From f7a427676c0fe3c12509e3d9a243301f93626b0a Mon Sep 17 00:00:00 2001
From: wm4 <wm4@nowhere>
Date: Fri, 5 Apr 2013 19:47:51 +0200
Subject: audio: add some setters for mp_audio, and require filters to use them

mp_audio has some redundant fields. Setters like mp_audio_set_format()
initialize these properly.

Also move the mp_audio struct to a the file audio.c.

We can remove a mysterious line of code from af.c:

    in.format |= af_bits2fmt(in.bps * 8);

I'm not sure if this was ever actually needed, or if it was some kind of
"make it work" quick-fix that works against the way things were supposed
to work. All filters etc. now set the format correctly, so if there ever
was a need for this code, it's definitely gone.
---
 audio/audio.c                  | 38 ++++++++++++++++++++++++++++++++++++++
 audio/audio.h                  | 38 ++++++++++++++++++++++++++++++++++++++
 audio/decode/dec_audio.c       | 20 +++++++++-----------
 audio/filter/af.c              | 20 ++------------------
 audio/filter/af.h              | 21 +--------------------
 audio/filter/af_bs2b.c         |  8 +++-----
 audio/filter/af_center.c       |  5 ++---
 audio/filter/af_channels.c     |  7 +++----
 audio/filter/af_delay.c        |  5 +----
 audio/filter/af_drc.c          | 13 ++++---------
 audio/filter/af_equalizer.c    |  6 ++----
 audio/filter/af_export.c       |  6 ++----
 audio/filter/af_extrastereo.c  | 11 ++++-------
 audio/filter/af_format.c       | 19 +++++++------------
 audio/filter/af_hrtf.c         | 11 +++++------
 audio/filter/af_karaoke.c      |  6 ++----
 audio/filter/af_ladspa.c       |  6 ++----
 audio/filter/af_lavcac3enc.c   | 16 +++++++---------
 audio/filter/af_lavrresample.c | 16 ++++++----------
 audio/filter/af_pan.c          | 10 ++++------
 audio/filter/af_scaletempo.c   | 17 +++++------------
 audio/filter/af_sinesuppress.c |  7 +++----
 audio/filter/af_sub.c          |  5 ++---
 audio/filter/af_surround.c     | 12 +++++-------
 audio/filter/af_sweep.c        |  8 +++-----
 audio/filter/af_volume.c       |  9 +++------
 audio/format.c                 |  2 ++
 audio/out/ao.h                 |  2 +-
 28 files changed, 166 insertions(+), 178 deletions(-)
 create mode 100644 audio/audio.c
 create mode 100644 audio/audio.h

(limited to 'audio')

diff --git a/audio/audio.c b/audio/audio.c
new file mode 100644
index 0000000000..04ebe390b8
--- /dev/null
+++ b/audio/audio.c
@@ -0,0 +1,38 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <assert.h>
+
+#include "audio.h"
+
+void mp_audio_set_format(struct mp_audio *mpa, int format)
+{
+    mpa->format = format;
+    mpa->bps = af_fmt2bits(format) / 8;
+}
+
+void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels)
+{
+    mpa->nch = num_channels;
+}
+
+void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src)
+{
+    mp_audio_set_format(dst, src->format);
+    mp_audio_set_num_channels(dst, src->nch);
+    dst->rate = src->rate;
+}
diff --git a/audio/audio.h b/audio/audio.h
new file mode 100644
index 0000000000..f39069cf73
--- /dev/null
+++ b/audio/audio.h
@@ -0,0 +1,38 @@
+/*
+ * This file is part of mpv.
+ *
+ * mpv is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mpv is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mpv.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef MP_AUDIO_H
+#define MP_AUDIO_H
+
+#include "format.h"
+
+// Audio data chunk
+struct mp_audio {
+    void *audio; // data buffer
+    int len;    // buffer length (in bytes)
+    int rate;   // sample rate
+    int nch;    // number of channels, use mp_audio_set_channels() to set
+    int format; // format (AF_FORMAT_...), use mp_audio_set_format() to set
+    // Redundant fields, for convenience
+    int bps;    // bytes per sample (redundant with format)
+};
+
+void mp_audio_set_format(struct mp_audio *mpa, int format);
+void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels);
+void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src);
+
+#endif
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
index c7ae3155ed..6c8804eec9 100644
--- a/audio/decode/dec_audio.c
+++ b/audio/decode/dec_audio.c
@@ -205,16 +205,14 @@ int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
     if (!afs)
         afs = af_new(sh_audio->opts);
     // input format: same as codec's output format:
-    afs->input.rate   = in_samplerate;
-    afs->input.nch    = sh_audio->channels;
-    afs->input.format = sh_audio->sample_format;
-    af_fix_parameters(&(afs->input));
+    afs->input.rate = in_samplerate;
+    mp_audio_set_num_channels(&afs->input, sh_audio->channels);
+    mp_audio_set_format(&afs->input, sh_audio->sample_format);
 
