diff options
Diffstat (limited to 'src')
| -rw-r--r-- | src/main.cpp | 81 |
1 files changed, 36 insertions, 45 deletions
diff --git a/src/main.cpp b/src/main.cpp index 9567102..06669aa 100644 --- a/src/main.cpp +++ b/src/main.cpp @@ -205,7 +205,7 @@ static AVCodecID audio_codec_get_id(AudioCodec audio_codec) { return AV_CODEC_ID_AAC; } -static AVSampleFormat audio_codec_get_sample_format(AudioCodec audio_codec, const AVCodec *codec) { +static AVSampleFormat audio_codec_get_sample_format(AudioCodec audio_codec, const AVCodec *codec, bool mix_audio) { switch(audio_codec) { case AudioCodec::AAC: { return AV_SAMPLE_FMT_FLTP; @@ -222,6 +222,10 @@ static AVSampleFormat audio_codec_get_sample_format(AudioCodec audio_codec, cons } } + // Amix only works with float audio + if(mix_audio) + supports_s16 = false; + if(!supports_s16 && !supports_flt) { fprintf(stderr, "Warning: opus audio codec is chosen but your ffmpeg version does not support s16/flt sample format and performance might be slightly worse. You can either rebuild ffmpeg with libopus instead of the built-in opus, use the flatpak version of gpu screen recorder or record with flac audio codec instead (-ac flac). Falling back to fltp audio sample format instead.\n"); } @@ -271,7 +275,7 @@ 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, bool mix_audio) { 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)); @@ -282,7 +286,7 @@ static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_code assert(codec->type == AVMEDIA_TYPE_AUDIO); codec_context->codec_id = codec->id; - codec_context->sample_fmt = audio_codec_get_sample_format(audio_codec, codec); + codec_context->sample_fmt = audio_codec_get_sample_format(audio_codec, codec, mix_audio); codec_context->bit_rate = audio_codec_get_get_bitrate(audio_codec); codec_context->sample_rate = 48000; if(audio_codec == AudioCodec::AAC) @@ -295,9 +299,10 @@ static AVCodecContext* create_audio_codec_context(int fps, AudioCodec audio_code #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; + codec_context->thread_count = 1; codec_context->flags |= AV_CODEC_FLAG_GLOBAL_HEADER; return codec_context; @@ -323,7 +328,7 @@ static AVCodecContext *create_video_codec_context(AVPixelFormat pix_fmt, codec_context->framerate.den = 1; codec_context->sample_aspect_ratio.num = 0; codec_context->sample_aspect_ratio.den = 0; - // High values reeduce file size but increases time it takes to seek + // High values reduce file size but increases time it takes to seek if(is_livestream) { codec_context->flags |= (AV_CODEC_FLAG_CLOSED_GOP | AV_CODEC_FLAG_LOW_DELAY); codec_context->flags2 |= AV_CODEC_FLAG2_FAST; @@ -974,6 +979,7 @@ struct AudioTrack { AVFilterGraph *graph = nullptr; AVFilterContext *sink = nullptr; int stream_index = 0; + int64_t pts = 0; }; static std::future<void> save_replay_thread; @@ -1449,7 +1455,7 @@ 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 = "aac"; + audio_codec_to_use = "opus"; if(strcmp(audio_codec_to_use, "aac") == 0) { audio_codec = AudioCodec::AAC; @@ -1462,12 +1468,6 @@ 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) @@ -1538,6 +1538,7 @@ int main(int argc, char **argv) { if(!audio_input_arg.values.empty()) audio_inputs = get_pulseaudio_inputs(); std::vector<MergedAudioInputs> requested_audio_inputs; + bool uses_amix = false; // Manually check if the audio inputs we give exist. This is only needed for pipewire, not pulseaudio. // Pipewire instead DEFAULTS TO THE DEFAULT AUDIO INPUT. THAT'S RETARDED. @@ -1547,6 +1548,9 @@ int main(int argc, char **argv) { continue; requested_audio_inputs.push_back({parse_audio_input_arg(audio_input)}); + if(requested_audio_inputs.back().audio_inputs.size() > 1) + uses_amix = true; + for(AudioInput &request_audio_input : requested_audio_inputs.back().audio_inputs) { bool match = false; for(const auto &existing_audio_input : audio_inputs) { @@ -1937,6 +1941,10 @@ int main(int argc, char **argv) { audio_codec_to_use = "aac"; audio_codec = AudioCodec::AAC; fprintf(stderr, "Warning: flac audio codec is only supported by .mp4 and .mkv files, falling back to aac instead\n"); + } else if(uses_amix) { + audio_codec_to_use = "opus"; + audio_codec = AudioCodec::OPUS; + fprintf(stderr, "Warning: flac audio codec is not supported when mixing audio sources, falling back to opus instead\n"); } break; } @@ -2063,7 +2071,7 @@ int main(int argc, char **argv) { framerate_mode_str = "cfr"; } - if(is_livestream) { + if(is_livestream && recording_saved_script) { fprintf(stderr, "Warning: live stream detected, -sc script is ignored\n"); recording_saved_script = nullptr; } @@ -2087,7 +2095,8 @@ 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); + const bool use_amix = merged_audio_inputs.audio_inputs.size() > 1; + AVCodecContext *audio_codec_context = create_audio_codec_context(fps, audio_codec, use_amix); AVStream *audio_stream = nullptr; if(replay_buffer_size_secs == -1) @@ -2108,7 +2117,6 @@ int main(int argc, char **argv) { std::vector<AVFilterContext*> src_filter_ctx; AVFilterGraph *graph = nullptr; AVFilterContext *sink = nullptr; - bool use_amix = merged_audio_inputs.audio_inputs.size() > 1; if(use_amix) { int err = init_filter_graph(audio_codec_context, &graph, &sink, src_filter_ctx, merged_audio_inputs.audio_inputs.size()); if(err < 0) { @@ -2133,15 +2141,16 @@ int main(int argc, char **argv) { if(audio_input.name.empty()) { audio_device.sound_device.handle = NULL; audio_device.sound_device.frames = 0; - audio_device.frame = NULL; } else { if(sound_device_get_by_name(&audio_device.sound_device, audio_input.name.c_str(), audio_input.description.c_str(), num_channels, audio_codec_context->frame_size, audio_codec_context_get_audio_format(audio_codec_context)) != 0) { fprintf(stderr, "Error: failed to get \"%s\" sound device\n", audio_input.name.c_str()); _exit(1); } - audio_device.frame = create_audio_frame(audio_codec_context); } + audio_device.frame = create_audio_frame(audio_codec_context); + audio_device.frame->pts = 0; + audio_devices.push_back(std::move(audio_device)); } @@ -2182,8 +2191,8 @@ int main(int argc, char **argv) { const double start_time_pts = clock_get_monotonic_seconds(); - double start_time = clock_get_monotonic_seconds(); // todo - target_fps to make first frame start immediately? - double frame_timer_start = start_time; + double start_time = clock_get_monotonic_seconds(); + double frame_timer_start = start_time - target_fps; // We want to capture the first frame immediately int fps_counter = 0; AVFrame *frame = av_frame_alloc(); @@ -2239,7 +2248,6 @@ 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; @@ -2259,7 +2267,7 @@ int main(int argc, char **argv) { } // 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_device.frame->nb_samples); + int64_t num_missing_frames = std::round((this_audio_frame_time - received_audio_time) / target_audio_hz / (int64_t)audio_track.codec_context->frame_size); if(got_audio_data) num_missing_frames = std::max((int64_t)0, num_missing_frames - 1); @@ -2278,7 +2286,7 @@ int main(int argc, char **argv) { //audio_track.frame->data[0] = empty_audio; received_audio_time = this_audio_frame_time; if(needs_audio_conversion) - swr_convert(swr, &audio_device.frame->data[0], audio_device.frame->nb_samples, (const uint8_t**)&empty_audio, audio_track.codec_context->frame_size); + swr_convert(swr, &audio_device.frame->data[0], audio_track.codec_context->frame_size, (const uint8_t**)&empty_audio, audio_track.codec_context->frame_size); else audio_device.frame->data[0] = empty_audio; @@ -2291,12 +2299,6 @@ int main(int argc, char **argv) { fprintf(stderr, "Error: failed to add audio frame to filter\n"); } } else { - audio_device.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE; - const bool same_pts = audio_device.frame->pts == prev_pts; - prev_pts = audio_device.frame->pts; - if(same_pts) - continue; - ret = avcodec_send_frame(audio_track.codec_context, audio_device.frame); if(ret >= 0) { // TODO: Move to separate thread because this could write to network (for example when livestreaming) @@ -2305,6 +2307,7 @@ int main(int argc, char **argv) { fprintf(stderr, "Failed to encode audio!\n"); } } + audio_device.frame->pts += audio_track.codec_context->frame_size; } } @@ -2314,16 +2317,10 @@ int main(int argc, char **argv) { if(got_audio_data) { // TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format? if(needs_audio_conversion) - swr_convert(swr, &audio_device.frame->data[0], audio_device.frame->nb_samples, (const uint8_t**)&sound_buffer, audio_track.codec_context->frame_size); + swr_convert(swr, &audio_device.frame->data[0], audio_track.codec_context->frame_size, (const uint8_t**)&sound_buffer, audio_track.codec_context->frame_size); else audio_device.frame->data[0] = (uint8_t*)sound_buffer; - audio_device.frame->pts = (this_audio_frame_time - record_start_time) * (double)AV_TIME_BASE; - const bool same_pts = audio_device.frame->pts == prev_pts; - prev_pts = audio_device.frame->pts; - if(same_pts) - continue; - if(audio_track.graph) { std::lock_guard<std::mutex> lock(audio_filter_mutex); // TODO: av_buffersrc_add_frame @@ -2339,6 +2336,8 @@ int main(int argc, char **argv) { fprintf(stderr, "Failed to encode audio!\n"); } } + + audio_device.frame->pts += audio_track.codec_context->frame_size; } } @@ -2356,7 +2355,6 @@ int main(int argc, char **argv) { int64_t video_pts_counter = 0; int64_t video_prev_pts = 0; - int64_t audio_prev_pts = 0; while(running) { double frame_start = clock_get_monotonic_seconds(); @@ -2377,15 +2375,7 @@ int main(int argc, char **argv) { int err = 0; while ((err = av_buffersink_get_frame(audio_track.sink, aframe)) >= 0) { - 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; - } - + aframe->pts = audio_track.pts; 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) @@ -2394,6 +2384,7 @@ int main(int argc, char **argv) { fprintf(stderr, "Failed to encode audio!\n"); } av_frame_unref(aframe); + audio_track.pts += audio_track.codec_context->frame_size; } } } |