audio/filter: fix mul/delay scale and values

Before this commit, the af_instance->mul/delay values were in bytes.
Using bytes is confusing for non-interleaved audio, so switch mul to
samples, and delay to seconds. For delay, seconds are more intuitive
than bytes or samples, because it's used for the latency calculation.
We also might want to replace the delay mechanism with real PTS
tracking inside the filter chain some time in the future, and PTS
will also require time-adjustments to be done in seconds.

For most filters, we just remove the redundant mul=1 initialization.
(Setting this used to be required, but not anymore.)

1 files changed, 3 insertions, 3 deletions

diff --git a/audio/filter/af_lavfi.c b/audio/filter/af_lavfi.c
index e77110c067..24ff8c5985 100644
--- a/audio/filter/af_lavfi.c
+++ b/audio/filter/af_lavfi.c
@@ -197,7 +197,8 @@ static int control(struct af_instance *af, int cmd, void *arg)
 
         p->timebase_out = l_out->time_base;
 
-        af->mul = (double) (out->rate * out->nch) / (in->rate * in->nch);
+        // Blatantly incorrect; we don't know what the filters do.
+        af->mul = out->rate / (double)in->rate;
 
         return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
     }
@@ -270,7 +271,7 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
         // Need pts past the last output sample.
         out_time += r->samples / (double)r->rate;
 
-        af->delay = (in_time - out_time) * r->rate * r->sstride;
+        af->delay = in_time - out_time;
     }
 
     *data = *r;
@@ -286,7 +287,6 @@ static int af_open(struct af_instance *af)
     af->control = control;
     af->uninit = uninit;
     af->play = play;
-    af->mul = 1;
     struct priv *priv = af->priv;
     af->data = talloc_zero(priv, struct mp_audio),
     // Removing this requires fixing AVFrame.data vs. AVFrame.extended_data