     // output format: same as ao driver's input format (if missing, fallback to input)
-    afs->output.rate   = *out_samplerate;
-    afs->output.nch    = *out_channels;
-    afs->output.format = *out_format;
-    af_fix_parameters(&(afs->output));
+    afs->output.rate = *out_samplerate;
+    mp_audio_set_num_channels(&afs->output, *out_channels);
+    mp_audio_set_format(&afs->output, *out_format);
 
     // filter config:
     memcpy(&afs->cfg, &af_cfg, sizeof(struct af_cfg));
@@ -284,10 +282,10 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
         .audio = sh->a_buffer,
         .len = len,
         .rate = sh->samplerate,
-        .nch = sh->channels,
-        .format = sh->sample_format
     };
-    af_fix_parameters(&filter_input);
+    mp_audio_set_format(&filter_input, sh->sample_format);
+    mp_audio_set_num_channels(&filter_input, sh->channels);
+
     struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
     if (!filter_output)
         return -1;
diff --git a/audio/filter/af.c b/audio/filter/af.c
index 9f534ad2fb..4d2e5e3ac3 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -84,17 +84,14 @@ static struct af_info* filter_list[] = {
 static bool af_config_equals(struct mp_audio *a, struct mp_audio *b)
 {
     return a->format == b->format
-        && a->bps    == b->bps
         && a->nch    == b->nch
         && a->rate   == b->rate;
 }
 
 static void af_copy_unset_fields(struct mp_audio *dst, struct mp_audio *src)
 {
-    if (dst->format == AF_FORMAT_UNKNOWN) {
-        dst->format = src->format;
-        dst->bps = src->bps;
-    }
+    if (dst->format == AF_FORMAT_UNKNOWN)
+        mp_audio_set_format(dst, src->format);
     if (dst->nch == 0)
         dst->nch = src->nch;
     if (dst->rate == 0)
@@ -459,9 +456,6 @@ int af_reinit(struct af_stream *s)
             af = af->next;
             break;
         case AF_FALSE: { // Configuration filter is needed
-            // Set output bits per sample (unknown why this would be needed)
-            in.format |= af_bits2fmt(in.bps * 8);
-
             if (af_fix_channels(s, &af, in) == AF_OK) {
                 retry++;
                 continue;
@@ -743,13 +737,3 @@ void af_help(void)
         i++;
     }
 }
-
-void af_fix_parameters(struct mp_audio *data)
-{
-    if (data->nch < 0 || data->nch > AF_NCH) {
-        mp_msg(MSGT_AFILTER, MSGL_ERR,
-               "Invalid number of channels %i, assuming 2.\n", data->nch);
-        data->nch = 2;
-    }
-    data->bps = af_fmt2bits(data->format) / 8;
-}
diff --git a/audio/filter/af.h b/audio/filter/af.h
index 96e5c66607..4ccc70792f 100644
--- a/audio/filter/af.h
+++ b/audio/filter/af.h
@@ -27,6 +27,7 @@
 #include "core/options.h"
 #include "audio/format.h"
 #include "audio/chmap.h"
+#include "audio/audio.h"
 #include "control.h"
 #include "core/mp_msg.h"
 
@@ -35,17 +36,6 @@ struct af_instance;
 // Number of channels
 #define AF_NCH MP_NUM_CHANNELS
 
-// Audio data chunk
-struct mp_audio {
-    void *audio; // data buffer
-    int len;    // buffer length
-    int rate;   // sample rate
-    int nch;    // number of channels
-    int format; // format
-    int bps;    // bytes per sample
-};
-
-
 // Flags used for defining the behavior of an audio filter
 #define AF_FLAGS_REENTRANT      0x00000000
 #define AF_FLAGS_NOT_REENTRANT  0x00000001
@@ -295,15 +285,6 @@ float af_softclip(float a);
 /** Print a list of all available audio filters */
 void af_help(void);
 
