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Set audio timeout to a low value again

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dec05eba
2024-04-13 01:46:31 +02:00
parent 1c274cd448
commit bc55369230
2 changed files with 42 additions and 9 deletions
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47
src/main.cpp
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@@ -2410,21 +2410,26 @@ int main(int argc, char **argv) {
swr_init(swr);
}
const int64_t no_input_sleep_ms = 500;
double received_audio_time = clock_get_monotonic_seconds();
const double timeout_sec = 1000.0 / (double)audio_track.codec_context->sample_rate;
const int64_t timeout_ms = std::round(timeout_sec * 1000.0);
while(running) {
void *sound_buffer;
int sound_buffer_size = -1;
if(audio_device.sound_device.handle)
sound_buffer_size = sound_device_read_next_chunk(&audio_device.sound_device, &sound_buffer, 0.5);
sound_buffer_size = sound_device_read_next_chunk(&audio_device.sound_device, &sound_buffer, timeout_ms * 1000.0);
const bool got_audio_data = sound_buffer_size >= 0;
const double this_audio_frame_time = clock_get_monotonic_seconds() - paused_time_offset;
if(paused) {
if(got_audio_data)
received_audio_time = this_audio_frame_time;
if(!audio_device.sound_device.handle)
usleep(no_input_sleep_ms * 1000);
usleep(timeout_ms * 1000);
continue;
}
@@ -2435,20 +2440,44 @@ int main(int argc, char **argv) {
break;
}
if(!got_audio_data) {
// TODO: Is this |received_audio_time| really correct?
const double prev_audio_time = received_audio_time;
const double audio_receive_time_diff = this_audio_frame_time - received_audio_time;
int64_t num_missing_frames = std::round(audio_receive_time_diff / timeout_sec);
if(got_audio_data)
num_missing_frames = std::max((int64_t)0, num_missing_frames - 1);
if(!audio_device.sound_device.handle)
num_missing_frames = std::max((int64_t)1, num_missing_frames);
if(got_audio_data)
received_audio_time = this_audio_frame_time;
// Fucking hell is there a better way to do this? I JUST WANT TO KEEP VIDEO AND AUDIO SYNCED HOLY FUCK I WANT TO KILL MYSELF NOW.
// THIS PIECE OF SHIT WANTS EMPTY FRAMES OTHERWISE VIDEO PLAYS TOO FAST TO KEEP UP WITH AUDIO OR THE AUDIO PLAYS TOO EARLY.
// BUT WE CANT USE DELAYS TO GIVE DUMMY DATA BECAUSE PULSEAUDIO MIGHT GIVE AUDIO A BIG DELAYED!!!
// This garbage is needed because we want to produce constant frame rate videos instead of variable frame rate
// videos because bad software such as video editing software and VLC do not support variable frame rate software,
// despite nvidia shadowplay and xbox game bar producing variable frame rate videos.
// So we have to make sure we produce frames at the same relative rate as the video.
if(num_missing_frames >= 5 || !audio_device.sound_device.handle) {
// TODO:
//audio_track.frame->data[0] = empty_audio;
received_audio_time = this_audio_frame_time;
if(needs_audio_conversion)
swr_convert(swr, &audio_device.frame->data[0], audio_track.codec_context->frame_size, (const uint8_t**)&empty_audio, audio_track.codec_context->frame_size);
else
audio_device.frame->data[0] = empty_audio;
const int64_t new_pts = (this_audio_frame_time - record_start_time) * AV_TIME_BASE;
if(new_pts != audio_device.frame->pts) {
// TODO: Check if duplicate frame can be saved just by writing it with a different pts instead of sending it again
std::lock_guard<std::mutex> lock(audio_filter_mutex);
for(int i = 0; i < num_missing_frames; ++i) {
const int64_t new_pts = ((prev_audio_time - record_start_time) + timeout_sec * i) * AV_TIME_BASE;
if(new_pts == audio_device.frame->pts)
continue;
audio_device.frame->pts = new_pts;
if(audio_track.graph) {
std::lock_guard<std::mutex> lock(audio_filter_mutex);
// TODO: av_buffersrc_add_frame
if(av_buffersrc_write_frame(audio_device.src_filter_ctx, audio_device.frame) < 0) {
fprintf(stderr, "Error: failed to add audio frame to filter\n");
@@ -2466,7 +2495,7 @@ int main(int argc, char **argv) {
}
if(!audio_device.sound_device.handle)
usleep(no_input_sleep_ms * 1000);
usleep(timeout_ms * 1000);
if(got_audio_data) {
// TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?