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Time based audio latency, test, might fix some shits

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dec05eba
2024-04-11 14:40:27 +02:00
parent f8322c3c28
commit 52688dad72
3 changed files with 85 additions and 163 deletions
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155
src/main.cpp
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@@ -315,7 +315,7 @@ static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_code
#endif
codec_context->time_base.num = 1;
codec_context->time_base.den = codec_context->sample_rate;
codec_context->time_base.den = AV_TIME_BASE;
codec_context->framerate.num = fps;
codec_context->framerate.den = 1;
codec_context->thread_count = 1;
@@ -1699,10 +1699,10 @@ int main(int argc, char **argv) {
usage();
}
AudioCodec audio_codec = AudioCodec::OPUS;
AudioCodec audio_codec = AudioCodec::AAC;
const char *audio_codec_to_use = args["-ac"].value();
if(!audio_codec_to_use)
audio_codec_to_use = "opus";
audio_codec_to_use = "aac";
if(strcmp(audio_codec_to_use, "aac") == 0) {
audio_codec = AudioCodec::AAC;
@@ -1715,10 +1715,10 @@ 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;
if(audio_codec == AudioCodec::OPUS || audio_codec == AudioCodec::FLAC) {
fprintf(stderr, "Warning: opus and flac audio codecs has been temporary disabled, using aac audio codec instead\n");
audio_codec_to_use = "aac";
audio_codec = AudioCodec::AAC;
}
bool overclock = false;
@@ -2397,58 +2397,21 @@ int main(int argc, char **argv) {
swr_init(swr);
}
const double target_audio_hz = 1.0 / (double)audio_track.codec_context->sample_rate;
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);
// Remove this for now, it doesn't work well for everybody. The timing is different depending on system
#if 0
// 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.
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;
}
#endif
const int64_t no_input_sleep_ms = 500;
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);
sound_buffer_size = sound_device_read_next_chunk(&audio_device.sound_device, &sound_buffer, 0.5);
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(timeout_ms * 1000);
usleep(no_input_sleep_ms * 1000);
continue;
}
@@ -2459,63 +2422,19 @@ int main(int argc, char **argv) {
break;
}
// TODO: Is this |received_audio_time| really correct?
int64_t num_missing_frames = std::round((this_audio_frame_time - received_audio_time) / target_audio_hz / (int64_t)audio_track.codec_context->frame_size);
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) {
if(!got_audio_data) {
// 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;
// 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) {
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");
}
} else {
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;
}
}
if(!audio_device.sound_device.handle)
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?
if(needs_audio_conversion)
swr_convert(swr, &audio_device.frame->data[0], audio_track.codec_context->frame_size, (const uint8_t**)&sound_buffer, audio_track.codec_context->frame_size);
else
audio_device.frame->data[0] = (uint8_t*)sound_buffer;
const int64_t new_pts = (this_audio_frame_time - record_start_time) * AV_TIME_BASE;
if(new_pts == audio_device.frame->pts)
continue;
audio_device.frame->pts = new_pts;
//audio_device.frame->linesize[0] = sound_buffer_size / 2;
if(audio_track.graph) {
std::lock_guard<std::mutex> lock(audio_filter_mutex);
@@ -2532,8 +2451,39 @@ int main(int argc, char **argv) {
fprintf(stderr, "Failed to encode audio!\n");
}
}
}
audio_device.frame->pts += audio_track.codec_context->frame_size;
if(!audio_device.sound_device.handle)
usleep(no_input_sleep_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?
if(needs_audio_conversion)
swr_convert(swr, &audio_device.frame->data[0], audio_track.codec_context->frame_size, (const uint8_t**)&sound_buffer, audio_track.codec_context->frame_size);
else
audio_device.frame->data[0] = (uint8_t*)sound_buffer;
const int64_t new_pts = (this_audio_frame_time - record_start_time) * AV_TIME_BASE;
if(new_pts == audio_device.frame->pts)
continue;
audio_device.frame->pts = new_pts;
//audio_device.frame->linesize[0] = sound_buffer_size / 2;
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 {
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");
}
}
}
}
@@ -2571,7 +2521,11 @@ int main(int argc, char **argv) {
int err = 0;
while ((err = av_buffersink_get_frame(audio_track.sink, aframe)) >= 0) {
aframe->pts = audio_track.pts;
const int64_t new_pts = ((clock_get_monotonic_seconds() - paused_time_offset) - record_start_time) * AV_TIME_BASE;
if(new_pts == aframe->pts)
continue;
aframe->pts = new_pts;
//aframe->linesize[0] = sound_buffer_size / 2;
err = avcodec_send_frame(audio_track.codec_context, aframe);
if(err >= 0){
// TODO: Move to separate thread because this could write to network (for example when livestreaming)
@@ -2580,7 +2534,6 @@ int main(int argc, char **argv) {
fprintf(stderr, "Failed to encode audio!\n");
}
av_frame_unref(aframe);
audio_track.pts += audio_track.codec_context->frame_size;
}
}
}