-/**
- * \brief fill the missing parameters in the struct mp_audio structure
- * \param data structure to fill
- * \ingroup af_filter
- *
- * Currently only sets bps based on format
- */
-void af_fix_parameters(struct mp_audio *data);
-
 /** Memory reallocation macro: if a local buffer is used (i.e. if the
    filter doesn't operate on the incoming buffer this macro must be
    called to ensure the buffer is big enough.
diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c
index aebcc3b201..f8003a70a3 100644
--- a/audio/filter/af_bs2b.c
+++ b/audio/filter/af_bs2b.c
@@ -105,9 +105,8 @@ static int control(struct af_instance *af, int cmd, void *arg)
 
         format           = ((struct mp_audio*)arg)->format;
         af->data->rate   = ((struct mp_audio*)arg)->rate;
-        af->data->nch    = 2;     // bs2b is useful only for 2ch audio
-        af->data->bps    = ((struct mp_audio*)arg)->bps;
-        af->data->format = format;
+        mp_audio_set_num_channels(af->data, 2);     // bs2b is useful only for 2ch audio
+        mp_audio_set_format(af->data, format);
 
         /* check for formats supported by libbs2b
            and assign corresponding handlers */
@@ -162,8 +161,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
                 break;
             default:
                 af->play = play_f;
-                af->data->format = AF_FORMAT_FLOAT_NE;
-                af->data->bps = 4;
+                mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
                 break;
         }
 
diff --git a/audio/filter/af_center.c b/audio/filter/af_center.c
index aa9aae8514..f92cefd335 100644
--- a/audio/filter/af_center.c
+++ b/audio/filter/af_center.c
@@ -48,9 +48,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!arg) return AF_ERROR;
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = max(s->ch+1,((struct mp_audio*)arg)->nch);
-    af->data->format = AF_FORMAT_FLOAT_NE;
-    af->data->bps    = 4;
+    mp_audio_set_num_channels(af->data, max(s->ch+1,((struct mp_audio*)arg)->nch));
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
     return af_test_output(af,(struct mp_audio*)arg);
   }
diff --git a/audio/filter/af_channels.c b/audio/filter/af_channels.c
index ac396a51cd..f6369936d6 100644
--- a/audio/filter/af_channels.c
+++ b/audio/filter/af_channels.c
@@ -164,8 +164,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     }
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->format = ((struct mp_audio*)arg)->format;
-    af->data->bps    = ((struct mp_audio*)arg)->bps;
+    mp_audio_set_format(af->data, ((struct mp_audio*)arg)->format);
     af->mul          = (double)af->data->nch / ((struct mp_audio*)arg)->nch;
     return check_routes(s,((struct mp_audio*)arg)->nch,af->data->nch);
   case AF_CONTROL_COMMAND_LINE:{
@@ -208,7 +207,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
       return AF_ERROR;
     }
 
-    af->data->nch=((int*)arg)[0];
+    mp_audio_set_num_channels(af->data, ((int*)arg)[0]);
     if(!s->router)
       mp_msg(MSGT_AFILTER, MSGL_V, "[channels] Changing number of channels"
 	     " to %i\n",af->data->nch);
@@ -248,7 +247,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
   // Set output data
   c->audio = l->audio;
   c->len   = c->len / c->nch * l->nch;
-  c->nch   = l->nch;
+  mp_audio_set_num_channels(c, l->nch);
 
   return c;
 }
diff --git a/audio/filter/af_delay.c b/audio/filter/af_delay.c
index ce8d71980b..eb3e9e19cb 100644
--- a/audio/filter/af_delay.c
+++ b/audio/filter/af_delay.c
@@ -52,10 +52,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     for(i=0;i<af->data->nch;i++)
       free(s->q[i]);
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
-    af->data->format = ((struct mp_audio*)arg)->format;
-    af->data->bps    = ((struct mp_audio*)arg)->bps;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
 
