PapaTutuWawa/gpu-screen-recorder
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Is this the final solution to the audio crackling problem? increase pts by number of samples and add dummy audio frames between

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dec05eba
2022-09-20 20:11:56 +02:00
parent a668cac2bb
commit f810161203
2 changed files with 53 additions and 78 deletions
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50
src/main.cpp
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@@ -483,7 +483,7 @@ static AVCodecContext* create_audio_codec_context(AVFormatContext *av_format_con
#endif
codec_context->time_base.num = 1;
codec_context->time_base.den = AV_TIME_BASE;
codec_context->time_base.den = codec_context->sample_rate;
codec_context->framerate.num = fps;
codec_context->framerate.den = 1;
@@ -1462,11 +1462,12 @@ int main(int argc, char **argv) {
av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
swr_init(swr);
int64_t pts = 0;
const double target_audio_hz = 1.0 / (double)audio_track.codec_context->sample_rate;
while(running) {
void *sound_buffer;
int sound_buffer_size = sound_device_read_next_chunk(&audio_track.sound_device, &sound_buffer);
if(sound_buffer_size < 0)
sound_buffer = empty_audio;
int ret = av_frame_make_writable(audio_track.frame);
if (ret < 0) {
@@ -1474,15 +1475,42 @@ int main(int argc, char **argv) {
break;
}
// TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?
swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&sound_buffer, audio_track.sound_device.frames);
audio_track.frame->pts = (clock_get_monotonic_seconds() - start_time_pts) * AV_TIME_BASE;
const double this_audio_frame_time = clock_get_monotonic_seconds();
const int64_t expected_frames = std::round((this_audio_frame_time - start_time_pts) / target_audio_hz);
const int64_t num_missing_frames = std::max(0L, (expected_frames - pts) / audio_track.frame->nb_samples);
// Jesus 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!!!
if(num_missing_frames >= 5) {
// TODO:
//audio_track.frame->data[0] = empty_audio;
swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&empty_audio, audio_track.sound_device.frames);
// TODO: Check if duplicate frame can be saved just by writing it with a different pts instead of sending it again
for(int i = 0; i < num_missing_frames; ++i) {
audio_track.frame->pts = pts;
pts += audio_track.frame->nb_samples;
ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
if(ret >= 0){
receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_track.frame, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, *write_output_mutex);
} else {
fprintf(stderr, "Failed to encode audio!\n");
}
}
}
ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
if(ret >= 0){
receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_track.frame, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, *write_output_mutex);
} else {
fprintf(stderr, "Failed to encode audio!\n");
if(sound_buffer_size >= 0) {
// TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?
swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&sound_buffer, audio_track.sound_device.frames);
audio_track.frame->pts = pts;
pts += audio_track.frame->nb_samples;
ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
if(ret >= 0){
receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_track.frame, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, *write_output_mutex);
} else {
fprintf(stderr, "Failed to encode audio!\n");
}
}
}