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, 0 insertions, 2 deletions

diff --git a/audio/filter/af_channels.c b/audio/filter/af_channels.c
index 7e05a9e234..f77807a1af 100644
--- a/audio/filter/af_channels.c
+++ b/audio/filter/af_channels.c
@@ -168,7 +168,6 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
     mp_audio_set_format(af->data, ((struct mp_audio*)arg)->format);
-    af->mul          = (double)af->data->nch / ((struct mp_audio*)arg)->nch;
     mp_audio_force_interleaved_format(af->data);
     int r = af_test_output(af,(struct mp_audio*)arg);
     if (r != AF_OK)
@@ -255,7 +254,6 @@ static int af_open(struct af_instance* af){
   af->control=control;
   af->uninit=uninit;
   af->play=play;
-  af->mul=1;
   af->setup=calloc(1,sizeof(af_channels_t));
   if(af->setup == NULL)
     return AF_ERROR;