     // Allocate new delay queues
     for(i=0;i<af->data->nch;i++){
diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c
index ba1e10c0a7..e293f7f616 100644
--- a/audio/filter/af_drc.c
+++ b/audio/filter/af_drc.c
@@ -87,15 +87,10 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Sanity check
     if(!arg) return AF_ERROR;
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
-
-    if(((struct mp_audio*)arg)->format == (AF_FORMAT_S16_NE)){
-      af->data->format = AF_FORMAT_S16_NE;
-      af->data->bps    = 2;
-    }else{
-      af->data->format = AF_FORMAT_FLOAT_NE;
-      af->data->bps    = 4;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+
+    if(((struct mp_audio*)arg)->format != (AF_FORMAT_S16_NE)){
+      mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
     }
     return af_test_output(af,(struct mp_audio*)arg);
   case AF_CONTROL_COMMAND_LINE:{
diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c
index a05b2df399..6441b9b116 100644
--- a/audio/filter/af_equalizer.c
+++ b/audio/filter/af_equalizer.c
@@ -96,10 +96,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Sanity check
     if(!arg) return AF_ERROR;
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
-    af->data->format = AF_FORMAT_FLOAT_NE;
-    af->data->bps    = 4;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
     // Calculate number of active filters
     s->K=KM;
diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c
index 5e3a1869ee..ea0aa938ca 100644
--- a/audio/filter/af_export.c
+++ b/audio/filter/af_export.c
@@ -85,10 +85,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
       close(s->fd);
 
     // Accept only int16_t as input format (which sucks)
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
-    af->data->format = AF_FORMAT_S16_NE;
-    af->data->bps    = 2;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
 
     // If buffer length isn't set, set it to the default value
     if(s->sz == 0)
diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c
index d9120fb8d9..ed1fd27898 100644
--- a/audio/filter/af_extrastereo.c
+++ b/audio/filter/af_extrastereo.c
@@ -47,17 +47,14 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Sanity check
     if(!arg) return AF_ERROR;
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = 2;
-    if (((struct mp_audio*)arg)->format == AF_FORMAT_FLOAT_NE)
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_num_channels(af->data, 2);
+    if (af->data->format == AF_FORMAT_FLOAT_NE)
     {
-	af->data->format = AF_FORMAT_FLOAT_NE;
-	af->data->bps = 4;
 	af->play = play_float;
     }// else
     {
-	af->data->format = AF_FORMAT_S16_NE;
-	af->data->bps = 2;
+        mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
 	af->play = play_s16;
     }
 
diff --git a/audio/filter/af_format.c b/audio/filter/af_format.c
index 720cff0bf0..5988d25118 100644
--- a/audio/filter/af_format.c
+++ b/audio/filter/af_format.c
@@ -94,8 +94,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     int supported_ac3 = 0;
 
     // Make sure this filter isn't redundant
-    if(af->data->format == data->format &&
-       af->data->bps == data->bps)
+    if(af->data->format == data->format)
       return AF_DETACH;
 
     // A bit complex because we can convert AC3
@@ -121,7 +120,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
 	   buf1, buf2);
 
     af->data->rate = data->rate;
-    af->data->nch  = data->nch;
+    mp_audio_set_num_channels(af->data, data->nch);
     af->mul        = (double)af->data->bps / data->bps;
 
     af->play = play; // set default
@@ -164,8 +163,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!AF_FORMAT_IS_AC3(*(int*)arg) && AF_OK != check_format(*(int*)arg))
       return AF_ERROR;
 
-    af->data->format = *(int*)arg;
-    af->data->bps = af_fmt2bits(af->data->format)/8;
+    mp_audio_set_format(af->data, *(int*)arg);
 
     return AF_OK;
   }
@@ -194,7 +192,7 @@ static struct mp_audio* play_swapendian(struct af_instance* af, struct mp_audio*
   endian(c->audio,l->audio,len,c->bps);
 
   c->audio = l->audio;
-  c->format = l->format;
+  mp_audio_set_format(c, l->format);
 
   return c;
 }
@@ -211,9 +209,8 @@ static struct mp_audio* play_float_s16(struct af_instance* af, struct mp_audio*
   float2int(c->audio, l->audio, len, 2);
 
   c->audio = l->audio;
+  mp_audio_set_format(c, l->format);
   c->len = len*2;
-  c->bps = 2;
-  c->format = l->format;
 
   return c;
 }
@@ -230,9 +227,8 @@ static struct mp_audio* play_s16_float(struct af_instance* af, struct mp_audio*
   int2float(c->audio, l->audio, len, 2);
 
   c->audio = l->audio;
+  mp_audio_set_format(c, l->format);
   c->len = len*4;
-  c->bps = 4;
-  c->format = l->format;
 
   return c;
 }
@@ -284,9 +280,8 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
 
