audio: refactor how data is passed to AO

This replaces the two buffers (ao_chain.ao_buffer in the core, and
buffer_state.buffers in the AO) with a single queue. Instead of having a
byte based buffer, the queue is simply a list of audio frames, as output
by the decoder. This should make dataflow simpler and reduce copying.

It also attempts to simplify fill_audio_out_buffers(), the function I
always hated most, because it's full of subtle and buggy logic.

Unfortunately, I got assaulted by corner cases, dumb features (attempt
at seamless looping, really?), and other crap, so it got pretty
complicated again. fill_audio_out_buffers() is still full of subtle and
buggy logic. Maybe it got worse. On the other hand, maybe there really
is some progress. Who knows.

Originally, the data flow parts was meant to be in f_output_chain, but
due to tricky interactions with the playloop code, it's now in the dummy
filter in audio.c.

At least this improves the way the audio PTS is passed to the encoder in
encoding mode. Now it attempts to pass frames directly, along with the
pts, which should minimize timestamp problems. But to be honest, encoder
mode is one big kludge that shouldn't exist in this way.

This commit should be considered pre-alpha code. There are lots of bugs
still hiding.

1 files changed, 4 insertions, 1 deletions

diff --git a/player/video.c b/player/video.c
index 2ae581fcfc..387a366666 100644
--- a/player/video.c
+++ b/player/video.c
@@ -278,8 +278,11 @@ void reinit_video_chain_src(struct MPContext *mpctx, struct track *track)
     vo_set_paused(vo_c->vo, get_internal_paused(mpctx));
 
     // If we switch on video again, ensure audio position matches up.
-    if (mpctx->ao_chain)
+    if (mpctx->ao_chain && mpctx->ao_chain->ao) {
+        ao_reset(mpctx->ao_chain->ao);
+        mpctx->ao_chain->start_pts_known = false;
         mpctx->audio_status = STATUS_SYNCING;
+    }
 
     reset_video_state(mpctx);
     reset_subtitle_state(mpctx);