audio/filter: switch remaining in-place filters to refcounting

Adds about 7 lines of boilerplate per filter. This could be avoided by
providing a different entrypoint (something like af->filter_inplace),
which would basically mirror the old interface exactly for this kind of
filter. But I feel like it would just be a hack to support all those
old, useless filters better. (The ideal solution would be using a
language that can do closures to provide a compat. wrapper, but
whatever.)

af_bs2b has terribly repetitious code for setting up filter functions
for each format (most of them useless, in addition to bs2b being
useless), so I did something terrible with macros.

af_sinesuppress had commented code for float filtering (maybe it was
broken; it has been commented every since it was added in 2006). Remove
this code.

9 files changed, 134 insertions, 127 deletions

diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c
index 29d646a9a9..874dbc6786 100644
--- a/audio/filter/af_bs2b.c
+++ b/audio/filter/af_bs2b.c
@@ -37,26 +37,49 @@ struct af_bs2b {
     t_bs2bdp filter;    ///< instance of a library filter
 };
 
-#define FILTER(name, type) \
-static int filter_##name(struct af_instance *af, struct mp_audio *data, int f) \
+#define DEF_FILTER(fmt, name) \
+static int filter_##name(struct af_instance *af, struct mp_audio *data) \
 { \
-    /* filter is called for all pairs of samples available in the buffer */ \
+    if (!data) \
+        return 0; \
+    if (af_make_writeable(af, data) < 0) { \
+        talloc_free(data); \
+        return -1; \
+    } \
     bs2b_cross_feed_##name(((struct af_bs2b*)(af->priv))->filter, \
-        (type*)(data->planes[0]), data->samples); \
-\
+        (data->planes[0]), data->samples); \
+    af_add_output_frame(af, data); \
     return 0; \
 }
 
-FILTER(f, float)
-FILTER(s32, int32_t)
-FILTER(u32, uint32_t)
-FILTER(s24, bs2b_int24_t)
-FILTER(u24, bs2b_uint24_t)
-FILTER(s16, int16_t)
-FILTER(u16, uint16_t)
-FILTER(s8, int8_t)
-FILTER(u8, uint8_t)
-
+#define GET_FILTER(fmt, name) \
+    case AF_FORMAT_##fmt: return filter_##name;
+
+#define FILTERS \
+    FILTER(FLOAT,   f)      \
+    FILTER(S32,     s32)    \
+    FILTER(U32,     u32)    \
+    FILTER(S24,     s24)    \
+    FILTER(U24,     u24)    \
+    FILTER(S16,     s16)    \
+    FILTER(U16,     u16)    \
+    FILTER(S8,      s8)     \
+    FILTER(U8,      u8)
+
+#define FILTER DEF_FILTER
+FILTERS
+#undef FILTER
+
+typedef int (*filter)(struct af_instance *af, struct mp_audio *d);
+static filter get_filter(int fmt)
+{
+    switch (fmt) {
+#define FILTER GET_FILTER
+FILTERS
+#undef FILTER
+    default: return NULL;
+    }
+}
 
 /// Initialization and runtime control
 static int control(struct af_instance *af, int cmd, void *arg)
@@ -76,38 +99,10 @@ static int control(struct af_instance *af, int cmd, void *arg)
 
         /* check for formats supported by libbs2b
            and assign corresponding handlers */
-        switch (format) {
-            case AF_FORMAT_FLOAT:
-                af->filter = filter_f;
-                break;
-            case AF_FORMAT_S32:
-                af->filter = filter_s32;
-                break;
-            case AF_FORMAT_U32:
-                af->filter = filter_u32;
-                break;
-            case AF_FORMAT_S24:
-                af->filter = filter_s24;
-                break;
-            case AF_FORMAT_U24:
-                af->filter = filter_u24;
-                break;
-            case AF_FORMAT_S16:
-                af->filter = filter_s16;
-                break;
-            case AF_FORMAT_U16:
-                af->filter = filter_u16;
-                break;
-            case AF_FORMAT_S8:
-                af->filter = filter_s8;
-                break;
-            case AF_FORMAT_U8:
-                af->filter = filter_u8;
-                break;
-            default:
-                af->filter = filter_f;
-                mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
-                break;
+        af->filter_frame = get_filter(format);
+        if (!af->filter_frame) {
+            af->filter_frame = filter_f;
+            mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
         }
 
         // bs2b have srate limits, try to resample if needed
diff --git a/audio/filter/af_center.c b/audio/filter/af_center.c
index 3039e50302..ff2a37dee9 100644
--- a/audio/filter/af_center.c
+++ b/audio/filter/af_center.c
@@ -58,9 +58,14 @@ static int control(struct af_instance* af, int cmd, void* arg)
   return AF_UNKNOWN;
 }
 
