diff options
| -rw-r--r-- | TODO | 4 | ||||
| -rw-r--r-- | src/main.cpp | 47 |
2 files changed, 42 insertions, 9 deletions
@@ -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.
\ No newline at end of file diff --git a/src/main.cpp b/src/main.cpp index 9fb8ba7..a500304 100644 --- a/src/main.cpp +++ b/src/main.cpp @@ -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? |