   // Set output data
   c->audio  = l->audio;
+  mp_audio_set_format(c, l->format);
   c->len    = len*l->bps;
-  c->bps    = l->bps;
-  c->format = l->format;
   return c;
 }
 
diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c
index 72e2ec7c66..1e3244e2eb 100644
--- a/audio/filter/af_hrtf.c
+++ b/audio/filter/af_hrtf.c
@@ -301,7 +301,7 @@ static int control(struct af_instance *af, int cmd, void* arg)
 		   af->data->rate);
 	    return AF_ERROR;
 	}
-	af->data->nch    = ((struct mp_audio*)arg)->nch;
+	mp_audio_set_num_channels(af->data, ((struct mp_audio*)arg)->nch);
 	    if(af->data->nch == 2) {
  	       /* 2 channel input */
  	       if(s->decode_mode != HRTF_MIX_MATRIX2CH) {
@@ -310,13 +310,12 @@ static int control(struct af_instance *af, int cmd, void* arg)
 	       }
 	    }
 	    else if (af->data->nch < 5)
-	      af->data->nch = 5;
-	af->data->format = AF_FORMAT_S16_NE;
-	af->data->bps    = 2;
+	      mp_audio_set_num_channels(af->data, 5);
+        mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
 	test_output_res = af_test_output(af, (struct mp_audio*)arg);
 	af->mul = 2.0 / af->data->nch;
 	// after testing input set the real output format
-	af->data->nch = 2;
+        mp_audio_set_num_channels(af->data, 2);
 	s->print_flag = 1;
 	return test_output_res;
     case AF_CONTROL_COMMAND_LINE:
@@ -574,7 +573,7 @@ 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->nch   = 2;
+    mp_audio_set_num_channels(data, 2);
 
     return data;
 }
diff --git a/audio/filter/af_karaoke.c b/audio/filter/af_karaoke.c
index 965eb8f40d..faed389625 100644
--- a/audio/filter/af_karaoke.c
+++ b/audio/filter/af_karaoke.c
@@ -34,10 +34,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
 {
 	switch(cmd){
 		case AF_CONTROL_REINIT:
-		af->data->rate	= ((struct mp_audio*)arg)->rate;
-		af->data->nch	= ((struct mp_audio*)arg)->nch;
-		af->data->format= AF_FORMAT_FLOAT_NE;
-		af->data->bps	= 4;
+                mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+                mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 		return af_test_output(af,(struct mp_audio*)arg);
 	}
 	return AF_UNKNOWN;
diff --git a/audio/filter/af_ladspa.c b/audio/filter/af_ladspa.c
index c1b3f24360..f7ec0d278f 100644
--- a/audio/filter/af_ladspa.c
+++ b/audio/filter/af_ladspa.c
@@ -498,10 +498,8 @@ static int control(struct af_instance *af, int cmd, void *arg) {
 
         /* accept FLOAT, let af_format do conversion */
 
-        af->data->rate   = ((struct mp_audio*)arg)->rate;
-        af->data->nch    = ((struct mp_audio*)arg)->nch;
-        af->data->format = AF_FORMAT_FLOAT_NE;
-        af->data->bps    = 4;
+        mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+        mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
         /* arg->len is not set here yet, so init of buffers and connecting the
          * filter, has to be done in play() :-/
diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c
index 94d59a11ca..5be8df2b51 100644
--- a/audio/filter/af_lavcac3enc.c
+++ b/audio/filter/af_lavcac3enc.c
@@ -75,16 +75,15 @@ static int control(struct af_instance *af, int cmd, void *arg)
         if (AF_FORMAT_IS_AC3(data->format) || data->nch < s->min_channel_num)
             return AF_DETACH;
 
