Temporary disable opus/flac because it breaks multiple audio, fix possible audio/video desync on nvidia
This commit is contained in:
parent
e1c613666e
commit
ec22eb6da4
3
TODO
3
TODO
@ -59,4 +59,5 @@ Intel is a bit weird with monitor capture and multiple monitors. If one of the m
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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.
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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.
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How about if multiple monitors are rotated?
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How about if multiple monitors are rotated?
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When using multiple monitors kms grab the target monitor instead of the whole screen.
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When using multiple monitors kms grab the target monitor instead of the whole screen.
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Enable opus/flac again. It's broken right now when merging audio inputs. The audio gets a lot of static noise!
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98
src/main.cpp
98
src/main.cpp
@ -260,7 +260,7 @@ static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_code
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#endif
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#endif
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codec_context->time_base.num = 1;
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codec_context->time_base.num = 1;
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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->time_base.den = AV_TIME_BASE;
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codec_context->framerate.num = fps;
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codec_context->framerate.num = fps;
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codec_context->framerate.den = 1;
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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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codec_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
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@ -633,7 +633,7 @@ static void open_video(AVCodecContext *codec_context, VideoQuality video_quality
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}
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}
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static void usage_header() {
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static void usage_header() {
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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");
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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");
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}
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}
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static void usage_full() {
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static void usage_full() {
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@ -677,6 +677,8 @@ static void usage_full() {
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fprintf(stderr, " is dropped when you record a game. Only needed if you are recording a game that is bottlenecked by GPU.\n");
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fprintf(stderr, " is dropped when you record a game. Only needed if you are recording a game that is bottlenecked by GPU.\n");
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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");
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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");
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fprintf(stderr, "\n");
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fprintf(stderr, "\n");
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fprintf(stderr, " -fm Framerate mode. Should be either 'cfr' or 'vfr'. Defaults to 'cfr' on NVIDIA and 'vfr' on AMD/Intel.\n");
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fprintf(stderr, "\n");
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fprintf(stderr, " -v Prints per second, fps updates. Optional, set to 'yes' by default.\n");
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fprintf(stderr, " -v Prints per second, fps updates. Optional, set to 'yes' by default.\n");
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fprintf(stderr, "\n");
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fprintf(stderr, "\n");
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fprintf(stderr, " -h Show this help.\n");
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fprintf(stderr, " -h Show this help.\n");
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@ -781,7 +783,6 @@ struct AudioTrack {
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std::vector<AudioDevice> audio_devices;
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std::vector<AudioDevice> audio_devices;
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AVFilterGraph *graph = nullptr;
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AVFilterGraph *graph = nullptr;
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AVFilterContext *sink = nullptr;
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AVFilterContext *sink = nullptr;
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int64_t pts = 0;
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int stream_index = 0;
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int stream_index = 0;
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};
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};
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@ -944,8 +945,7 @@ static bool is_livestream_path(const char *str) {
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}
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}
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// TODO: Proper cleanup
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// TODO: Proper cleanup
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static int init_filter_graph(AVCodecContext *audio_codec_context, AVFilterGraph **graph, AVFilterContext **sink, std::vector<AVFilterContext*> &src_filter_ctx, size_t num_sources)
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static int init_filter_graph(AVCodecContext *audio_codec_context, AVFilterGraph **graph, AVFilterContext **sink, std::vector<AVFilterContext*> &src_filter_ctx, size_t num_sources) {
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{
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char ch_layout[64];
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char ch_layout[64];
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int err = 0;
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int err = 0;
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@ -973,10 +973,11 @@ static int init_filter_graph(AVCodecContext *audio_codec_context, AVFilterGraph
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#else
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#else
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av_channel_layout_describe(&audio_codec_context->ch_layout, ch_layout, sizeof(ch_layout));
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av_channel_layout_describe(&audio_codec_context->ch_layout, ch_layout, sizeof(ch_layout));
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#endif
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#endif
