}
}
video_frame->received_timestamp = next_frame_start;
- audio_frame->received_timestamp = next_frame_start;
+
+ // The easiest way to get all the rate conversions etc. right is to move the
+ // audio PTS into the video PTS timebase and go from there. (We'll get some
+ // rounding issues, but they should not be a big problem.)
+ int64_t audio_pts_as_video_pts = av_rescale_q(audio_pts, audio_timebase, video_timebase);
+ audio_frame->received_timestamp = compute_frame_start(audio_pts_as_video_pts, pts_origin, video_timebase, start, rate);
+
+ if (audio_frame->len != 0) {
+ // The received timestamps in Nageru are measured after we've just received the frame.
+ // However, pts (especially audio pts) is at the _beginning_ of the frame.
+ // If we have locked audio, the distinction doesn't really matter, as pts is
+ // on a relative scale and a fixed offset is fine. But if we don't, we will have
+ // a different number of samples each time, which will cause huge audio jitter
+ // and throw off the resampler.
+ //
+ // In a sense, we should have compensated by adding the frame and audio lengths
+ // to video_frame->received_timestamp and audio_frame->received_timestamp respectively,
+ // but that would mean extra waiting in sleep_until(). All we need is that they
+ // are correct relative to each other, though (and to the other frames we send),
+ // so just align the end of the audio frame, and we're fine.
+ size_t num_samples = (audio_frame->len * 8) / audio_format.bits_per_sample / audio_format.num_channels;
+ double offset = double(num_samples) / OUTPUT_FREQUENCY -
+ double(video_format.frame_rate_den) / video_format.frame_rate_nom;
+ audio_frame->received_timestamp += duration_cast<steady_clock::duration>(duration<double>(offset));
+ }
+
bool finished_wakeup = producer_thread_should_quit.sleep_until(next_frame_start);
if (finished_wakeup) {
if (audio_frame->len > 0) {
AVFrameWithDeleter video_avframe = av_frame_alloc_unique();
bool eof = false;
*audio_pts = -1;
+ bool has_audio = false;
do {
AVPacket pkt;
unique_ptr<AVPacket, decltype(av_packet_unref)*> pkt_cleanup(
return AVFrameWithDeleter(nullptr);
}
} else if (pkt.stream_index == audio_stream_index) {
- if (*audio_pts == -1) {
- *audio_pts = pkt.pts;
- }
+ has_audio = true;
if (avcodec_send_packet(audio_codec_ctx, &pkt) < 0) {
fprintf(stderr, "%s: Cannot send packet to audio codec.\n", pathname.c_str());
*error = true;
}
// Decode audio, if any.
- if (*audio_pts != -1) {
+ if (has_audio) {
for ( ;; ) {
int err = avcodec_receive_frame(audio_codec_ctx, audio_avframe.get());
if (err == 0) {
+ if (*audio_pts == -1) {
+ *audio_pts = audio_avframe->pts;
+ }
convert_audio(audio_avframe.get(), audio_frame, audio_format);
} else if (err == AVERROR(EAGAIN)) {
break;