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, 5 insertions, 6 deletions

diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c
index 5b80bf0eec..01148dc6a6 100644
--- a/audio/filter/af_hrtf.c
+++ b/audio/filter/af_hrtf.c
@@ -367,7 +367,7 @@ static void uninit(struct af_instance *af)
 	free(af->setup);
     }
     if(af->data)
-	free(af->data->audio);
+	free(af->data->planes[0]);
     free(af->data);
 }
 
@@ -385,9 +385,9 @@ damped (without any real 3D acoustical image, however).
 static struct mp_audio* play(struct af_instance *af, struct mp_audio *data)
 {
     af_hrtf_t *s = af->setup;
-    short *in = data->audio; // Input audio data
+    short *in = data->planes[0]; // Input audio data
     short *out = NULL; // Output audio data
-    short *end = in + data->len / sizeof(short); // Loop end
+    short *end = in + data->samples * data->nch; // Loop end
     float common, left, right, diff, left_b, right_b;
     const int dblen = s->dlbuflen, hlen = s->hrflen, blen = s->basslen;
 
@@ -425,7 +425,7 @@ static struct mp_audio* play(struct af_instance *af, struct mp_audio *data)
 		 "channel\n");
     }
 
-    out = af->data->audio;
+    out = af->data->planes[0];
 
     /* MPlayer's 5 channel layout (notation for the variable):
      *
@@ -565,8 +565,7 @@ static struct mp_audio* play(struct af_instance *af, struct mp_audio *data)
     }
 
     /* Set output data */
-    data->audio = af->data->audio;
-    data->len   = data->len / data->nch * 2;
+    data->planes[0] = af->data->planes[0];
     mp_audio_set_num_channels(data, 2);
 
     return data;