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av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
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av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
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av_opt_set (abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(audio_codec_context->sample_fmt), AV_OPT_SEARCH_CHILDREN);
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av_opt_set (abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(audio_codec_context->sample_fmt), AV_OPT_SEARCH_CHILDREN);
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av_opt_set_q (abuffer_ctx, "time_base", { 1, audio_codec_context->sample_rate }, AV_OPT_SEARCH_CHILDREN);
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av_opt_set_q (abuffer_ctx, "time_base", audio_codec_context->time_base, AV_OPT_SEARCH_CHILDREN);
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av_opt_set_int(abuffer_ctx, "sample_rate", audio_codec_context->sample_rate, AV_OPT_SEARCH_CHILDREN);
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av_opt_set_int(abuffer_ctx, "sample_rate", audio_codec_context->sample_rate, AV_OPT_SEARCH_CHILDREN);
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av_opt_set_int(abuffer_ctx, "bit_rate", audio_codec_context->bit_rate, AV_OPT_SEARCH_CHILDREN);
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err = avfilter_init_str(abuffer_ctx, NULL);
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err = avfilter_init_str(abuffer_ctx, NULL);
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if (err < 0) {
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if (err < 0) {
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@ -997,9 +998,7 @@ static int init_filter_graph(AVCodecContext *audio_codec_context, AVFilterGraph
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snprintf(args, sizeof(args), "inputs=%d", (int)num_sources);
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snprintf(args, sizeof(args), "inputs=%d", (int)num_sources);
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AVFilterContext *mix_ctx;
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AVFilterContext *mix_ctx;
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err = avfilter_graph_create_filter(&mix_ctx, mix_filter, "amix",
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err = avfilter_graph_create_filter(&mix_ctx, mix_filter, "amix", args, NULL, filter_graph);
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args, NULL, filter_graph);
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if (err < 0) {
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if (err < 0) {
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av_log(NULL, AV_LOG_ERROR, "Cannot create audio amix filter\n");
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av_log(NULL, AV_LOG_ERROR, "Cannot create audio amix filter\n");
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return err;
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return err;
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@ -1095,6 +1094,7 @@ int main(int argc, char **argv) {
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{ "-k", Arg { {}, true, false } },
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{ "-k", Arg { {}, true, false } },
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{ "-ac", Arg { {}, true, false } },
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{ "-ac", Arg { {}, true, false } },
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{ "-oc", Arg { {}, true, false } },
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{ "-oc", Arg { {}, true, false } },
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{ "-fm", Arg { {}, true, false } },
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{ "-pixfmt", Arg { {}, true, false } },
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{ "-pixfmt", Arg { {}, true, false } },
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{ "-v", Arg { {}, true, false } },
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{ "-v", Arg { {}, true, false } },
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};
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};
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@ -1143,7 +1143,7 @@ int main(int argc, char **argv) {
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AudioCodec audio_codec = AudioCodec::OPUS;
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AudioCodec audio_codec = AudioCodec::OPUS;
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const char *audio_codec_to_use = args["-ac"].value();
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const char *audio_codec_to_use = args["-ac"].value();
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if(!audio_codec_to_use)
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if(!audio_codec_to_use)
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audio_codec_to_use = "opus";
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audio_codec_to_use = "aac";
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if(strcmp(audio_codec_to_use, "aac") == 0) {
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if(strcmp(audio_codec_to_use, "aac") == 0) {
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audio_codec = AudioCodec::AAC;
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audio_codec = AudioCodec::AAC;
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@ -1156,6 +1156,12 @@ int main(int argc, char **argv) {
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usage();
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usage();
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}
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}
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if(audio_codec != AudioCodec::AAC) {
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audio_codec_to_use = "aac";
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audio_codec = AudioCodec::AAC;
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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");
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}
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bool overclock = false;
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bool overclock = false;
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const char *overclock_str = args["-oc"].value();
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const char *overclock_str = args["-oc"].value();
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if(!overclock_str)
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if(!overclock_str)
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@ -1305,8 +1311,20 @@ int main(int argc, char **argv) {
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}
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}
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// TODO: Fix constant framerate not working properly on amd/intel because capture framerate gets locked to the same framerate as
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// TODO: Fix constant framerate not working properly on amd/intel because capture framerate gets locked to the same framerate as
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// game framerate, which doesn't work well when you need to encode multiple duplicate frames.
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// game framerate, which doesn't work well when you need to encode multiple duplicate frames (AMD/Intel is slow at encoding!).
