diff options
Diffstat (limited to 'src')
| -rw-r--r-- | src/main.cpp | 7 |
1 files changed, 6 insertions, 1 deletions
diff --git a/src/main.cpp b/src/main.cpp index 2365f56..faab93b 100644 --- a/src/main.cpp +++ b/src/main.cpp @@ -241,7 +241,9 @@ static AVSampleFormat audio_codec_get_sample_format(AudioCodec audio_codec, cons 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"); + 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.\n"); + fprintf(stderr, " You can either rebuild ffmpeg with libopus instead of the built-in opus, use the flatpak version of gpu screen recorder or record with aac audio codec instead (-ac aac).\n"); + fprintf(stderr, " Falling back to fltp audio sample format instead.\n"); } if(supports_s16) @@ -2399,6 +2401,8 @@ int main(int argc, char **argv) { 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); + // Remove this for now, it doesn't work well for everybody. The timing is different depending on system + #if 0 // Move audio forward by around 252 ms (for opus/aac), or 42ms for flac. This is just a shitty way to handle audio latency but pulseaudio latency calculation // returns much lower value which isn't helpful. if(needs_audio_conversion) @@ -2428,6 +2432,7 @@ int main(int argc, char **argv) { } audio_device.frame->pts += audio_track.codec_context->frame_size; } + #endif while(running) { void *sound_buffer; |