PapaTutuWawa/gpu-screen-recorder
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Add code for variable framerate

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This commit is contained in:
dec05eba 2023-03-26 20:54:08 +02:00
parent d92ecea69e
commit bd925e0beb
1 changed files with 68 additions and 15 deletions
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73
src/main.cpp
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@ -153,6 +153,11 @@ enum class PixelFormat {
YUV444
};
enum class FramerateMode {
CONSTANT,
VARIABLE
};
static int x11_error_handler(Display *dpy, XErrorEvent *ev) {
return 0;
}
@ -306,7 +311,7 @@ static AVSampleFormat audio_format_to_sample_format(const AudioFormat audio_form
return AV_SAMPLE_FMT_S16;
}
static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_codec) {
static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_codec, FramerateMode framerate_mode) {
const AVCodec *codec = avcodec_find_encoder(audio_codec_get_id(audio_codec));
if (!codec) {
fprintf(stderr, "Error: Could not find %s audio encoder\n", audio_codec_get_name(audio_codec));
@ -330,7 +335,7 @@ 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 = framerate_mode == FramerateMode::CONSTANT ? codec_context->sample_rate : AV_TIME_BASE;
codec_context->framerate.num = fps;
codec_context->framerate.den = 1;
codec_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
@ -340,7 +345,7 @@ static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_code
static AVCodecContext *create_video_codec_context(AVPixelFormat pix_fmt,
VideoQuality video_quality,
int fps, const AVCodec *codec, bool is_livestream, gpu_vendor vendor) {
int fps, const AVCodec *codec, bool is_livestream, gpu_vendor vendor, FramerateMode framerate_mode) {
AVCodecContext *codec_context = avcodec_alloc_context3(codec);
@ -353,7 +358,7 @@ static AVCodecContext *create_video_codec_context(AVPixelFormat pix_fmt,
// timebase should be 1/framerate and timestamp increments should be
// identical to 1
codec_context->time_base.num = 1;
codec_context->time_base.den = fps;
codec_context->time_base.den = framerate_mode == FramerateMode::CONSTANT ? fps : AV_TIME_BASE;
codec_context->framerate.num = fps;
codec_context->framerate.den = 1;
codec_context->sample_aspect_ratio.num = 0;
@ -448,7 +453,7 @@ static bool check_if_codec_valid_for_hardware(const AVCodec *codec, gpu_vendor v
bool success = false;
// Do not use AV_PIX_FMT_CUDA because we dont want to do full check with hardware context
AVCodecContext *codec_context = create_video_codec_context(AV_PIX_FMT_YUV420P, VideoQuality::VERY_HIGH, 60, codec, false, vendor);
AVCodecContext *codec_context = create_video_codec_context(AV_PIX_FMT_YUV420P, VideoQuality::VERY_HIGH, 60, codec, false, vendor, FramerateMode::CONSTANT);
codec_context->width = 1920;
codec_context->height = 1080;
if(codec_context) {
@ -751,6 +756,7 @@ struct AudioDevice {
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
};
// TODO: Cleanup
struct AudioTrack {
AVCodecContext *codec_context = nullptr;
AVFrame *frame = nullptr;
@ -1194,6 +1200,8 @@ int main(int argc, char **argv) {
usage();
}
FramerateMode framerate_mode = FramerateMode::CONSTANT;
const Arg &audio_input_arg = args["-a"];
const std::vector<AudioInput> audio_inputs = get_pulseaudio_inputs();
std::vector<MergedAudioInputs> requested_audio_inputs;
@ -1554,7 +1562,7 @@ int main(int argc, char **argv) {
AVStream *video_stream = nullptr;
std::vector<AudioTrack> audio_tracks;
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);
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);
if(replay_buffer_size_secs == -1)
video_stream = create_stream(av_format_context, video_codec_context);
@ -1569,7 +1577,7 @@ int main(int argc, char **argv) {
int audio_stream_index = VIDEO_STREAM_INDEX + 1;
for(const MergedAudioInputs &merged_audio_inputs : requested_audio_inputs) {
AVCodecContext *audio_codec_context = create_audio_codec_context(fps, audio_codec);
AVCodecContext *audio_codec_context = create_audio_codec_context(fps, audio_codec, framerate_mode);
AVStream *audio_stream = nullptr;
if(replay_buffer_size_secs == -1)
@ -1699,7 +1707,7 @@ int main(int argc, char **argv) {
for(AudioTrack &audio_track : audio_tracks) {
for(AudioDevice &audio_device : audio_track.audio_devices) {
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 {