-// Filter data through filter
-static int filter(struct af_instance* af, struct mp_audio* data, int flags)
+static int filter_frame(struct af_instance* af, struct mp_audio* data)
 {
+  if (!data)
+    return 0;
+  if (af_make_writeable(af, data) < 0) {
+    talloc_free(data);
+    return -1;
+  }
   struct mp_audio*    c   = data;        // Current working data
   af_center_t*  s   = af->priv; // Setup for this instance
   float*        a   = c->planes[0];      // Audio data
@@ -75,13 +80,14 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
     a[i+ch] = (a[i]/2) + (a[i+1]/2);
   }
 
+  af_add_output_frame(af, data);
   return 0;
 }
 
 // Allocate memory and set function pointers
 static int af_open(struct af_instance* af){
   af->control=control;
-  af->filter=filter;
+  af->filter_frame = filter_frame;
   return AF_OK;
 }
 
diff --git a/audio/filter/af_delay.c b/audio/filter/af_delay.c
index 56013b3270..8de26ece77 100644
--- a/audio/filter/af_delay.c
+++ b/audio/filter/af_delay.c
@@ -96,15 +96,19 @@ static void uninit(struct af_instance* af)
       free(((af_delay_t*)(af->priv))->q[i]);
 }
 
-// Filter data through filter
-static int filter(struct af_instance* af, struct mp_audio* data, int flags)
+static int filter_frame(struct af_instance *af, struct mp_audio *c)
 {
-  struct mp_audio*      c   = data;      // Current working data
+  if (!c)
+    return 0;
   af_delay_t*   s   = af->priv; // Setup for this instance
   int           nch = c->nch;    // Number of channels
   int           len = mp_audio_psize(c)/c->bps; // Number of sample in data chunk
   int           ri  = 0;
   int           ch,i;
+  if (af_make_writeable(af, c) < 0) {
+    talloc_free(c);
+    return -1;
+  }
   for(ch=0;ch<nch;ch++){
     switch(c->bps){
     case 1:{
@@ -152,6 +156,7 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
     }
   }
   s->ri = ri;
+  af_add_output_frame(af, c);
   return 0;
 }
 
@@ -159,7 +164,7 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
 static int af_open(struct af_instance* af){
     af->control=control;
     af->uninit=uninit;
-    af->filter=filter;
+    af->filter_frame = filter_frame;
     af_delay_t *s = af->priv;
     int n = 1;
     int i = 0;
diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c
index 4ebe03a228..4b39793e09 100644
--- a/audio/filter/af_extrastereo.c
+++ b/audio/filter/af_extrastereo.c
@@ -35,9 +35,6 @@ typedef struct af_extrastereo_s
     float mul;
 }af_extrastereo_t;
 
-static int play_s16(struct af_instance* af, struct mp_audio* data, int f);
-static int play_float(struct af_instance* af, struct mp_audio* data, int f);
-
 // Initialization and runtime control
 static int control(struct af_instance* af, int cmd, void* arg)
 {
@@ -49,14 +46,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
     mp_audio_copy_config(af->data, (struct mp_audio*)arg);
     mp_audio_force_interleaved_format(af->data);
     mp_audio_set_num_channels(af->data, 2);
-    if (af->data->format == AF_FORMAT_FLOAT)
-    {
-        af->filter = play_float;
-    }// else
-    {
+    if (af->data->format != AF_FORMAT_FLOAT)
         mp_audio_set_format(af->data, AF_FORMAT_S16);
-        af->filter = play_s16;
-    }
 
     return af_test_output(af,(struct mp_audio*)arg);
   }
@@ -65,9 +56,8 @@ static int control(struct af_instance* af, int cmd, void* arg)
 }
 
 // Filter data through filter
-static int play_s16(struct af_instance* af, struct mp_audio* data, int f)
+static void play_s16(af_extrastereo_t *s, struct mp_audio* data)
 {
-  af_extrastereo_t *s = af->priv;
   register int i = 0;
   int16_t *a = (int16_t*)data->planes[0];       // Audio data
   int len = data->samples*data->nch;            // Number of samples
@@ -83,13 +73,10 @@ static int play_s16(struct af_instance* af, struct mp_audio* data, int f)
     a[i] = MPCLAMP(l, SHRT_MIN, SHRT_MAX);
     a[i + 1] = MPCLAMP(r, SHRT_MIN, SHRT_MAX);
   }
-
-  return 0;
 }
 
-static int play_float(struct af_instance* af, struct mp_audio* data, int f)
+static void play_float(af_extrastereo_t *s, struct mp_audio* data)
 {
-  af_extrastereo_t *s = af->priv;
   register int i = 0;
   float *a = (float*)data->planes[0];   // Audio data
   int len = data->samples * data->nch;  // Number of samples
@@ -105,14 +92,29 @@ static int play_float(struct af_instance* af, struct mp_audio* data, int f)
     a[i] = af_softclip(l);
     a[i + 1] = af_softclip(r);
   }
+}
 
