audio/filter: prepare filter chain for non-interleaved audio

Based on earlier work by Stefano Pigozzi.

There are 2 changes:

1. Instead of mp_audio.audio, mp_audio.planes[0] must be used.

2. mp_audio.len used to contain the size of the audio in bytes. Now
   mp_audio.samples must be used. (Where 1 sample is the smallest unit
   of audio that covers all channels.)

Also, some filters need changes to reject non-interleaved formats
properly.

Nothing uses the non-interleaved features yet, but this is needed so
that things don't just break when doing so.

1 files changed, 2 insertions, 2 deletions

diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c
index 45143075a1..a6ebdc322e 100644
--- a/audio/filter/af_export.c
+++ b/audio/filter/af_export.c
@@ -213,9 +213,9 @@ static struct mp_audio* play( struct af_instance* af, struct mp_audio* data )
 {
   struct mp_audio*   	c   = data;	     // Current working data
   af_export_t* 	s   = af->setup;     // Setup for this instance
-  int16_t* 	a   = c->audio;	     // Incomming sound
+  int16_t* 	a   = c->planes[0];	     // Incomming sound
   int 		nch = c->nch;	     // Number of channels
-  int		len = c->len/c->bps; // Number of sample in data chunk
+  int		len = c->samples*c->nch; // Number of sample in data chunk
   int 		sz  = s->sz;         // buffer size (in samples)
   int 		flag = 0;	     // Set to 1 if buffer is filled