviceInfo inputDevice, uint16_t recordSize, uint32_t sampleRate, uint16_t m_bandCount, QObject *parent) : QObject(parent)
{
    m_RecordSize = recordSize;
    m_Buffer = new SoundBuffer(recordSize);
    m_Analyser = new BeatAnalyser(m_bandCount,sampleRate,recordSize);
    m_inputDevice = QAudioDeviceInfo(inputDevice);

    m_FFT = new FFT(recordSize);
    m_FFT->setSoundBuffer(m_Buffer);
    m_Analyser->setFFT(m_FFT);

    QAudioFormat audioFormat;
    audioFormat.setSampleRate(sampleRate);
    audioFormat.setChannelCount(1);
    audioFormat.setSampleSize(16);
    audioFormat.setSampleType(QAudioFormat::SignedInt);
    audioFormat.setByteOrder(QAudioFormat::LittleEndian);
    audioFormat.setCodec("audio/pcm");

    //QAudioDeviceInfo info(QAudioDeviceInfo::defaultInputDevice());

    if (!m_inputDevice.isFormatSupported(audioFormat)) {
        qWarning() << "Default format not supported - trying to use nearest";
        audioFormat = m_inputDevice.nearestFormat(audioFormat);
    }

    m_audioInput = new QAudioInput(m_inputDevice, audioFormat, this);
    m_ioDevice = m_audioInput->start();
    connect(m_ioDevice, SIGNAL(readyRead()), this, SLOT(readAudio()));

}

int CRecordData::StartRecord(bool bIsSection)
{
    if(true == bIsSection)
    {

    }
    else
    {
        SetNumofRecDatSec(0);

        // 初始化一个buffer储存raw数据
        QBuffer* bufRecord = new QBuffer();
        bufRecord->open( QIODevice::WriteOnly | QIODevice::Truncate );

        SetRecDatSec(m_iNumOfRecDatSec, bufRecord);
        SetNumofRecDatSec(m_iNumOfRecDatSec+1);

        // 设置录音格式
        QAudioFormat format;
        format.setSampleRate(8000);
        format.setChannelCount(1);
        format.setSampleSize(8);
        format.setCodec("audio/pcm");
        format.setByteOrder(QAudioFormat::LittleEndian);
        format.setSampleType(QAudioFormat::UnSignedInt);

        // 设置录音设备
        audioInput = new QAudioInput(format, this);
        audioInput->start(bufRecord);
    }

}

void AudioProcessorQt::start() {
	if (!input()) {
		return;
	}

	if (!m_device) {
		m_device = new AudioDevice(this);
	}

	if (!m_audioOutput) {
		QAudioFormat format;
		format.setSampleRate(44100);
		format.setChannelCount(2);
		format.setSampleSize(16);
		format.setCodec("audio/pcm");
		format.setByteOrder(QAudioFormat::LittleEndian);
		format.setSampleType(QAudioFormat::SignedInt);

		m_audioOutput = new QAudioOutput(format, this);
		m_audioOutput->setCategory("game");
	}

	m_device->setInput(input());
	m_device->setFormat(m_audioOutput->format());
	m_audioOutput->setBufferSize(input()->audioBuffers * 4);

	m_audioOutput->start(m_device);
}

void RtpAudioStream::startPlaying()
{
    if (audio_output_) return;

    QAudioFormat format;
    format.setSampleRate(audio_out_rate_);
    format.setSampleSize(sample_bytes_ * 8); // bits
    format.setSampleType(QAudioFormat::SignedInt);
    format.setChannelCount(1);
    format.setCodec("audio/pcm");

