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, 6 insertions, 6 deletions

diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index e381a12a3c..ef7993c83a 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -270,20 +270,20 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
 
     // Filter
     struct mp_audio filter_input = {
-        .audio = sh->a_buffer,
-        .len = len,
+        .planes = {sh->a_buffer},
         .rate = sh->samplerate,
     };
     mp_audio_set_format(&filter_input, sh->sample_format);
     mp_audio_set_channels(&filter_input, &sh->channels);
+    filter_input.samples = len / filter_input.sstride;
 
     struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
     if (!filter_output)
         return -1;
-    set_min_out_buffer_size(outbuf, outbuf->len + filter_output->len);
-    memcpy(outbuf->start + outbuf->len, filter_output->audio,
-           filter_output->len);
-    outbuf->len += filter_output->len;
+    int outlen = filter_output->samples * filter_output->sstride;
+    set_min_out_buffer_size(outbuf, outbuf->len + outlen);
+    memcpy(outbuf->start + outbuf->len, filter_output->planes[0], outlen);
+    outbuf->len += outlen;
 
     // remove processed data from decoder buffer:
     sh->a_buffer_len -= len;