本文整理汇总了C++中QAudioFormat::setCodec方法的典型用法代码示例。如果您正苦于以下问题:C++ QAudioFormat::setCodec方法的具体用法?C++ QAudioFormat::setCodec怎么用?C++ QAudioFormat::setCodec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::setCodec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: setQuality
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;
}
示例2: start
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);
}
示例3: QObject
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()));
}
示例4: startPlaying
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();
}
}
示例5: test_2
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~";
}
示例6: start
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);
}
示例7: QObject
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()));
}
示例8: slotConnected
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);
}
示例9: nearestFormat
QAudioFormat QAudioDeviceInfo::nearestFormat(const QAudioFormat &settings) const
{
if (isFormatSupported(settings))
return settings;
QAudioFormat nearest = settings;
nearest.setCodec(QLatin1String("audio/pcm"));
if (nearest.sampleType() == QAudioFormat::Unknown) {
QAudioFormat preferred = preferredFormat();
nearest.setSampleType(preferred.sampleType());
}
QMap<int,int> testFrequencies;
QList<int> frequenciesAvailable = supportedFrequencies();
QMap<int,int> testSampleSizes;
QList<int> sampleSizesAvailable = supportedSampleSizes();
// Get sorted sampleSizes (equal to and ascending values only)
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;
if (size >= settings.sampleSize()) {
int diff = larger - smaller;
testSampleSizes.insert(diff, size);
}
}
示例10: connect
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;
}
示例11: recordAudioSample
void AudioCore::recordAudioSample(int t = 500, bool writeToFile = false)
{
latestSampleDuration = t;
writeSampleToFile = writeToFile;
sampleAudioBuffer.open(QBuffer::ReadWrite);
QAudioFormat format;
// Настраиваем желаемый формат, например:
format.setSampleRate(AudioCore::SampleRate);
format.setSampleSize(8 * AudioCore::SampleSize);
format.setChannelCount(1);
format.setCodec("audio/pcm");
format.setByteOrder(QAudioFormat::LittleEndian);
format.setSampleType(QAudioFormat::UnSignedInt);
// ВЫбрать устройство ввода 0:
QAudioDeviceInfo info(QAudioDeviceInfo::availableDevices(QAudio::AudioInput).at(0));
qDebug() << "Selected input device =" << info.deviceName();
if (!info.isFormatSupported(format)) {
qWarning() << "Default format not supported, trying to use the nearest.";
format = info.nearestFormat(format);
}
audio = new QAudioInput(format);
// Очень важно, иначе будут шумы и все картинки ужасно некрасивые.
audio->setVolume(AudioCore::AudioLevel);
connect(audio, SIGNAL(stateChanged(QAudio::State)), this, SLOT(handleStateChanged(QAudio::State)));
QTimer::singleShot(t, this, SLOT(stopRecording()));
audio->start(&sampleAudioBuffer);
}
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);
}
}
示例13: InitializeAudio
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);
}
示例14: StartRecord
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);
}
}
示例15: startStreaming
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();
}
}