diff options
Diffstat (limited to 'src/main.cpp')
| -rw-r--r-- | src/main.cpp | 36 |
1 files changed, 34 insertions, 2 deletions
diff --git a/src/main.cpp b/src/main.cpp index bb54ca2..be0bba2 100644 --- a/src/main.cpp +++ b/src/main.cpp @@ -1713,6 +1713,12 @@ int main(int argc, char **argv) { usage(); } + if(audio_codec == AudioCodec::FLAC) { + fprintf(stderr, "Warning: flac audio codec has been temporary disabled, using opus audio codec instead\n"); + audio_codec_to_use = "opus"; + audio_codec = AudioCodec::OPUS; + } + bool overclock = false; const char *overclock_str = args["-oc"].value(); if(!overclock_str) @@ -2388,9 +2394,35 @@ 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); - // Move audio back by around 252 ms. This is just a shitty way to handle audio latency but pulseaudio latency calculation + // 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. - audio_device.frame->pts = audio_track.codec_context->frame_size * 12; + 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; + + int num_frames_to_delay = 12; + if(audio_codec == AudioCodec::FLAC) + num_frames_to_delay = 2; + + for(int i = 0; i < num_frames_to_delay; ++i) { + if(audio_track.graph) { + std::lock_guard<std::mutex> lock(audio_filter_mutex); + // TODO: av_buffersrc_add_frame + if(av_buffersrc_write_frame(audio_device.src_filter_ctx, audio_device.frame) < 0) { + fprintf(stderr, "Error: failed to add audio frame to filter\n"); + } + } else { + int 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) + receive_frames(audio_track.codec_context, audio_track.stream_index, audio_track.stream, audio_device.frame->pts, av_format_context, record_start_time, frame_data_queue, replay_buffer_size_secs, frames_erased, write_output_mutex, paused_time_offset); + } else { + fprintf(stderr, "Failed to encode audio!\n"); + } + } + audio_device.frame->pts += audio_track.codec_context->frame_size; + } while(running) { void *sound_buffer; |