Demo::Demo(QWidget *parent) :
QMainWindow(parent),
ui(new Ui::Demo)
{
ui->setupUi(this);
//create pointers
QJFastFIRFilter *fastfir;
QJSlowFIRFilter *slowfir;
//create fast fir LPF
fastfir = new QJFastFIRFilter(this);
fastfir->setKernel(QJFilterDesign::LowPassHanning(800,48000,1001));
//create slow fir LPF
slowfir = new QJSlowFIRFilter(this);
slowfir->setKernel(QJFilterDesign::LowPassHanning(800,48000,1001));
//make some data
QVector<kffsamp_t> buf;
buf.resize(48000*10);
//time slow fir
timer.start();
slowfir->Update(buf);
nMilliseconds = timer.elapsed();
ui->plainTextEdit->appendPlainText(((QString)"Slow FIR took %1 ms for %2 double samples with a kernel of 1001 samples").arg(nMilliseconds).arg(buf.size()));
//time fast fir
timer.start();
fastfir->Update(buf);
nMilliseconds = timer.elapsed();
ui->plainTextEdit->appendPlainText(((QString)"Fast FIR took %1 ms for %2 double samples with a kernel of 1001 samples").arg(nMilliseconds).arg(buf.size()));
//use the fast fir to LPF some random samples and output them to the sound card
ui->plainTextEdit->appendPlainText("Outputting 800Hz low pass filtered random samples to the soundcard using the fast fir");
QAudioFormat format;
format.setChannelCount(1);
format.setCodec("audio/pcm");
format.setSampleRate(48000);
format.setSampleSize(16);
format.setByteOrder(QAudioFormat::LittleEndian);
format.setSampleType(QAudioFormat::SignedInt);
generator = new Generator(format, this);
QAudioOutput *audiooutput = new QAudioOutput(format, this);
generator->start();
audiooutput->start(generator);
}
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~";
}
void WavSound::play() {
QFile *file = new QFile(fileName);
file->open(QIODevice::ReadOnly);
QAudioFormat f;
readHeader(&f, file);
QAudioFormat format = f;
QAudioDeviceInfo info(QAudioDeviceInfo::defaultOutputDevice());
if (!info.isFormatSupported(format)) {
qWarning() << "Raw audio format not supported by backend, cannot play audio.";
return;
}
//seek(0x4e*4); //...avoiding clicks from the wav files
file->seek(0x4e);
QAudioOutput *audio = new QAudioOutput(format, NULL);
if(file->size()<30000)
audio->setBufferSize(10000);
audio->setVolume(0.5);
connect(audio, SIGNAL(stateChanged(QAudio::State)), this, SLOT(handleStateChanged(QAudio::State)));
audio->start(file);
}
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);
}
}
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);
}
void push(double *left, double *right, int length, int offset) {
if (!device) {
device = out->start();
}
int remain = length;
int finished = 0;
while (0 < remain) {
int amount = kUnitBufferLength / 4 <= remain ? kUnitBufferLength / 4 : remain;
int index = 0;
for (int i = 0; i < amount; i++) {
int16_t leftData = (int16_t)(left[i + finished + offset] * 32767);
int16_t rightData = (int16_t)(right[i + finished + offset] * 32767);
buffer[index++] = static_cast<uint8_t *>(&leftData)[0];
buffer[index++] = static_cast<uint8_t *>(&leftData)[1];
buffer[index++] = static_cast<uint8_t *>(&rightData)[0];
buffer[index++] = static_cast<uint8_t *>(&rightData)[1];
}
int bytesToWrite = amount * 4;
int bytesWritten = 0;
while (bytesWritten < bytesToWrite) {
int actualWrittenBytes = device->write(
(const char *)(buffer + bytesWritten),
(qint64)(bytesToWrite - bytesWritten));
bytesWritten += actualWrittenBytes;
}
remain -= amount;
finished += amount;
}
}
int test_1(QObject* a)
{
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;
}
char filename[1024];
printf("input file name :");
scanf("%s",filename);
fflush(stdin);
fflush(stdout);
printf("start audio \n");
QAudioOutput *audio;
audio = new QAudioOutput(format, a);
QIODevice *out = audio->start();
#if 0
qDebug()<< out->isSequential() << " swquential"; // 不是顺序设备，是随机设备
#endif
decodeAndPlay(filename,out);
}
void NotificationSound::playSound()
{
//QSound::play(":/sound/notification_sound.wav");
//return;
qreal frequency = ((sFrequency <= 0)?2000:sFrequency);
qreal mseconds = ((sDuration <= 0)?1000:sDuration);
quint8 volume = ((sVolume <= 0)?1:sVolume);
qreal sampleRate = 2.0 * M_PI / (192000./frequency);
QByteArray bytebuf;
bytebuf.resize(mseconds * 192);
for (int i=0; i < bytebuf.size(); i++) {
bytebuf[i] = (quint8)(qreal(255) * qSin(qreal(i) * sampleRate));
}
QDataStream stream(&bytebuf, QIODevice::ReadWrite);
QAudioFormat format;
format.setSampleRate(192000);
format.setChannelCount(1);
format.setSampleSize(8);
format.setCodec("audio/pcm");
format.setByteOrder(QAudioFormat::LittleEndian);
format.setSampleType(QAudioFormat::UnSignedInt);
QAudioOutput * audio = new QAudioOutput(format, this);
audio->setVolume(1.0 * (qreal(volume + 1)/4));
audio->setBufferSize(bytebuf.size());
audio->start(stream.device());
QEventLoop loop;
QTimer::singleShot(mseconds*2, &loop, SLOT(quit()));
loop.exec();
}
bool TSoundOutputDeviceImp::doStopDevice() {
m_isPlaying = false;
m_audioOutput->stop();
return true;
}
