public/gpu-screen-recorder
1
0
Fork 0
mirror of https://repo.dec05eba.com/gpu-screen-recorder synced 2026-03-31 17:17:16 +09:00
Code Issues Actions Packages Projects Releases Wiki Activity

More audio bullshit, repeat write audio frames for audio delay

Browse Source
This commit is contained in:
dec05eba
2024-04-10 00:11:09 +02:00
parent e7aa4a5499
commit 356714a6eb
1 changed files with 34 additions and 2 deletions
Download Patch File Download Diff File Expand all files Collapse all files

36
src/main.cpp
View File

@@ -1713,6 +1713,12 @@ int main(int argc, char **argv) {
usage();
}
if(audio_codec == AudioCodec::FLAC) {
fprintf(stderr, "Warning: flac audio codec has been temporary disabled, using opus audio codec instead\n");
audio_codec_to_use = "opus";
audio_codec = AudioCodec::OPUS;
}
bool overclock = false;
const char *overclock_str = args["-oc"].value();
if(!overclock_str)
@@ -2388,9 +2394,35 @@ int main(int argc, char **argv) {
double received_audio_time = clock_get_monotonic_seconds();
const int64_t timeout_ms = std::round((1000.0 / (double)audio_track.codec_context->sample_rate) * 1000.0);
// Move audio back by around 252 ms. This is just a shitty way to handle audio latency but pulseaudio latency calculation
// Move audio forward by around 252 ms (for opus/aac), or 42ms for flac. This is just a shitty way to handle audio latency but pulseaudio latency calculation
// returns much lower value which isn't helpful.
audio_device.frame->pts = audio_track.codec_context->frame_size * 12;
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;
int num_frames_to_delay = 12;
if(audio_codec == AudioCodec::FLAC)
num_frames_to_delay = 2;
for(int i = 0; i < num_frames_to_delay; ++i) {
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");
}
} else {
int ret = avcodec_send_frame(audio_track.codec_context, audio_device.frame);
if(ret >= 0) {
// TODO: Move to separate thread because this could write to network (for example when livestreaming)
receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_device.frame->pts, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, write_output_mutex, paused_time_offset);
} else {
fprintf(stderr, "Failed to encode audio!\n");
}
}
audio_device.frame->pts += audio_track.codec_context->frame_size;
}
while(running) {
void *sound_buffer;