-  return 0;
+static int filter_frame(struct af_instance *af, struct mp_audio *data)
+{
+    if (!data)
+        return 0;
+    if (af_make_writeable(af, data) < 0) {
+        talloc_free(data);
+        return -1;
+    }
+    if (data->format == AF_FORMAT_FLOAT) {
+        play_float(af->priv, data);
+    } else {
+        play_s16(af->priv, data);
+    }
+    af_add_output_frame(af, data);
+    return 0;
 }
 
 // Allocate memory and set function pointers
 static int af_open(struct af_instance* af){
   af->control=control;
-  af->filter=play_s16;
+  af->filter_frame = filter_frame;
 
   return AF_OK;
 }
diff --git a/audio/filter/af_karaoke.c b/audio/filter/af_karaoke.c
index 3187a41f2f..f69b936811 100644
--- a/audio/filter/af_karaoke.c
+++ b/audio/filter/af_karaoke.c
@@ -41,10 +41,15 @@ static int control(struct af_instance* af, int cmd, void* arg)
         return AF_UNKNOWN;
 }
 
-// Filter data through filter
-static int play(struct af_instance* af, struct mp_audio* data, int flags)
+static int filter_frame(struct af_instance *af, struct mp_audio *c)
 {
-        struct mp_audio*        c       = data;          // Current working data
+        if (!c)
+                return 0;
+        if (af_make_writeable(af, c) < 0) {
+                talloc_free(c);
+                return 0;
+        }
+
         float*          a       = c->planes[0];  // Audio data
         int                     nch     = c->nch;        // Number of channels
         int                     len     = c->samples*nch;        // Number of samples in current audio block
@@ -62,13 +67,14 @@ static int play(struct af_instance* af, struct mp_audio* data, int flags)
                 a[i+1]=a[i];
         }
 
+        af_add_output_frame(af, c);
         return 0;
 }
 
 // Allocate memory and set function pointers
 static int af_open(struct af_instance* af){
         af->control     = control;
-        af->filter      = play;
+        af->filter_frame = filter_frame;
         return AF_OK;
 }
 
diff --git a/audio/filter/af_ladspa.c b/audio/filter/af_ladspa.c
index ab76c35fe9..6a8cd802f6 100644
--- a/audio/filter/af_ladspa.c
+++ b/audio/filter/af_ladspa.c
@@ -565,7 +565,10 @@ static void uninit(struct af_instance *af) {
  * \return      Either AF_ERROR or AF_OK
  */
 
-static int filter(struct af_instance *af, struct mp_audio *data, int flags) {
+static int filter_frame(struct af_instance *af, struct mp_audio *data)
+{
+    if (!data)
+        return 0;
     af_ladspa_t *setup = af->priv;
     const LADSPA_Descriptor *pdes = setup->plugin_descriptor;
     float *audio = (float*)data->planes[0];
@@ -574,8 +577,14 @@ static int filter(struct af_instance *af, struct mp_audio *data, int flags) {
     int rate = data->rate;
     int i, p;
 
-    if (setup->status !=AF_OK)
+    if (setup->status !=AF_OK) {
+        talloc_free(data);
+        return -1;
+    }
+    if (af_make_writeable(af, data) < 0) {
+        talloc_free(data);
         return -1;
+    }
 
     /* See if it's the first call. If so, setup inbufs/outbufs, instantiate
      * plugin, connect ports and activate plugin
@@ -721,6 +730,7 @@ static int filter(struct af_instance *af, struct mp_audio *data, int flags) {
 
     /* done */
 
+    af_add_output_frame(af, data);
     return 0;
 }
 
@@ -737,7 +747,7 @@ static int af_open(struct af_instance *af) {
 
     af->control=control;
     af->uninit=uninit;
-    af->filter=filter;
+    af->filter_frame = filter_frame;
 
     af_ladspa_t *setup = af->priv;
 
diff --git a/audio/filter/af_sinesuppress.c b/audio/filter/af_sinesuppress.c
index 95d2478b7e..32e074d508 100644
--- a/audio/filter/af_sinesuppress.c
+++ b/audio/filter/af_sinesuppress.c
@@ -41,9 +41,6 @@ typedef struct af_sinesuppress_s
     double pos;
 }af_sinesuppress_t;
 
-static int play_s16(struct af_instance* af, struct mp_audio* data, int f);
-//static struct mp_audio* play_float(struct af_instance* af, struct mp_audio* data);
-
 // Initialization and runtime control
 static int control(struct af_instance* af, int cmd, void* arg)
 {
@@ -54,18 +51,7 @@ static int control(struct af_instance* af, int cmd, void* arg)
 
