af_scaletempo2: fix audio-video de-sync caused by speed changes

Fixes #12028

There was an additional issue that audio was always delayed by half the
configured search-interval. This was caused by the `out` buffer length
not being included in the delay calculation.

Notes:
- Every WSOLA iteration advances the input buffer by _some amount_, and
  produces data in the output buffer always of size `ola_hop_size`.
- `mp_scaletempo2_fill_buffer` is always called with `ola_hop_size`
- Thus, the rendered frames are always cleared immediately after
  processing, and `num_complete_frames` is 0 in the delay calculation.
- The factors contributing to delay are:
  - the pending samples in the input buffer according to the search
    block position, and
  - the pending rendered samples in the output buffer (always empty in
    practice).

The frame_delay code looked like that of the rubberband filter, which
might not work for scaletempo2. Sometimes a different amount of input
audio was consumed by scaletempo2 than expected. It may have been caused
by speed changes being a more dynamic process in scaletempo2. This can
be seen by where `playback_rate` is used in `run_one_wsola_iteration`:
`playback_rate` is only referenced after the iteration, when updating
the time and removing old data from buffers.

In scaletempo2, the playback speed is applied by changing the amount the
search block is moved. That apparently averages out correctly at
constant playback speed, but when the speed changes, the error in this
assumption probably spikes. This error accumulated across all speed
changes because of the persistent `frame_delay` value.

With the removal of the persistent `frame_delay`, there should be no way
for the audio to drift off. By deriving the delay from filter buffer
positions, and the buffers are filled only as much as needed, the delay
always stays within buffer bounds.

1 files changed, 4 insertions, 8 deletions

diff --git a/audio/filter/af_scaletempo2.c b/audio/filter/af_scaletempo2.c
index 1a822ecd50..4e9d5f9464 100644
--- a/audio/filter/af_scaletempo2.c
+++ b/audio/filter/af_scaletempo2.c
@@ -15,7 +15,6 @@ struct priv {
     bool sent_final;
     struct mp_aframe *pending;
     bool initialized;
-    double frame_delay;
     float speed;
 };
 
@@ -67,7 +66,6 @@ static void process(struct mp_filter *f)
             uint8_t **planes = mp_aframe_get_data_ro(p->pending);
             int read = mp_scaletempo2_fill_input_buffer(&p->data,
                 planes, frame_size, final);
-            p->frame_delay += read;
             mp_aframe_skip_samples(p->pending, read);
         }
         p->sent_final |= final;
@@ -109,11 +107,11 @@ static void process(struct mp_filter *f)
             (float**)planes, out_samples, p->speed);
 
         double pts = mp_aframe_get_pts(p->pending);
-        p->frame_delay -= out_samples * p->speed;
-
         if (pts != MP_NOPTS_VALUE) {
-            double delay = p->frame_delay / mp_aframe_get_effective_rate(out);
-            mp_aframe_set_pts(out, pts - delay);
+            double frame_delay = p->data.input_buffer_frames - p->data.search_block_index
+                                 + p->data.num_complete_frames * p->speed
+                                 + out_samples * p->speed;
+            mp_aframe_set_pts(out, pts - frame_delay / mp_aframe_get_effective_rate(out));
         }
 
         mp_aframe_set_size(out, out_samples);
@@ -137,7 +135,6 @@ static bool init_scaletempo2(struct mp_filter *f)
     mp_aframe_reset(p->cur_format);
     p->initialized = true;
     p->sent_final = false;
-    p->frame_delay = 0;
     mp_aframe_config_copy(p->cur_format, p->pending);
 
     mp_scaletempo2_init(&p->data, mp_aframe_get_channels(p->pending),
@@ -163,7 +160,6 @@ static void reset(struct mp_filter *f)
 {
     struct priv *p = f->priv;
     mp_scaletempo2_reset(&p->data);
-    p->frame_delay = 0;
     p->initialized = false;
     TA_FREEP(&p->pending);
 }