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More audio/video delay compensation

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dec05eba
2024-05-13 09:02:58 +02:00
parent 2cd031db48
commit e965da23c8
1 changed files with 11 additions and 11 deletions
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22
src/main.cpp
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@@ -1047,15 +1047,15 @@ static void run_recording_saved_script_async(const char *script_file, const char
static double audio_codec_get_desired_delay(AudioCodec audio_codec) {
switch(audio_codec) {
case AudioCodec::OPUS:
return 0.04;
return 0.03757;
case AudioCodec::AAC:
return 0.04 * 1.5;
return 0.03757 * 1.9375;
case AudioCodec::FLAC:
// TODO: Test
return 0.04;
return 0.03757;
}
assert(false);
return 0.04;
return 0.03757;
}
struct AudioDevice {
@@ -2337,7 +2337,7 @@ int main(int argc, char **argv) {
const double timeout_sec = 1000.0 / audio_fps / 1000.0;
const double audio_startup_time_seconds = force_no_audio_offset ? 0 : audio_codec_get_desired_delay(audio_codec);// * ((double)audio_codec_context->frame_size / 1024.0);
const int num_audio_frames_shift = std::round(audio_startup_time_seconds / timeout_sec);
const double num_audio_frames_shift = audio_startup_time_seconds / timeout_sec;
std::vector<AudioDevice> audio_devices;
for(size_t i = 0; i < merged_audio_inputs.audio_inputs.size(); ++i) {
@@ -2457,8 +2457,8 @@ int main(int argc, char **argv) {
const double audio_fps = (double)audio_track.codec_context->sample_rate / (double)audio_track.codec_context->frame_size;
const int64_t timeout_ms = std::round(1000.0 / audio_fps);
const double timeout_sec = 1000.0 / audio_fps / 1000.0;
const double audio_startup_time_seconds = force_no_audio_offset ? 0 : audio_codec_get_desired_delay(audio_codec);// * ((double)audio_track.codec_context->frame_size / 1024.0);
bool first_frame = true;
int64_t num_received_frames = 0;
while(running) {
void *sound_buffer;
@@ -2476,7 +2476,7 @@ int main(int argc, char **argv) {
//const double time_after_read_seconds = clock_get_monotonic_seconds();
//const double time_to_read_seconds = time_after_read_seconds - time_before_read_seconds;
//fprintf(stderr, "time to read: %f, %s, %f\n", time_to_read_seconds, got_audio_data ? "yes" : "no", timeout_sec);
const double this_audio_frame_time = (clock_get_monotonic_seconds() - audio_startup_time_seconds) - paused_time_offset;
const double this_audio_frame_time = clock_get_monotonic_seconds() - paused_time_offset;
if(paused) {
if(!audio_device.sound_device.handle)
@@ -2493,7 +2493,6 @@ int main(int argc, char **argv) {
// TODO: Is this |received_audio_time| really correct?
const int64_t num_expected_frames = std::round((this_audio_frame_time - record_start_time) / timeout_sec);
const int64_t num_received_frames = audio_device.frame->pts / (int64_t)audio_track.codec_context->frame_size;
int64_t num_missing_frames = std::max((int64_t)0LL, num_expected_frames - num_received_frames);
if(got_audio_data)
@@ -2518,6 +2517,7 @@ int main(int argc, char **argv) {
else
audio_device.frame->data[0] = empty_audio;
}
first_frame = false;
// TODO: Check if duplicate frame can be saved just by writing it with a different pts instead of sending it again
std::lock_guard<std::mutex> lock(audio_filter_mutex);
@@ -2538,9 +2538,8 @@ int main(int argc, char **argv) {
}
audio_device.frame->pts += audio_track.codec_context->frame_size;
num_received_frames++;
}
first_frame = false;
}
if(!audio_device.sound_device.handle)
@@ -2552,6 +2551,7 @@ int main(int argc, char **argv) {
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;
first_frame = false;
if(audio_track.graph) {
std::lock_guard<std::mutex> lock(audio_filter_mutex);
@@ -2570,7 +2570,7 @@ int main(int argc, char **argv) {
}
audio_device.frame->pts += audio_track.codec_context->frame_size;
first_frame = false;
num_received_frames++;
}
}