     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)
-    {
-        af->data->format = AF_FORMAT_FLOAT;
-        af->data->bps = 4;
-        af->play = play_float;
-    }// else
-#endif
-    {
-        mp_audio_set_format(af->data, AF_FORMAT_S16);
-        af->filter = play_s16;
-    }
+    mp_audio_set_format(af->data, AF_FORMAT_S16);
 
     return af_test_output(af,(struct mp_audio*)arg);
   }
@@ -73,9 +59,15 @@ static int control(struct af_instance* af, int cmd, void* arg)
   return AF_UNKNOWN;
 }
 
-// Filter data through filter
-static int play_s16(struct af_instance* af, struct mp_audio* data, int f)
+static int play_s16(struct af_instance *af, struct mp_audio *data)
 {
+  if (!data)
+    return 0;
+  if (af_make_writeable(af, data) < 0) {
+    talloc_free(data);
+    return -1;
+  }
+
   af_sinesuppress_t *s = af->priv;
   register int i = 0;
   int16_t *a = (int16_t*)data->planes[0];       // Audio data
@@ -101,37 +93,14 @@ static int play_s16(struct af_instance* af, struct mp_audio* data, int f)
 
    MP_VERBOSE(af, "f:%8.2f: amp:%8.2f\n", s->freq, sqrt(s->real*s->real + s->imag*s->imag) / s->ref);
 
+  af_add_output_frame(af, data);
   return 0;
 }
 
-#if 0
-static struct mp_audio* play_float(struct af_instance* af, struct mp_audio* data)
-{
-  af_sinesuppress_t *s = af->setup;
-  register int i = 0;
-  float *a = (float*)data->audio;       // Audio data
-  int len = data->len/4;                // Number of samples
-  float avg, l, r;
-
-  for (i = 0; i < len; i+=2)
-  {
-    avg = (a[i] + a[i + 1]) / 2;
-
-/*    l = avg + (s->mul * (a[i] - avg));
-    r = avg + (s->mul * (a[i + 1] - avg));*/
-
-    a[i] = af_softclip(l);
-    a[i + 1] = af_softclip(r);
-  }
-
-  return data;
-}
-#endif
-
 // Allocate memory and set function pointers
 static int af_open(struct af_instance* af){
   af->control=control;
-  af->filter=play_s16;
+  af->filter_frame = play_s16;
   return AF_OK;
 }
 
diff --git a/audio/filter/af_sub.c b/audio/filter/af_sub.c
index 5642c4573d..243c9cef70 100644
--- a/audio/filter/af_sub.c
+++ b/audio/filter/af_sub.c
@@ -98,9 +98,15 @@ static int control(struct af_instance* af, int cmd, void* arg)
 }
 #endif
 
-// Filter data through filter
-static int filter(struct af_instance* af, struct mp_audio* data, int flags)
+static int filter_frame(struct af_instance *af, struct mp_audio *data)
 {
+  if (!data)
+    return 0;
+  if (af_make_writeable(af, data) < 0) {
+    talloc_free(data);
+    return -1;
+  }
+
   struct mp_audio*    c   = data;        // Current working data
   af_sub_t*     s   = af->priv; // Setup for this instance
   float*        a   = c->planes[0];      // Audio data
@@ -117,13 +123,14 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
     IIR(x , s->w[1], s->q[1], a[i+ch]);
   }
 
+  af_add_output_frame(af, data);
   return 0;
 }
 
 // Allocate memory and set function pointers
 static int af_open(struct af_instance* af){
   af->control=control;
-  af->filter=filter;
+  af->filter_frame = filter_frame;
   return AF_OK;
 }
 
diff --git a/audio/filter/af_sweep.c b/audio/filter/af_sweep.c
index 982cb6210c..854ef1446c 100644
--- a/audio/filter/af_sweep.c
+++ b/audio/filter/af_sweep.c
@@ -48,9 +48,15 @@ static int control(struct af_instance* af, int cmd, void* arg)
   return AF_UNKNOWN;
 }
 
-// Filter data through filter
-static int filter(struct af_instance* af, struct mp_audio* data, int f)
+static int filter_frame(struct af_instance *af, struct mp_audio *data)
 {
+  if (!data)
+    return 0;
+  if (af_make_writeable(af, data) < 0) {
+      talloc_free(data);
+    return 0;
+  }
+
   af_sweept *s = af->priv;
   int i, j;
   int16_t *in = (int16_t*)data->planes[0];
@@ -64,12 +70,13 @@ static int filter(struct af_instance* af, struct mp_audio* data, int f)
       if(2*s->x*s->delta >= 3.141592) s->x=0;
   }
 
+  af_add_output_frame(af, data);
   return 0;
 }
 
 static int af_open(struct af_instance* af){
   af->control=control;
-  af->filter=filter;
+  af->filter_frame = filter_frame;
   return AF_OK;
 }