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 {
const AVSampleFormat sound_device_sample_format = audio_format_to_sample_format(audio_codec_context_get_audio_format(audio_track.codec_context));
const bool needs_audio_conversion = audio_track.codec_context->sample_fmt != sound_device_sample_format;
SwrContext *swr = nullptr;
@ -1721,6 +1729,7 @@ int main(int argc, char **argv) {
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);
int64_t prev_pts = 0;
while(running) {
void *sound_buffer;
@ -1739,6 +1748,7 @@ 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.frame->nb_samples);
if(got_audio_data)
num_missing_frames = std::max((int64_t)0, num_missing_frames - 1);
@ -1771,8 +1781,17 @@ int main(int argc, char **argv) {
fprintf(stderr, "Error: failed to add audio frame to filter\n");
}
} else {
if(framerate_mode == FramerateMode::CONSTANT) {
audio_track.frame->pts = audio_track.pts;
audio_track.pts += audio_track.frame->nb_samples;
} else {
audio_track.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
const bool same_pts = audio_track.frame->pts == prev_pts;
prev_pts = audio_track.frame->pts;
if(same_pts)
continue;
}
ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
if(ret >= 0) {
// TODO: Move to separate thread because this could write to network (for example when livestreaming)
@ -1801,8 +1820,17 @@ int main(int argc, char **argv) {
fprintf(stderr, "Error: failed to add audio frame to filter\n");
}
} else {
if(framerate_mode == FramerateMode::CONSTANT) {
audio_track.frame->pts = audio_track.pts;
audio_track.pts += audio_track.frame->nb_samples;
} else {
audio_track.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
const bool same_pts = audio_track.frame->pts == prev_pts;
prev_pts = audio_track.frame->pts;
if(same_pts)
continue;
}
ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
if(ret >= 0) {
// TODO: Move to separate thread because this could write to network (for example when livestreaming)
@ -1838,7 +1866,9 @@ int main(int argc, char **argv) {
std::mutex video_frame_mutex;
std::thread video_send_encode_thread([&]() {
int64_t video_pts_counter = 0;
int64_t video_prev_pts = 0;
AVFrame *video_frame = nullptr;
while(running) {
{
std::unique_lock<std::mutex> lock(video_frame_mutex);
@ -1856,11 +1886,19 @@ int main(int argc, char **argv) {
const double this_video_frame_time = clock_get_monotonic_seconds();
const int64_t expected_frames = std::round((this_video_frame_time - start_time_pts) / target_fps);
const int num_frames = std::max(0L, expected_frames - video_pts_counter);
const int num_frames = framerate_mode == FramerateMode::CONSTANT ? std::max(0L, expected_frames - video_pts_counter) : 1;
// 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_frames; ++i) {
if(framerate_mode == FramerateMode::CONSTANT) {
video_frame->pts = video_pts_counter + i;
} else {
video_frame->pts = (this_video_frame_time - record_start_time) * (double)AV_TIME_BASE;
const bool same_pts = video_frame->pts == video_prev_pts;
video_prev_pts = video_frame->pts;
if(same_pts)
continue;
}
int ret = avcodec_send_frame(video_codec_context, video_frame);
if(ret == 0) {
@ -1878,6 +1916,7 @@ int main(int argc, char **argv) {
}
});
int64_t audio_prev_pts = 0;
while(running) {
double frame_start = clock_get_monotonic_seconds();
@ -1899,6 +1938,19 @@ int main(int argc, char **argv) {
while ((err = av_buffersink_get_frame(audio_track.sink, aframe)) >= 0) {
aframe->pts = audio_track.pts;
audio_track.pts += audio_track.codec_context->frame_size;
if(framerate_mode == FramerateMode::CONSTANT) {
aframe->pts = audio_track.pts;
audio_track.pts += audio_track.codec_context->frame_size;
} else {
const double this_audio_frame_time = clock_get_monotonic_seconds();
aframe->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
const bool same_pts = aframe->pts == audio_prev_pts;
audio_prev_pts = aframe->pts;
if(same_pts)
continue;
}
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)
@ -1953,7 +2005,6 @@ int main(int argc, char **argv) {
}
running = 0;
av_frame_free(&aframe);
if(save_replay_thread.valid()) {
save_replay_thread.get();
@ -1973,6 +2024,8 @@ int main(int argc, char **argv) {
}
video_send_encode_thread.join();
av_frame_free(&aframe);
if(latest_video_frame) {
av_frame_free(&latest_video_frame);
latest_video_frame = nullptr;