public/gpu-screen-recorder
1
0
Fork 0
mirror of https://repo.dec05eba.com/gpu-screen-recorder synced 2026-03-31 09:07:13 +09:00
Code Issues Actions Packages Projects Releases Wiki Activity

Set audio timeout to a low value again

Browse Source
This commit is contained in:
dec05eba
2024-04-13 01:46:31 +02:00
parent 1c274cd448
commit bc55369230
2 changed files with 42 additions and 9 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
TODO
View File

@@ -117,3 +117,7 @@ Support drm plane rotation. Neither X11 nor any Wayland compositor currently rot
Investigate if there is a way to do gpu->gpu copy directly without touching system ram to enable video encoding on a different gpu. On nvidia this is possible with cudaMemcpyPeer, but how about from an intel/amd gpu to an nvidia gpu or the other way around or any combination of iGPU and dedicated GPU?
Maybe something with clEnqueueMigrateMemObjects? on AMD something with DirectGMA maybe?
Fix opus/flac ( variable framerate audio :( ).
Change aac bitrate by default to 160kbps and opus/flac to 128kbps and remove the -ab option.

49
src/main.cpp
View File

@@ -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) {
audio_device.frame->pts = new_pts;
if(audio_track.graph) {
// 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) {
// 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?