    // RTP_STREAM_DEBUG("playing %s %d samples @ %u Hz",
    //                 tempfile_->fileName().toUtf8().constData(),
    //                 (int) tempfile_->size(), audio_out_rate_);

    audio_output_ = new QAudioOutput(format, this);
    audio_output_->setNotifyInterval(65); // ~15 fps
    connect(audio_output_, SIGNAL(stateChanged(QAudio::State)), this, SLOT(outputStateChanged()));
    connect(audio_output_, SIGNAL(notify()), this, SLOT(outputNotify()));
    tempfile_->seek(0);
    audio_output_->start(tempfile_);
    emit startedPlaying();
    // QTBUG-6548 StoppedState is not always emitted on error, force a cleanup
    // in case playback fails immediately.
    if (audio_output_ && audio_output_->state() == QAudio::StoppedState) {
        outputStateChanged();
    }
}

void InitializeAudio()
{
    m_format.setSampleRate(SampleRate); //set frequency to 44100
    m_format.setChannelCount(1); //set channels to mono
    m_format.setSampleSize(16); //set sample sze to 16 bit
    m_format.setSampleType(QAudioFormat::SignedInt ); //Sample type as usigned integer sample UnSignedInt
    m_format.setByteOrder(QAudioFormat::LittleEndian); //Byte order
    m_format.setCodec("audio/pcm"); //set codec as simple audio/pcm

    QAudioDeviceInfo infoIn(QAudioDeviceInfo::defaultInputDevice());
    if (!infoIn.isFormatSupported(m_format))
    {
        //Default format not supported - trying to use nearest
        m_format = infoIn.nearestFormat(m_format);
    }

    QAudioDeviceInfo infoOut(QAudioDeviceInfo::defaultOutputDevice());

    if (!infoOut.isFormatSupported(m_format))
    {
       //Default format not supported - trying to use nearest
        m_format = infoOut.nearestFormat(m_format);
    }

    m_audioInput = new QAudioInput(m_Inputdevice, m_format);
    m_audioOutput = new QAudioOutput(m_Outputdevice, m_format);
}

void QSpotifyAudioThreadWorker::startStreaming(int channels, int sampleRate)
{
    qDebug() << "QSpotifyAudioThreadWorker::startStreaming";
    if (!m_audioOutput) {
        QAudioFormat af;
        af.setChannelCount(channels);
        af.setCodec("audio/pcm");
        af.setSampleRate(sampleRate);
        af.setSampleSize(16);
        af.setSampleType(QAudioFormat::SignedInt);

        QAudioDeviceInfo info(QAudioDeviceInfo::defaultOutputDevice());
        if (!info.isFormatSupported(af)) {
            QList<QAudioDeviceInfo> devices = QAudioDeviceInfo::availableDevices(QAudio::AudioOutput);
            for (int i = 0; i < devices.size(); i++) {
                QAudioDeviceInfo dev = devices[i];
                qWarning() << dev.deviceName();
            }
            QCoreApplication::postEvent(QSpotifySession::instance(), new QEvent(QEvent::Type(StopEventType)));
            return;
        }

        m_audioOutput = new QAudioOutput(af);
        connect(m_audioOutput, SIGNAL(stateChanged(QAudio::State)), QSpotifySession::instance(), SLOT(audioStateChange(QAudio::State)));
        m_audioOutput->setBufferSize(BUF_SIZE);

        startAudioOutput();
    }
}

void MainWindow::on_pushButton_clicked()
{
      QIODevice *QID;
      //QID->open( QIODevice::WriteOnly);
      QBuffer myQB;

     //QID(myQB);
    //cb(128000,64000);
     //dFile.setFileName("../RecordTest.raw");
     microphoneBuffer->open( QIODevice::ReadWrite);
     QAudioFormat format;
     // Set up the desired format, for example:
     format.setSampleRate(16000);
     format.setChannelCount(1);
     format.setSampleSize(16);
     format.setCodec("audio/pcm");
     format.setByteOrder(QAudioFormat::LittleEndian);
     format.setSampleType(QAudioFormat::UnSignedInt);

     QAudioDeviceInfo info = QAudioDeviceInfo::defaultInputDevice();
     if (!info.isFormatSupported(format))
     {
         qWarning() << "Default format not supported, trying to use the nearest.";
         format = info.nearestFormat(format);
     }

     audio = new QAudioInput(format, this);
     connect(audio, SIGNAL(stateChanged(QAudio::State)), this, SLOT(handleStateChanged(QAudio::State)));