-        af->data->format = s->in_sampleformat;
-        af->data->bps = af_fmt2bits(s->in_sampleformat) / 8;
+        mp_audio_set_format(af->data, s->in_sampleformat);
         if (data->rate == 48000 || data->rate == 44100 || data->rate == 32000)
             af->data->rate = data->rate;
         else
             af->data->rate = 48000;
         if (data->nch > AC3_MAX_CHANNELS)
-            af->data->nch = AC3_MAX_CHANNELS;
+            mp_audio_set_num_channels(af->data, AC3_MAX_CHANNELS);
         else
-            af->data->nch = data->nch;
+            mp_audio_set_num_channels(af->data, data->nch);
         test_output_res = af_test_output(af, data);
 
         s->pending_len = 0;
@@ -123,9 +122,8 @@ static int control(struct af_instance *af, int cmd, void *arg)
                    "encoder frame size %d\n", s->lavc_actx->frame_size);
             return AF_ERROR;
         }
-        af->data->format = AF_FORMAT_AC3_BE;
-        af->data->bps = 2;
-        af->data->nch = 2;
+        mp_audio_set_format(af->data, AF_FORMAT_AC3_BE);
+        mp_audio_set_num_channels(af->data, 2);
         return test_output_res;
     case AF_CONTROL_COMMAND_LINE:
         mp_msg(MSGT_AFILTER, MSGL_DBG2, "af_lavcac3enc cmdline: %s.\n", (char*)arg);
@@ -316,8 +314,8 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
         buf += len;
     }
     c->audio = l->audio;
-    c->nch   = 2;
-    c->bps   = 2;
+    mp_audio_set_num_channels(c, 2);
+    mp_audio_set_format(c, af->data->format);
     c->len   = outsize;
     mp_msg(MSGT_AFILTER, MSGL_DBG2, "play return size %d, pending %d\n",
            outsize, s->pending_len);
diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c
index 2dd2c613c7..1f142792df 100644
--- a/audio/filter/af_lavrresample.c
+++ b/audio/filter/af_lavrresample.c
@@ -137,21 +137,19 @@ static int control(struct af_instance *af, int cmd, void *arg)
             out->rate = in->rate;
 
         if (out->nch == 0)
-            out->nch = in->nch;
+            mp_audio_set_num_channels(out, in->nch);
 
         enum AVSampleFormat in_samplefmt = af_to_avformat(in->format);
         if (in_samplefmt == AV_SAMPLE_FMT_NONE) {
-            in->format = AF_FORMAT_FLOAT_NE;
+            mp_audio_set_format(in, AF_FORMAT_FLOAT_NE);
             in_samplefmt = af_to_avformat(in->format);
         }
         enum AVSampleFormat out_samplefmt = af_to_avformat(out->format);
         if (out_samplefmt == AV_SAMPLE_FMT_NONE) {
-            out->format = in->format;
+            mp_audio_set_format(out, in->format);
             out_samplefmt = in_samplefmt;
         }
 
-        out->bps    = af_fmt2bits(out->format) / 8;
-        in->bps     = af_fmt2bits(in->format) / 8;
         af->mul     = (double) (out->rate * out->nch) / (in->rate * in->nch);
         af->delay   = out->nch * s->opts.filter_size / FFMIN(af->mul, 1);
 
@@ -196,16 +194,14 @@ static int control(struct af_instance *af, int cmd, void *arg)
         }
 
         return ((in->format == orig_in.format) &&
-                (in->bps    == orig_in.bps) &&
-                (in->nch    == orig_in.nch))
+                (in->nch == orig_in.nch))
                ? AF_OK : AF_FALSE;
     }
     case AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET: {
         if (af_to_avformat(*(int*)arg) == AV_SAMPLE_FMT_NONE)
             return AF_FALSE;
 
-        af->data->format = *(int*)arg;
-        af->data->bps = af_fmt2bits(af->data->format)/8;
+        mp_audio_set_format(af->data, *(int*)arg);
         return AF_OK;
     }
     case AF_CONTROL_CHANNELS | AF_CONTROL_SET: {
@@ -214,7 +210,7 @@ static int control(struct af_instance *af, int cmd, void *arg)
         if (nch < 1 || nch > AF_NCH)
             return AF_ERROR;
 
