diff options
Diffstat (limited to 'src/main.cpp')
| -rw-r--r-- | src/main.cpp | 50 |
1 files changed, 39 insertions, 11 deletions
diff --git a/src/main.cpp b/src/main.cpp index 1c02984..1b88cd9 100644 --- a/src/main.cpp +++ b/src/main.cpp @@ -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"); + } + } + } + + 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); - 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"); + 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"); + } } } |