PapaTutuWawa/gpu-screen-recorder
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Temporary disable opus/flac because it breaks multiple audio, fix possible audio/video desync on nvidia

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This commit is contained in:
dec05eba 2023-04-27 13:57:01 +02:00
parent e1c613666e
commit ec22eb6da4
2 changed files with 52 additions and 49 deletions
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3
TODO
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@ -59,4 +59,5 @@ Intel is a bit weird with monitor capture and multiple monitors. If one of the m
Is that only the case when the primary monitor is rotated? Also the primary monitor becomes position 0, 0 so crtc (x11 randr) position doesn't match the drm pos. Maybe get monitor position and size from drm instead.
How about if multiple monitors are rotated?
When using multiple monitors kms grab the target monitor instead of the whole screen.
When using multiple monitors kms grab the target monitor instead of the whole screen.
Enable opus/flac again. It's broken right now when merging audio inputs. The audio gets a lot of static noise!

98
src/main.cpp
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@ -260,7 +260,7 @@ static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_code
#endif
codec_context->time_base.num = 1;
codec_context->time_base.den = framerate_mode == FramerateMode::CONSTANT ? codec_context->sample_rate : AV_TIME_BASE;
codec_context->time_base.den = AV_TIME_BASE;
codec_context->framerate.num = fps;
codec_context->framerate.den = 1;
codec_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
@ -633,7 +633,7 @@ static void open_video(AVCodecContext *codec_context, VideoQuality video_quality
}
static void usage_header() {
fprintf(stderr, "usage: gpu-screen-recorder -w <window_id|monitor|focused> [-c <container_format>] [-s WxH] -f <fps> [-a <audio_input>] [-q <quality>] [-r <replay_buffer_size_sec>] [-k h264|h265] [-ac aac|opus|flac] [-oc yes|no] [-v yes|no] [-h|--help] [-o <output_file>]\n");
fprintf(stderr, "usage: gpu-screen-recorder -w <window_id|monitor|focused> [-c <container_format>] [-s WxH] -f <fps> [-a <audio_input>] [-q <quality>] [-r <replay_buffer_size_sec>] [-k h264|h265] [-ac aac|opus|flac] [-oc yes|no] [-fm cfr|vfr] [-v yes|no] [-h|--help] [-o <output_file>]\n");
}
static void usage_full() {
@ -677,6 +677,8 @@ static void usage_full() {
fprintf(stderr, " is dropped when you record a game. Only needed if you are recording a game that is bottlenecked by GPU.\n");
fprintf(stderr, " Works only if your have \"Coolbits\" set to \"12\" in NVIDIA X settings, see README for more information. Note! use at your own risk! Optional, disabled by default.\n");
fprintf(stderr, "\n");
fprintf(stderr, " -fm Framerate mode. Should be either 'cfr' or 'vfr'. Defaults to 'cfr' on NVIDIA and 'vfr' on AMD/Intel.\n");
fprintf(stderr, "\n");
fprintf(stderr, " -v Prints per second, fps updates. Optional, set to 'yes' by default.\n");
fprintf(stderr, "\n");
fprintf(stderr, " -h Show this help.\n");
@ -781,7 +783,6 @@ struct AudioTrack {
std::vector<AudioDevice> audio_devices;
AVFilterGraph *graph = nullptr;
AVFilterContext *sink = nullptr;
int64_t pts = 0;
int stream_index = 0;
};
@ -944,8 +945,7 @@ static bool is_livestream_path(const char *str) {
}
// TODO: Proper cleanup
static int init_filter_graph(AVCodecContext *audio_codec_context, AVFilterGraph **graph, AVFilterContext **sink, std::vector<AVFilterContext*> &src_filter_ctx, size_t num_sources)
{
static int init_filter_graph(AVCodecContext *audio_codec_context, AVFilterGraph **graph, AVFilterContext **sink, std::vector<AVFilterContext*> &src_filter_ctx, size_t num_sources) {
char ch_layout[64];
int err = 0;
@ -973,10 +973,11 @@ static int init_filter_graph(AVCodecContext *audio_codec_context, AVFilterGraph
#else
av_channel_layout_describe(&audio_codec_context->ch_layout, ch_layout, sizeof(ch_layout));
#endif
av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
av_opt_set (abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(audio_codec_context->sample_fmt), AV_OPT_SEARCH_CHILDREN);