-        af->data->nch = nch;
+        mp_audio_set_num_channels(af->data, nch);
         return AF_OK;
     }
     case AF_CONTROL_COMMAND_LINE: {
diff --git a/audio/filter/af_pan.c b/audio/filter/af_pan.c
index 6468d14042..398ffe3b33 100644
--- a/audio/filter/af_pan.c
+++ b/audio/filter/af_pan.c
@@ -45,15 +45,13 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!arg) return AF_ERROR;
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->format = AF_FORMAT_FLOAT_NE;
-    af->data->bps    = 4;
-    af->data->nch    = s->nch ? s->nch: ((struct mp_audio*)arg)->nch;
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
+    mp_audio_set_num_channels(af->data, s->nch ? s->nch: ((struct mp_audio*)arg)->nch);
     af->mul          = (double)af->data->nch / ((struct mp_audio*)arg)->nch;
 
     if((af->data->format != ((struct mp_audio*)arg)->format) ||
        (af->data->bps != ((struct mp_audio*)arg)->bps)){
-      ((struct mp_audio*)arg)->format = af->data->format;
-      ((struct mp_audio*)arg)->bps = af->data->bps;
+      mp_audio_set_format((struct mp_audio*)arg, af->data->format);
       return AF_FALSE;
     }
     return AF_OK;
@@ -178,7 +176,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
   // Set output data
   c->audio = l->audio;
   c->len   = c->len / c->nch * l->nch;
-  c->nch   = l->nch;
+  mp_audio_set_num_channels(c, l->nch);
 
   return c;
 }
diff --git a/audio/filter/af_scaletempo.c b/audio/filter/af_scaletempo.c
index 657fd7f712..5cf0f3b082 100644
--- a/audio/filter/af_scaletempo.c
+++ b/audio/filter/af_scaletempo.c
@@ -302,29 +302,22 @@ static int control(struct af_instance* af, int cmd, void* arg)
            "[scaletempo] %.3f speed * %.3f scale_nominal = %.3f\n",
            s->speed, s->scale_nominal, s->scale);
 
+    mp_audio_copy_config(af->data, data);
+
     if (s->scale == 1.0) {
       if (s->speed_tempo && s->speed_pitch)
         return AF_DETACH;
-      af->data->format = data->format;
-      af->data->nch    = data->nch;
-      af->data->rate   = data->rate;
-      af->data->bps    = data->bps;
       af->delay = 0;
       af->mul = 1;
       return af_test_output(af, data);
     }
 
-    af->data->rate = data->rate;
-    af->data->nch  = data->nch;
-    if ( data->format == AF_FORMAT_S16_LE
-         || data->format == AF_FORMAT_S16_BE ) {
+    if (data->format == AF_FORMAT_S16_NE) {
       use_int = 1;
-      af->data->format  = AF_FORMAT_S16_NE;
-      af->data->bps     = bps = 2;
     } else {
-      af->data->format = AF_FORMAT_FLOAT_NE;
-      af->data->bps    = bps = 4;
+      mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
     }
+    bps = af->data->bps;
 
     frames_stride           = srate * s->ms_stride;
     s->bytes_stride         = frames_stride * bps * nch;
diff --git a/audio/filter/af_sinesuppress.c b/audio/filter/af_sinesuppress.c
index c9a77ab498..10f0b650ec 100644
--- a/audio/filter/af_sinesuppress.c
+++ b/audio/filter/af_sinesuppress.c
@@ -54,8 +54,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Sanity check
     if(!arg) return AF_ERROR;
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = 1;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_num_channels(af->data, 1);
 #if 0
     if (((struct mp_audio*)arg)->format == AF_FORMAT_FLOAT_NE)
     {
@@ -65,8 +65,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
     }// else
 #endif
     {
-	af->data->format = AF_FORMAT_S16_NE;
-	af->data->bps = 2;
+        mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
 	af->play = play_s16;
     }
 
diff --git a/audio/filter/af_sub.c b/audio/filter/af_sub.c
index 4af28d9141..1ae65a0d11 100644
--- a/audio/filter/af_sub.c
+++ b/audio/filter/af_sub.c
@@ -70,9 +70,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     if(!arg) return AF_ERROR;
 
     af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = max(s->ch+1,((struct mp_audio*)arg)->nch);
-    af->data->format = AF_FORMAT_FLOAT_NE;
-    af->data->bps    = 4;
+    mp_audio_set_num_channels(af->data, max(s->ch+1,((struct mp_audio*)arg)->nch));
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
     // Design low-pass filter
     s->k = 1.0;
diff --git a/audio/filter/af_surround.c b/audio/filter/af_surround.c
index 57288d6ba2..c63105481f 100644
--- a/audio/filter/af_surround.c
+++ b/audio/filter/af_surround.c
@@ -92,10 +92,9 @@ static int control(struct af_instance* af, int cmd, void* arg)
   switch(cmd){
   case AF_CONTROL_REINIT:{
     float fc;
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch*2;
-    af->data->format = AF_FORMAT_FLOAT_NE;
-    af->data->bps    = 4;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
+    mp_audio_set_num_channels(af->data, ((struct mp_audio*)arg)->nch*2);
+    mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
 
