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fixed audio monitor regression

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This commit is contained in:
itsmattkc
2019-01-22 12:57:00 +11:00
parent efc868d217
commit e7f05ff64b
5 changed files with 77 additions and 113 deletions
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playback/audio.cpp
+28 -34
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@@ -164,47 +164,36 @@ void AudioSenderThread::run() {
lock.unlock();
}
int AudioSenderThread::send_audio_to_output(int offset, int max) {
int AudioSenderThread::send_audio_to_output(unsigned long offset, int max) {
// send audio to device
unsigned long actual_write = audio_io_device->write((const char*) audio_ibuffer+offset, max);
unsigned long audio_ibuffer_limit = audio_ibuffer_read + actual_write;
// send samples to audio monitor cache
// TODO make this work for the footage viewer - currently, enabling it causes crash due to an ASSERT
Sequence* s = nullptr;
/*if (panel_footage_viewer->playing) {
s = panel_footage_viewer->seq;
}*/
if (panel_sequence_viewer->playing) {
s = panel_sequence_viewer->seq;
}
if (s != nullptr) {
if (panel_timeline->audio_monitor->sample_cache_offset == -1) {
panel_timeline->audio_monitor->sample_cache_offset = s->playhead;
}
int channel_count = av_get_channel_layout_nb_channels(s->audio_layout);
long sample_cache_playhead = panel_timeline->audio_monitor->sample_cache_offset + (panel_timeline->audio_monitor->sample_cache.size()/channel_count);
unsigned long next_buffer_offset, buffer_offset_adjusted;
int i;
unsigned long buffer_offset = get_buffer_offset_from_frame(s->frame_rate, sample_cache_playhead);
if (samples.size() != channel_count) samples.resize(channel_count);
samples.fill(0);
if (actual_write > 0) {
// average values and send to audio monitor
int channels = audio_output->format().channelCount();
unsigned long lim = offset + actual_write;
QVector<double> averages;
averages.resize(channels);
averages.fill(0);
// TODO: I don't like this, but i'm not sure if there's a smarter way to do it
while (static_cast<unsigned long>(buffer_offset) < audio_ibuffer_limit) {
sample_cache_playhead++;
next_buffer_offset = qMin(get_buffer_offset_from_frame(s->frame_rate, sample_cache_playhead), static_cast<unsigned long>(audio_ibuffer_limit));
while (buffer_offset < next_buffer_offset) {
for (i=0;i<samples.size();i++) {
buffer_offset_adjusted = buffer_offset%audio_ibuffer_size;
samples[i] = qMax(qAbs(qint16(((audio_ibuffer[buffer_offset_adjusted+1] & 0xFF) << 8) | (audio_ibuffer[buffer_offset_adjusted] & 0xFF))), samples[i]);
buffer_offset += 2;
}
}
panel_timeline->audio_monitor->sample_cache.append(samples);
buffer_offset = next_buffer_offset;
for (int i=0;i<channels;i++) {
averages[i] = 0;
}
int counter = 0;
qint16 sample;
for (unsigned long i=offset;i<lim;i+=2) {
sample = qint16(((audio_ibuffer[i+1] & 0xFF) << 8) | (audio_ibuffer[i] & 0xFF));
averages[counter%channels] += qAbs(double(sample)/32768.0);
counter++;
}
double divider = counter / channels;
for (int i=0;i<channels;i++) {
averages[i] = log_volume(1.0-(averages[i]/divider));
}
panel_timeline->audio_monitor->set_value(averages);
}
memset(audio_ibuffer+offset, 0, actual_write);
@@ -214,6 +203,11 @@ int AudioSenderThread::send_audio_to_output(int offset, int max) {
return actual_write;
}
double log_volume(double linear) {
// expects a value between 0 and 1 (or more if amplifying)
return (qExp(linear)-1)/(M_E-1);
}
void int32_to_char_array(qint32 i, char* array) {
memcpy(array, &i, 4);
}