av_opt_set_q (abuffer_ctx, "time_base", { 1, audio_codec_context->sample_rate }, AV_OPT_SEARCH_CHILDREN);
av_opt_set_int(abuffer_ctx, "sample_rate", audio_codec_context->sample_rate, AV_OPT_SEARCH_CHILDREN);
av_opt_set_q (abuffer_ctx, "time_base", audio_codec_context->time_base, AV_OPT_SEARCH_CHILDREN);
av_opt_set_int(abuffer_ctx, "sample_rate", audio_codec_context->sample_rate, AV_OPT_SEARCH_CHILDREN);
av_opt_set_int(abuffer_ctx, "bit_rate", audio_codec_context->bit_rate, AV_OPT_SEARCH_CHILDREN);
err = avfilter_init_str(abuffer_ctx, NULL);
if (err < 0) {
@ -997,9 +998,7 @@ static int init_filter_graph(AVCodecContext *audio_codec_context, AVFilterGraph
snprintf(args, sizeof(args), "inputs=%d", (int)num_sources);
AVFilterContext *mix_ctx;
err = avfilter_graph_create_filter(&mix_ctx, mix_filter, "amix",
args, NULL, filter_graph);
err = avfilter_graph_create_filter(&mix_ctx, mix_filter, "amix", args, NULL, filter_graph);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio amix filter\n");
return err;
@ -1095,6 +1094,7 @@ int main(int argc, char **argv) {
{ "-k", Arg { {}, true, false } },
{ "-ac", Arg { {}, true, false } },
{ "-oc", Arg { {}, true, false } },
{ "-fm", Arg { {}, true, false } },
{ "-pixfmt", Arg { {}, true, false } },
{ "-v", Arg { {}, true, false } },
};
@ -1143,7 +1143,7 @@ int main(int argc, char **argv) {
AudioCodec audio_codec = AudioCodec::OPUS;
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;
@ -1156,6 +1156,12 @@ int main(int argc, char **argv) {
usage();
}
if(audio_codec != AudioCodec::AAC) {
audio_codec_to_use = "aac";
audio_codec = AudioCodec::AAC;
fprintf(stderr, "Info: audio codec is forcefully set to aac at the moment because of issues with opus/flac. This is a temporary issue\n");
}
bool overclock = false;
const char *overclock_str = args["-oc"].value();
if(!overclock_str)
@ -1305,8 +1311,20 @@ int main(int argc, char **argv) {
}
// TODO: Fix constant framerate not working properly on amd/intel because capture framerate gets locked to the same framerate as
// game framerate, which doesn't work well when you need to encode multiple duplicate frames.
const FramerateMode framerate_mode = gpu_inf.vendor == GSR_GPU_VENDOR_NVIDIA ? FramerateMode::CONSTANT : FramerateMode::VARIABLE;
// game framerate, which doesn't work well when you need to encode multiple duplicate frames (AMD/Intel is slow at encoding!).
FramerateMode framerate_mode;
const char *framerate_mode_str = args["-fm"].value();
if(!framerate_mode_str)
framerate_mode_str = gpu_inf.vendor == GSR_GPU_VENDOR_NVIDIA ? "cfr" : "vfr";
if(strcmp(framerate_mode_str, "cfr") == 0) {
framerate_mode = FramerateMode::CONSTANT;
} else if(strcmp(framerate_mode_str, "vfr") == 0) {
framerate_mode = FramerateMode::VARIABLE;
} else {
fprintf(stderr, "Error: -fm should either be either 'cfr' or 'vfr', got: '%s'\n", framerate_mode_str);
usage();
}
const char *screen_region = args["-s"].value();
const char *window_str = args["-w"].value();
@ -1654,7 +1672,6 @@ int main(int argc, char **argv) {
audio_track.audio_devices = std::move(audio_devices);
audio_track.graph = graph;
audio_track.sink = sink;
audio_track.pts = 0;
audio_track.stream_index = audio_stream_index;
audio_tracks.push_back(std::move(audio_track));
++audio_stream_index;
@ -1795,16 +1812,11 @@ 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;
}
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) {
@ -1827,6 +1839,12 @@ int main(int argc, char **argv) {
else
audio_track.frame->data[0] = (uint8_t*)sound_buffer;
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;
if(audio_track.graph) {
std::lock_guard<std::mutex> lock(audio_filter_mutex);
// TODO: av_buffersrc_add_frame
@ -1834,17 +1852,6 @@ 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)
@ -1891,18 +1898,13 @@ int main(int argc, char **argv) {
int err = 0;
while ((err = av_buffersink_get_frame(audio_track.sink, aframe)) >= 0) {
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) {
av_frame_unref(aframe);
continue;
}
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) {
av_frame_unref(aframe);
continue;
}
err = avcodec_send_frame(audio_track.codec_context, aframe);