PapaTutuWawa/gpu-screen-recorder
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

Fix opus and flac audio sources, fix crash when live streaming without an audio source

Browse Source
This commit is contained in:
dec05eba 2023-12-03 00:59:07 +01:00
parent 1260598e9e
commit 852882bae3
2 changed files with 37 additions and 45 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
kms/server/kms_server.c
View File

@ -143,6 +143,7 @@ static uint32_t plane_get_properties(int drmfd, uint32_t plane_id, bool *is_curs
if(!props)
return false;
// TODO: Dont do this every frame
for(uint32_t i = 0; i < props->count_props; ++i) {
drmModePropertyPtr prop = drmModeGetProperty(drmfd, props->props[i]);
if(!prop)

81
src/main.cpp
View File

@ -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;
}
}
}