     if (af->data->nch != 4){
       mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Only stereo input is supported.\n");
@@ -125,8 +124,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
     if((af->data->format != ((struct mp_audio*)arg)->format) ||
        (af->data->bps    != ((struct mp_audio*)arg)->bps)){
-      ((struct mp_audio*)arg)->format = af->data->format;
-      ((struct mp_audio*)arg)->bps = af->data->bps;
+      mp_audio_set_format((struct mp_audio*)arg, af->data->format);
       return AF_FALSE;
     }
     return AF_OK;
@@ -244,7 +242,7 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data){
   // Set output data
   data->audio = af->data->audio;
   data->len   *= 2;
-  data->nch   = af->data->nch;
+  mp_audio_set_num_channels(data, af->data->nch);
 
   return data;
 }
diff --git a/audio/filter/af_sweep.c b/audio/filter/af_sweep.c
index 6d1106fefc..6cc099f2d8 100644
--- a/audio/filter/af_sweep.c
+++ b/audio/filter/af_sweep.c
@@ -41,12 +41,10 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
   switch(cmd){
   case AF_CONTROL_REINIT:
-    af->data->nch    = data->nch;
-    af->data->format = AF_FORMAT_S16_NE;
-    af->data->bps    = 2;
-    af->data->rate   = data->rate;
+    mp_audio_copy_config(af->data, data);
+    mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
 
-    return AF_OK;
+    return af_test_output(af, data);
   case AF_CONTROL_COMMAND_LINE:
     sscanf((char*)arg,"%lf", &s->delta);
     return AF_OK;
diff --git a/audio/filter/af_volume.c b/audio/filter/af_volume.c
index 5664443f03..82c31eaa12 100644
--- a/audio/filter/af_volume.c
+++ b/audio/filter/af_volume.c
@@ -66,12 +66,10 @@ static int control(struct af_instance* af, int cmd, void* arg)
     // Sanity check
     if(!arg) return AF_ERROR;
 
-    af->data->rate   = ((struct mp_audio*)arg)->rate;
-    af->data->nch    = ((struct mp_audio*)arg)->nch;
+    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
 
     if(s->fast && (((struct mp_audio*)arg)->format != (AF_FORMAT_FLOAT_NE))){
-      af->data->format = AF_FORMAT_S16_NE;
-      af->data->bps    = 2;
+      mp_audio_set_format(af->data, AF_FORMAT_S16_NE);
     }
     else{
       // Cutoff set to 10Hz for forgetting factor
@@ -79,8 +77,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
       float t = 2.0-cos(x);
       s->time = 1.0 - (t - sqrt(t*t - 1));
       mp_msg(MSGT_AFILTER, MSGL_DBG2, "[volume] Forgetting factor = %0.5f\n",s->time);
-      af->data->format = AF_FORMAT_FLOAT_NE;
-      af->data->bps    = 4;
+      mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
     }
     return af_test_output(af,(struct mp_audio*)arg);
   case AF_CONTROL_COMMAND_LINE:{
diff --git a/audio/format.c b/audio/format.c
index 9625857ada..012d24b010 100644
--- a/audio/format.c
+++ b/audio/format.c
@@ -29,6 +29,8 @@
 int af_fmt2bits(int format)
 {
     if (AF_FORMAT_IS_AC3(format)) return 16;
+    if (format == AF_FORMAT_UNKNOWN)
+        return 0;
     return (format & AF_FORMAT_BITS_MASK)+8;
 //    return (((format & AF_FORMAT_BITS_MASK)>>3)+1) * 8;
 #if 0
diff --git a/audio/out/ao.h b/audio/out/ao.h
index 4a7c928824..e57fff7a95 100644
--- a/audio/out/ao.h
+++ b/audio/out/ao.h
@@ -88,7 +88,7 @@ struct ao {
     int samplerate;
     int channels;
     int format;
-    int bps;
+    int bps; // bytes per second
     int outburst;
     int buffersize;
     double pts;
-- 
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