void AudioPeakMeterScopeWidget::refreshScope(const QSize& /*size*/, bool /*full*/)
{
SharedFrame sFrame;
while (m_queue.count() > 0) {
sFrame = m_queue.pop();
if (sFrame.is_valid() && sFrame.get_audio_samples() > 0) {
mlt_audio_format format = mlt_audio_s16;
int channels = sFrame.get_audio_channels();
int frequency = sFrame.get_audio_frequency();
int samples = sFrame.get_audio_samples();
Mlt::Frame mFrame = sFrame.clone(true, false, false);
m_filter->process(mFrame);
mFrame.get_audio( format, frequency, channels, samples );
QVector<double> levels;
for (int i = 0; i < channels; i++) {
QString s = QString("meta.media.audio_level.%1").arg(i);
double audioLevel = mFrame.get_double(s.toLatin1().constData());
if (audioLevel == 0.0) {
levels << -100.0;
} else {
levels << 20 * log10(audioLevel);
}
}
QMetaObject::invokeMethod(m_audioMeter, "showAudio", Qt::QueuedConnection, Q_ARG(const QVector<double>&, levels));
}
}
void MonitorAudioLevel::refreshScope(const QSize& /*size*/, bool /*full*/)
{
SharedFrame sFrame;
while (m_queue.count() > 0) {
sFrame = m_queue.pop();
if (sFrame.is_valid() && sFrame.get_audio_samples() > 0) {
mlt_audio_format format = mlt_audio_s16;
int channels = sFrame.get_audio_channels();
int frequency = sFrame.get_audio_frequency();
int samples = sFrame.get_audio_samples();
Mlt::Frame mFrame = sFrame.clone(true, false, false);
m_filter->process(mFrame);
mFrame.get_audio( format, frequency, channels, samples );
if (samples == 0) {
// There was an error processing audio from frame
continue;
}
QVector<int> levels;
for (int i = 0; i < audioChannels; i++) {
QString s = QString("meta.media.audio_level.%1").arg(i);
double audioLevel = mFrame.get_double(s.toLatin1().constData());
if (audioLevel == 0.0) {
levels << -100;
} else {
levels << (int) levelToDB(audioLevel);
}
}
QMetaObject::invokeMethod(this, "setAudioValues", Qt::QueuedConnection, Q_ARG(const QVector<int>&, levels));
}
}
void AudioGraphSpectrum::processSpectrum(const SharedFrame&frame)
{
if (!isVisible()) return;
mlt_audio_format format = mlt_audio_s16;
int channels = frame.get_audio_channels();
int frequency = frame.get_audio_frequency();
int samples = frame.get_audio_samples();
Mlt::Frame mFrame = frame.clone(true, false, false);
m_filter->process(mFrame);
mFrame.get_audio( format, frequency, channels, samples );
QVector<double> bands(AUDIBLE_BAND_COUNT);
float* bins = (float*)m_filter->get_data("bins");
int bin_count = m_filter->get_int("bin_count");
double bin_width = m_filter->get_double("bin_width");
int band = 0;
bool firstBandFound = false;
for (int bin = 0; bin < bin_count; bin++) {
// Loop through all the FFT bins and align bin frequencies with
// band frequencies.
double F = bin_width * (double)bin;
if (!firstBandFound) {
// Skip bins that come before the first band.
if (BAND_TAB[band + FIRST_AUDIBLE_BAND_INDEX].low > F) {
continue;
} else {
firstBandFound = true;
bands[band] = bins[bin];
}
} else if (BAND_TAB[band + FIRST_AUDIBLE_BAND_INDEX].high < F) {
// This bin is outside of this band - move to the next band.
band++;
if ((band + FIRST_AUDIBLE_BAND_INDEX) > LAST_AUDIBLE_BAND_INDEX) {
// Skip bins that come after the last band.
break;
}
bands[band] = bins[bin];
} else if (bands[band] < bins[bin] ) {
// Pick the highest bin level within this band to represent the
// whole band.
bands[band] = bins[bin];
}
}
// At this point, bands contains the magnitude of the signal for each
// band. Convert to dB.
for (band = 0; band < bands.size(); band++) {
double mag = bands[band];
double dB = mag > 0.0 ? 20 * log10( mag ) : -1000.0;
bands[band] = dB;
}
// Update the audio signal widget
QMetaObject::invokeMethod(m_graphWidget, "showAudio", Qt::QueuedConnection, Q_ARG(const QVector<double>&, bands));
}
void AudioEnvelope::loadEnvelope()
{
Q_ASSERT(m_envelope == NULL);
std::cout << "Loading envelope ..." << std::endl;
int samplingRate = m_info->info(0)->samplingRate();
mlt_audio_format format_s16 = mlt_audio_s16;
int channels = 1;
Mlt::Frame *frame;
int64_t position;
int samples;
m_envelope = new int64_t[m_envelopeSize];
m_envelopeMax = 0;
m_envelopeMean = 0;
QTime t;
t.start();
int count = 0;
m_producer->seek(m_offset);
m_producer->set_speed(1.0); // This is necessary, otherwise we don't get any new frames in the 2nd run.
for (int i = 0; i < m_envelopeSize; i++) {
frame = m_producer->get_frame(i);
position = mlt_frame_get_position(frame->get_frame());
samples = mlt_sample_calculator(m_producer->get_fps(), samplingRate, position);
int16_t *data = static_cast<int16_t*>(frame->get_audio(format_s16, samplingRate, channels, samples));
int64_t sum = 0;
for (int k = 0; k < samples; k++) {
sum += fabs(data[k]);
}
m_envelope[i] = sum;
m_envelopeMean += sum;
if (sum > m_envelopeMax) {
m_envelopeMax = sum;
}
// std::cout << position << "|" << m_producer->get_playtime()
// << "-" << m_producer->get_in() << "+" << m_producer->get_out() << " ";
delete frame;
count++;
if (m_length > 0 && count > m_length) {
break;
}
}
m_envelopeMean /= m_envelopeSize;
std::cout << "Calculating the envelope (" << m_envelopeSize << " frames) took "
<< t.elapsed() << " ms." << std::endl;
}