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even more improved corner pin and more consistent audio

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This commit is contained in:
itsmattkc
2018-11-12 20:49:15 +11:00
parent 18c112d83e
commit 4f3c807c6a
5 changed files with 45 additions and 31 deletions
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playback/audio.cpp
+5 -7
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@@ -68,9 +68,7 @@ void init_audio(Sequence* s) {
audio_thread->start(QThread::TimeCriticalPriority);
clear_audio_ibuffer();
}
dout << "h";
}
}
void stop_audio() {
@@ -88,9 +86,9 @@ void clear_audio_ibuffer() {
audio_ibuffer_read = 0;
}
int get_buffer_offset_from_frame(Sequence* s, long frame) {
int get_buffer_offset_from_frame(double framerate, long frame) {
if (frame >= audio_ibuffer_frame) {
return qFloor(((double) (frame - audio_ibuffer_frame)/s->frame_rate)*s->audio_frequency)*av_get_bytes_per_sample(AV_SAMPLE_FMT_S16)*av_get_channel_layout_nb_channels(s->audio_layout);
return qFloor(((double) (frame - audio_ibuffer_frame)/framerate)*audio_output->format().sampleRate())*av_get_bytes_per_sample(AV_SAMPLE_FMT_S16)*av_get_channel_layout_nb_channels(AV_CH_LAYOUT_STEREO);
} else {
dout << "[WARNING] Invalid values passed to get_buffer_offset_from_frame";
return 0;
@@ -160,14 +158,14 @@ int AudioSenderThread::send_audio_to_output(int offset, int max) {
int channel_count = av_get_channel_layout_nb_channels(s->audio_layout);
long sample_cache_playhead = panel_timeline->ui->audio_monitor->sample_cache_offset + (panel_timeline->ui->audio_monitor->sample_cache.size()/channel_count);
int next_buffer_offset, buffer_offset_adjusted, i;
int buffer_offset = get_buffer_offset_from_frame(s, sample_cache_playhead);
int 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);
// TODO: I don't like this, but i'm not sure if there's a smarter way to do it
while (buffer_offset < audio_ibuffer_limit) {
sample_cache_playhead++;
next_buffer_offset = qMin(get_buffer_offset_from_frame(s, sample_cache_playhead), audio_ibuffer_limit);
next_buffer_offset = qMin(get_buffer_offset_from_frame(s->frame_rate, sample_cache_playhead), audio_ibuffer_limit);
while (buffer_offset < next_buffer_offset) {
for (i=0;i<samples.size();i++) {
buffer_offset_adjusted = buffer_offset%audio_ibuffer_size;