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C++ QAudioFormat::channelCount方法代码示例

本文整理汇总了C++中QAudioFormat::channelCount方法的典型用法代码示例。如果您正苦于以下问题：C++ QAudioFormat::channelCount方法的具体用法？C++ QAudioFormat::channelCount怎么用？C++ QAudioFormat::channelCount使用的例子？那么, 这里精选的方法代码示例或许可以为您提供帮助。您也可以进一步了解该方法所在类QAudioFormat的用法示例。


在下文中一共展示了QAudioFormat::channelCount方法的15个代码示例，这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞，您的评价将有助于系统推荐出更棒的C++代码示例。

示例1: audioLength

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;
}
开发者ID:MarianMMX，项目名称:MarianMMX，代码行数:7，代码来源:utils.cpp

示例2: generateData

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;
	}
}
开发者ID:Alexandr-Galko，项目名称:Joker，代码行数:25，代码来源:generator.cpp

示例3: writeHeader

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
}
开发者ID:kimmoli，项目名称:audiodecoder，代码行数:35，代码来源:wavefilewriter.cpp

示例4: audioLength

// 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;
}
开发者ID:AlexSeiden，项目名称:dLite，代码行数:10，代码来源:utils.cpp

示例5: generateData

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;
    }
}
开发者ID:Vasyaaa，项目名称:auFM，代码行数:53，代码来源:aafm.cpp

示例6: formatToString

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;
}
开发者ID:MarianMMX，项目名称:MarianMMX，代码行数:50，代码来源:utils.cpp

示例7: if

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;
}
开发者ID:EQ4，项目名称:Visore，代码行数:34，代码来源:viaudioformat.cpp

示例8: setAudioFormat

void WaveRenderArea::setAudioFormat(const QAudioFormat &format)
{
  Q_D(WaveRenderArea);
  Q_ASSERT(format.channelCount() == 1);
  d->audioFormat = format;
  d->maxAmplitude = AudioInputDevice::maxAmplitudeForFormat(d->audioFormat);
}
开发者ID:ola-ct，项目名称:Qliq，代码行数:7，代码来源:waverenderarea.cpp

示例9: processLocalAudio

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();
        }
    }
}
开发者ID:RyanDowne，项目名称:hifi，代码行数:26，代码来源:AudioReflector.cpp

示例10: toAudioStreamBasicDescription

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;
}
开发者ID:jusa，项目名称:qtmultimedia-1，代码行数:32，代码来源:qaudio_mac.cpp

示例11: audioFormatToSampleSpec

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;
}
开发者ID:MarianMMX，项目名称:MarianMMX，代码行数:34，代码来源:qpulsehelpers.cpp

示例12: slotAudioModeChanged

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);
    }
}
开发者ID:ninoles，项目名称:qxmpp，代码行数:33，代码来源:example_4_callHandling.cpp

示例13: audioFormatToSLFormatPCM

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;

}
开发者ID:MarianMMX，项目名称:MarianMMX，代码行数:17，代码来源:qopenslesengine.cpp

示例14: generateData

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);

}
开发者ID:zach-capalbo，项目名称:Kromophone，代码行数:17，代码来源:AudioGenerator.cpp

示例15: file

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();
}
开发者ID:Elleo，项目名称:qtmultimedia，代码行数:58，代码来源:tst_qwavedecoder.cpp


注：本文中的QAudioFormat::channelCount方法示例由纯净天空整理自Github/MSDocs等开源代码及文档管理平台，相关代码片段筛选自各路编程大神贡献的开源项目，源码版权归原作者所有，传播和使用请参考对应项目的License；未经允许，请勿转载。