     //QTimer::singleShot(5000, this, SLOT(on_pushButton_2_clicked()));
     isRecording = true;
     audio->start(microphoneBuffer);
}

FrequencyAnalyzer::FrequencyAnalyzer(QObject *parent) :
    QObject(parent),
    d_ptr(new FrequencyAnalyzerPrivate(this))
{
    Q_D(FrequencyAnalyzer);

    QAudioDeviceInfo info = QAudioDeviceInfo::defaultInputDevice();

    qDebug() << "device name: " << info.deviceName() << "\n"
             << "supported frequency:" << info.supportedFrequencies() << "\n"
             << "supported codecs" << info.supportedCodecs() << "\n"
             << "supported sample sizes" << info.supportedSampleSizes() << "\n"
             << "supported sample types" << info.supportedSampleTypes() << "\n";

    QAudioFormat format = info.preferredFormat();
    format.setCodec("audio/pcm");
    format.setByteOrder(QAudioFormat::LittleEndian);
    format.setSampleType(QAudioFormat::SignedInt);
    format.setSampleSize(32);
    //format.setFrequency(d->sampling = 11025);
    //format.setFrequency(d->sampling = 22050);
    format.setFrequency(d->sampling = info.supportedFrequencies().last());
    format.setChannelCount(1);

    if (!info.isFormatSupported(format)) {
        qWarning("Format is unsupported");
        return;
    }

    d->input = new QAudioInput(info, format, this);
    connect(d->input, SIGNAL(stateChanged(QAudio::State)), SLOT(_q_onStateChanged()));
}

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);
}

int test_2()
{
    qyvlik::FFmpegStream* ffmpegStream = new qyvlik::FFmpegStream();

    QAudioFormat format;
    // Set up the format, eg.

    format.setSampleRate(44100);
    format.setChannelCount(2);
    format.setCodec("audio/pcm");
    format.setSampleType(QAudioFormat::SignedInt);
    format.setSampleSize(16);
    format.setByteOrder(QAudioFormat::LittleEndian);

    QAudioDeviceInfo info(QAudioDeviceInfo::defaultOutputDevice());
    if (!info.isFormatSupported(format)) {
        qDebug() << "Raw audio format not supported by backend, cannot play audio.";
        return -1;
    }

    ffmpegStream->setFileName("E:/Test/1.mp3");

    QAudioOutput *audio;
    audio = new QAudioOutput(format);
    audio->start(ffmpegStream);

    qDebug() << "Play Finished~";

}

AudioDecoder::AudioDecoder(bool isPlayback, bool isDelete)
    : m_cout(stdout, QIODevice::WriteOnly)
{
    m_isPlayback = isPlayback;
    m_isDelete = isDelete;

    // Make sure the data we receive is in correct PCM format.
    // Our wav file writer only supports SignedInt sample type.
    QAudioFormat format;
    format.setChannelCount(2);
    format.setSampleSize(16);
    format.setSampleRate(48000);
    format.setCodec("audio/pcm");
    format.setSampleType(QAudioFormat::SignedInt);
    m_decoder.setAudioFormat(format);

    connect(&m_decoder, SIGNAL(bufferReady()), this, SLOT(bufferReady()));
    connect(&m_decoder, SIGNAL(error(QAudioDecoder::Error)), this, SLOT(error(QAudioDecoder::Error)));
    connect(&m_decoder, SIGNAL(stateChanged(QAudioDecoder::State)), this, SLOT(stateChanged(QAudioDecoder::State)));
    connect(&m_decoder, SIGNAL(finished()), this, SLOT(finished()));
    connect(&m_decoder, SIGNAL(positionChanged(qint64)), this, SLOT(updateProgress()));
    connect(&m_decoder, SIGNAL(durationChanged(qint64)), this, SLOT(updateProgress()));

    connect(&m_soundEffect, SIGNAL(statusChanged()), this, SLOT(playbackStatusChanged()));
    connect(&m_soundEffect, SIGNAL(playingChanged()), this, SLOT(playingChanged()));

    m_progress = -1.0;
}