Add code for variable framerate
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d92ecea69e
commit
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83
src/main.cpp
83
src/main.cpp
@ -153,6 +153,11 @@ enum class PixelFormat {
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YUV444
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};
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enum class FramerateMode {
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CONSTANT,
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VARIABLE
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};
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static int x11_error_handler(Display *dpy, XErrorEvent *ev) {
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return 0;
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}
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@ -306,7 +311,7 @@ static AVSampleFormat audio_format_to_sample_format(const AudioFormat audio_form
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return AV_SAMPLE_FMT_S16;
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}
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static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_codec) {
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static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_codec, FramerateMode framerate_mode) {
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const AVCodec *codec = avcodec_find_encoder(audio_codec_get_id(audio_codec));
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if (!codec) {
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fprintf(stderr, "Error: Could not find %s audio encoder\n", audio_codec_get_name(audio_codec));
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@ -330,7 +335,7 @@ static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_code
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#endif
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codec_context->time_base.num = 1;
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codec_context->time_base.den = codec_context->sample_rate;
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codec_context->time_base.den = framerate_mode == FramerateMode::CONSTANT ? codec_context->sample_rate : AV_TIME_BASE;
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codec_context->framerate.num = fps;
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codec_context->framerate.den = 1;
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codec_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
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@ -340,7 +345,7 @@ static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_code
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static AVCodecContext *create_video_codec_context(AVPixelFormat pix_fmt,
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VideoQuality video_quality,
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int fps, const AVCodec *codec, bool is_livestream, gpu_vendor vendor) {
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int fps, const AVCodec *codec, bool is_livestream, gpu_vendor vendor, FramerateMode framerate_mode) {
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AVCodecContext *codec_context = avcodec_alloc_context3(codec);
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@ -353,7 +358,7 @@ static AVCodecContext *create_video_codec_context(AVPixelFormat pix_fmt,
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// timebase should be 1/framerate and timestamp increments should be
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// identical to 1
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codec_context->time_base.num = 1;
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codec_context->time_base.den = fps;
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codec_context->time_base.den = framerate_mode == FramerateMode::CONSTANT ? fps : AV_TIME_BASE;
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codec_context->framerate.num = fps;
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codec_context->framerate.den = 1;
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codec_context->sample_aspect_ratio.num = 0;
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@ -448,7 +453,7 @@ static bool check_if_codec_valid_for_hardware(const AVCodec *codec, gpu_vendor v
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bool success = false;
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// Do not use AV_PIX_FMT_CUDA because we dont want to do full check with hardware context
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AVCodecContext *codec_context = create_video_codec_context(AV_PIX_FMT_YUV420P, VideoQuality::VERY_HIGH, 60, codec, false, vendor);
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AVCodecContext *codec_context = create_video_codec_context(AV_PIX_FMT_YUV420P, VideoQuality::VERY_HIGH, 60, codec, false, vendor, FramerateMode::CONSTANT);
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codec_context->width = 1920;
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codec_context->height = 1080;
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if(codec_context) {
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@ -751,6 +756,7 @@ struct AudioDevice {
