PapaTutuWawa/gpu-screen-recorder
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Read audio data from pulseaudio as it's available and buffer it. Fixes audio recording on pulseaudio (and some pipewire configs)

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dec05eba
2022-09-22 01:44:45 +02:00
parent 5ba4c05953
commit 0059724fdc
3 changed files with 132 additions and 38 deletions
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15
src/main.cpp
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@@ -1470,10 +1470,16 @@ int main(int argc, char **argv) {
int64_t pts = 0;
const double target_audio_hz = 1.0 / (double)audio_track.codec_context->sample_rate;
double received_audio_time = clock_get_monotonic_seconds();
while(running) {
void *sound_buffer;
int sound_buffer_size = sound_device_read_next_chunk(&audio_track.sound_device, &sound_buffer);
const bool got_audio_data = sound_buffer_size >= 0;
const double this_audio_frame_time = clock_get_monotonic_seconds();
if(got_audio_data)
received_audio_time = this_audio_frame_time;
int ret = av_frame_make_writable(audio_track.frame);
if (ret < 0) {
@@ -1481,15 +1487,14 @@ int main(int argc, char **argv) {
break;
}
const double this_audio_frame_time = clock_get_monotonic_seconds();
const int64_t expected_frames = std::round((this_audio_frame_time - start_time_pts) / target_audio_hz);
const int64_t num_missing_frames = std::max(0L, (expected_frames - pts) / audio_track.frame->nb_samples);
const int64_t num_missing_frames = std::round((this_audio_frame_time - received_audio_time) / target_audio_hz / (int64_t)audio_track.frame->nb_samples);
// Jesus is there a better way to do this? I JUST WANT TO KEEP VIDEO AND AUDIO SYNCED HOLY FUCK I WANT TO KILL MYSELF NOW.
// THIS PIECE OF SHIT WANTS EMPTY FRAMES OTHERWISE VIDEO PLAYS TOO FAST TO KEEP UP WITH AUDIO OR THE AUDIO PLAYS TOO EARLY.
// BUT WE CANT USE DELAYS TO GIVE DUMMY DATA BECAUSE PULSEAUDIO MIGHT GIVE AUDIO A BIG DELAYED!!!
if(num_missing_frames >= 5) {
if(num_missing_frames >= 5 || (num_missing_frames > 0 && got_audio_data)) {
// TODO:
//audio_track.frame->data[0] = empty_audio;
received_audio_time = this_audio_frame_time;
swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&empty_audio, audio_track.sound_device.frames);
// 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_missing_frames; ++i) {
@@ -1504,7 +1509,7 @@ int main(int argc, char **argv) {
}
}
if(sound_buffer_size >= 0) {
if(got_audio_data) {
// TODO: Instead of converting audio, get float audio from alsa. Or does alsa do conversion internally to get this format?
swr_convert(swr, &audio_track.frame->data[0], audio_track.frame->nb_samples, (const uint8_t**)&sound_buffer, audio_track.sound_device.frames);