diff options
Diffstat (limited to 'src/main.cpp')
| -rw-r--r-- | src/main.cpp | 36 |
1 files changed, 25 insertions, 11 deletions
diff --git a/src/main.cpp b/src/main.cpp index fcef91f..d64d7f4 100644 --- a/src/main.cpp +++ b/src/main.cpp @@ -1521,6 +1521,10 @@ static int init_filter_graph(AVCodecContext* audio_codec_context, AVFilterGraph* goto fail; } + /* Make sure the sink always outputs frames with the exact amount of samples the audio encoder wants, + otherwise the audio encoder rejects the frame and that piece of audio is lost */ + av_buffersink_set_frame_size(abuffersink_ctx, audio_codec_context->frame_size); + *graph = filter_graph; *sink = abuffersink_ctx; @@ -4196,9 +4200,13 @@ 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; - bool first_frame = true; int64_t num_received_frames = 0; + // The sound device is opened before the recording starts, so it can contain old audio from before the recording started. + // Discard it so the recording doesn't start with old audio. + if(audio_device.sound_device.handle) + sound_device_flush(&audio_device.sound_device); + while(running) { void *sound_buffer; int sound_buffer_size = -1; @@ -4246,15 +4254,13 @@ int main(int argc, char **argv) { // despite nvidia shadowplay and xbox game bar producing variable frame rate videos. // So we have to make sure we produce frames at the same relative rate as the video. if((num_missing_frames >= 1 && got_audio_data) || num_missing_frames >= 5 || !audio_device.sound_device.handle) { - // TODO: - //audio_track.frame->data[0] = empty_audio; - if(first_frame || num_missing_frames >= 5) { - if(needs_audio_conversion) - 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; - } - first_frame = false; + // Fill the missing frames with silence. Duplicating the previous audio frame to fill the gap instead + // sounds like a stutter and it's especially noticeable at the start of the recording when the audio device + // hasn't started to deliver audio at a stable rate yet, which repeats the first audio frame multiple times. + if(needs_audio_conversion) + 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; // 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); @@ -4283,12 +4289,20 @@ int main(int argc, char **argv) { if(!audio_device.sound_device.handle) { av_usleep(timeout_ms * 1000); } else if(got_audio_data) { + // The frame has to be made writable again if the frame was already sent to the audio filter above (when filling missing frames) + // because the audio filter only references the frame data instead of copying it. Without this the sent frames data would be + // overwritten with the audio data below, causing the audio to repeat instead of the missing frames being silent. + ret = av_frame_make_writable(audio_device.frame); + if (ret < 0) { + fprintf(stderr, "Failed to make audio frame writable\n"); + break; + } + // 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_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; std::lock_guard<std::mutex> lock(audio_filter_mutex); |