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std::thread thread; // TODO: Instead of having a thread for each track, have one thread for all threads and read the data with non-blocking read
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};
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// TODO: Cleanup
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struct AudioTrack {
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AVCodecContext *codec_context = nullptr;
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AVFrame *frame = nullptr;
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@ -1194,6 +1200,8 @@ int main(int argc, char **argv) {
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usage();
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}
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FramerateMode framerate_mode = FramerateMode::CONSTANT;
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const Arg &audio_input_arg = args["-a"];
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const std::vector<AudioInput> audio_inputs = get_pulseaudio_inputs();
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std::vector<MergedAudioInputs> requested_audio_inputs;
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@ -1554,7 +1562,7 @@ int main(int argc, char **argv) {
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AVStream *video_stream = nullptr;
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std::vector<AudioTrack> audio_tracks;
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AVCodecContext *video_codec_context = create_video_codec_context(gpu_inf.vendor == GPU_VENDOR_NVIDIA ? AV_PIX_FMT_CUDA : AV_PIX_FMT_VAAPI, quality, fps, video_codec_f, is_livestream, gpu_inf.vendor);
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AVCodecContext *video_codec_context = create_video_codec_context(gpu_inf.vendor == GPU_VENDOR_NVIDIA ? AV_PIX_FMT_CUDA : AV_PIX_FMT_VAAPI, quality, fps, video_codec_f, is_livestream, gpu_inf.vendor, framerate_mode);
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if(replay_buffer_size_secs == -1)
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video_stream = create_stream(av_format_context, video_codec_context);
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@ -1569,7 +1577,7 @@ int main(int argc, char **argv) {
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int audio_stream_index = VIDEO_STREAM_INDEX + 1;
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for(const MergedAudioInputs &merged_audio_inputs : requested_audio_inputs) {
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AVCodecContext *audio_codec_context = create_audio_codec_context(fps, audio_codec);
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AVCodecContext *audio_codec_context = create_audio_codec_context(fps, audio_codec, framerate_mode);
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AVStream *audio_stream = nullptr;
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if(replay_buffer_size_secs == -1)
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@ -1699,7 +1707,7 @@ int main(int argc, char **argv) {
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for(AudioTrack &audio_track : audio_tracks) {
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for(AudioDevice &audio_device : audio_track.audio_devices) {
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audio_device.thread = std::thread([record_start_time, replay_buffer_size_secs, &frame_data_queue, &frames_erased, &audio_track, empty_audio, &audio_device, &audio_filter_mutex, &write_output_mutex](AVFormatContext *av_format_context) mutable {
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audio_device.thread = std::thread([record_start_time, replay_buffer_size_secs, &frame_data_queue, &frames_erased, &audio_track, empty_audio, &audio_device, &audio_filter_mutex, &write_output_mutex, framerate_mode](AVFormatContext *av_format_context) mutable {
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const AVSampleFormat sound_device_sample_format = audio_format_to_sample_format(audio_codec_context_get_audio_format(audio_track.codec_context));
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const bool needs_audio_conversion = audio_track.codec_context->sample_fmt != sound_device_sample_format;
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SwrContext *swr = nullptr;
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@ -1721,6 +1729,7 @@ int main(int argc, char **argv) {
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const double target_audio_hz = 1.0 / (double)audio_track.codec_context->sample_rate;
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double received_audio_time = clock_get_monotonic_seconds();
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const int64_t timeout_ms = std::round((1000.0 / (double)audio_track.codec_context->sample_rate) * 1000.0);
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int64_t prev_pts = 0;
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while(running) {
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void *sound_buffer;
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@ -1739,6 +1748,7 @@ int main(int argc, char **argv) {
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break;
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}
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// TODO: Is this |received_audio_time| really correct?