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const FramerateMode framerate_mode = gpu_inf.vendor == GSR_GPU_VENDOR_NVIDIA ? FramerateMode::CONSTANT : FramerateMode::VARIABLE;
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FramerateMode framerate_mode;
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const char *framerate_mode_str = args["-fm"].value();
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if(!framerate_mode_str)
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framerate_mode_str = gpu_inf.vendor == GSR_GPU_VENDOR_NVIDIA ? "cfr" : "vfr";
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if(strcmp(framerate_mode_str, "cfr") == 0) {
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framerate_mode = FramerateMode::CONSTANT;
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} else if(strcmp(framerate_mode_str, "vfr") == 0) {
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framerate_mode = FramerateMode::VARIABLE;
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} else {
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fprintf(stderr, "Error: -fm should either be either 'cfr' or 'vfr', got: '%s'\n", framerate_mode_str);
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usage();
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}
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const char *screen_region = args["-s"].value();
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const char *screen_region = args["-s"].value();
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const char *window_str = args["-w"].value();
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const char *window_str = args["-w"].value();
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@ -1654,7 +1672,6 @@ int main(int argc, char **argv) {
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audio_track.audio_devices = std::move(audio_devices);
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audio_track.audio_devices = std::move(audio_devices);
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audio_track.graph = graph;
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audio_track.graph = graph;
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audio_track.sink = sink;
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audio_track.sink = sink;
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audio_track.pts = 0;
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audio_track.stream_index = audio_stream_index;
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audio_track.stream_index = audio_stream_index;
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audio_tracks.push_back(std::move(audio_track));
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audio_tracks.push_back(std::move(audio_track));
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++audio_stream_index;
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++audio_stream_index;
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@ -1795,16 +1812,11 @@ 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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fprintf(stderr, "Error: failed to add audio frame to filter\n");
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}
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}
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} else {
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} else {
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if(framerate_mode == FramerateMode::CONSTANT) {
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audio_track.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
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audio_track.frame->pts = audio_track.pts;
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const bool same_pts = audio_track.frame->pts == prev_pts;
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audio_track.pts += audio_track.frame->nb_samples;
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prev_pts = audio_track.frame->pts;
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} else {
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if(same_pts)
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audio_track.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
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continue;
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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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ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
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if(ret >= 0) {
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if(ret >= 0) {
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@ -1827,6 +1839,12 @@ int main(int argc, char **argv) {
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else
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else
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audio_track.frame->data[0] = (uint8_t*)sound_buffer;
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audio_track.frame->data[0] = (uint8_t*)sound_buffer;
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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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if(audio_track.graph) {
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if(audio_track.graph) {
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std::lock_guard<std::mutex> lock(audio_filter_mutex);
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std::lock_guard<std::mutex> lock(audio_filter_mutex);
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// TODO: av_buffersrc_add_frame
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// TODO: av_buffersrc_add_frame
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@ -1834,17 +1852,6 @@ 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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fprintf(stderr, "Error: failed to add audio frame to filter\n");
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}
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}
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} else {
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} else {
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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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ret = avcodec_send_frame(audio_track.codec_context, audio_track.frame);
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if(ret >= 0) {
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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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// TODO: Move to separate thread because this could write to network (for example when livestreaming)
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@ -1891,18 +1898,13 @@ int main(int argc, char **argv) {
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int err = 0;
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int err = 0;
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while ((err = av_buffersink_get_frame(audio_track.sink, aframe)) >= 0) {
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while ((err = av_buffersink_get_frame(audio_track.sink, aframe)) >= 0) {
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if(framerate_mode == FramerateMode::CONSTANT) {
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const double this_audio_frame_time = clock_get_monotonic_seconds();
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aframe->pts = audio_track.pts;
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aframe->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
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audio_track.pts += audio_track.codec_context->frame_size;
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const bool same_pts = aframe->pts == audio_prev_pts;
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} else {
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audio_prev_pts = aframe->pts;
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const double this_audio_frame_time = clock_get_monotonic_seconds();
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if(same_pts) {
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aframe->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE;
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av_frame_unref(aframe);
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const bool same_pts = aframe->pts == audio_prev_pts;
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continue;
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audio_prev_pts = aframe->pts;
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if(same_pts) {
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av_frame_unref(aframe);
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continue;
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}
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}
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}
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err = avcodec_send_frame(audio_track.codec_context, aframe);
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err = avcodec_send_frame(audio_track.codec_context, aframe);
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