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int64_t num_missing_frames = std::round((this_audio_frame_time - received_audio_time) / target_audio_hz / (int64_t)audio_track.frame->nb_samples);
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if(got_audio_data)
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num_missing_frames = std::max((int64_t)0, num_missing_frames - 1);
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@ -1771,8 +1781,17 @@ int main(int argc, char **argv) {
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fprintf(stderr, "Error: failed to add audio frame to filter\n");
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}
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} else {
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audio_track.frame->pts = audio_track.pts;
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audio_track.pts += audio_track.frame->nb_samples;
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if(framerate_mode == FramerateMode::CONSTANT) {
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audio_track.frame->pts = audio_track.pts;
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audio_track.pts += audio_track.frame->nb_samples;
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} else {
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audio_track.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
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const bool same_pts = audio_track.frame->pts == prev_pts;
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prev_pts = audio_track.frame->pts;
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if(same_pts)
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continue;
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}
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ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
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if(ret >= 0) {
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// TODO: Move to separate thread because this could write to network (for example when livestreaming)
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@ -1801,8 +1820,17 @@ int main(int argc, char **argv) {
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fprintf(stderr, "Error: failed to add audio frame to filter\n");
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}
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} else {
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audio_track.frame->pts = audio_track.pts;
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audio_track.pts += audio_track.frame->nb_samples;
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if(framerate_mode == FramerateMode::CONSTANT) {
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audio_track.frame->pts = audio_track.pts;
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audio_track.pts += audio_track.frame->nb_samples;
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} else {
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audio_track.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
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const bool same_pts = audio_track.frame->pts == prev_pts;
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prev_pts = audio_track.frame->pts;
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if(same_pts)
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continue;
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}
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ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
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if(ret >= 0) {
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// TODO: Move to separate thread because this could write to network (for example when livestreaming)
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@ -1838,7 +1866,9 @@ int main(int argc, char **argv) {
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std::mutex video_frame_mutex;
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std::thread video_send_encode_thread([&]() {
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int64_t video_pts_counter = 0;
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int64_t video_prev_pts = 0;
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AVFrame *video_frame = nullptr;
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while(running) {
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{
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std::unique_lock<std::mutex> lock(video_frame_mutex);
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@ -1856,11 +1886,19 @@ int main(int argc, char **argv) {
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const double this_video_frame_time = clock_get_monotonic_seconds();
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const int64_t expected_frames = std::round((this_video_frame_time - start_time_pts) / target_fps);
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const int num_frames = std::max(0L, expected_frames - video_pts_counter);
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const int num_frames = framerate_mode == FramerateMode::CONSTANT ? std::max(0L, expected_frames - video_pts_counter) : 1;
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// TODO: Check if duplicate frame can be saved just by writing it with a different pts instead of sending it again
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for(int i = 0; i < num_frames; ++i) {
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video_frame->pts = video_pts_counter + i;
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if(framerate_mode == FramerateMode::CONSTANT) {
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video_frame->pts = video_pts_counter + i;
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} else {
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video_frame->pts = (this_video_frame_time - record_start_time) * (double)AV_TIME_BASE;
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const bool same_pts = video_frame->pts == video_prev_pts;
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video_prev_pts = video_frame->pts;
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if(same_pts)
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continue;
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}
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int ret = avcodec_send_frame(video_codec_context, video_frame);
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if(ret == 0) {
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@ -1878,6 +1916,7 @@ int main(int argc, char **argv) {
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}
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});
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int64_t audio_prev_pts = 0;
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while(running) {
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double frame_start = clock_get_monotonic_seconds();
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@ -1899,6 +1938,19 @@ int main(int argc, char **argv) {
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while ((err = av_buffersink_get_frame(audio_track.sink, aframe)) >= 0) {
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aframe->pts = audio_track.pts;
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audio_track.pts += audio_track.codec_context->frame_size;
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if(framerate_mode == FramerateMode::CONSTANT) {
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aframe->pts = audio_track.pts;
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audio_track.pts += audio_track.codec_context->frame_size;
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} else {
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const double this_audio_frame_time = clock_get_monotonic_seconds();
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aframe->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
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const bool same_pts = aframe->pts == audio_prev_pts;
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audio_prev_pts = aframe->pts;
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if(same_pts)
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continue;
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}
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err = avcodec_send_frame(audio_track.codec_context, aframe);
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if(err >= 0){
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// TODO: Move to separate thread because this could write to network (for example when livestreaming)
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@ -1953,7 +2005,6 @@ int main(int argc, char **argv) {
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}
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running = 0;
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av_frame_free(&aframe);
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if(save_replay_thread.valid()) {
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save_replay_thread.get();
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@ -1973,6 +2024,8 @@ int main(int argc, char **argv) {
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}
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video_send_encode_thread.join();
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av_frame_free(&aframe);
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if(latest_video_frame) {
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av_frame_free(&latest_video_frame);
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latest_video_frame = nullptr;
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