From 983f5efa3c3dfb8d8b36d48249cff11154b17fa3 Mon Sep 17 00:00:00 2001
From: wm4 <wm4@nowhere>
Date: Tue, 13 Jan 2015 20:16:10 +0100
Subject: audio/filters: use refcounted frames for some in-place filters

These are also quite simple, but require requesting write access to the
frames. The error handling (for OOM) is a bit annoying.
---
 audio/filter/af_convertsignendian.c | 12 ++++++++++--
 audio/filter/af_drc.c               | 13 +++++++++++--
 audio/filter/af_equalizer.c         | 13 ++++++++++---
 3 files changed, 31 insertions(+), 7 deletions(-)

diff --git a/audio/filter/af_convertsignendian.c b/audio/filter/af_convertsignendian.c
index 52ae8164df..abbd260a0f 100644
--- a/audio/filter/af_convertsignendian.c
+++ b/audio/filter/af_convertsignendian.c
@@ -71,8 +71,15 @@ static void si2us(void *data, int len, int bps)
     } while (i += bps);
 }
 
-static int filter(struct af_instance *af, struct mp_audio *data, int flags)
+static int filter(struct af_instance *af, struct mp_audio *data)
 {
+    if (!data)
+        return 0;
+    if (af_make_writeable(af, data) < 0) {
+        talloc_free(data);
+        return -1;
+    }
+
     int infmt = data->format;
     int outfmt = af->data->format;
     size_t len = data->samples * data->nch;
@@ -81,13 +88,14 @@ static int filter(struct af_instance *af, struct mp_audio *data, int flags)
         si2us(data->planes[0], len, data->bps);
 
     mp_audio_set_format(data, outfmt);
+    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;
     return AF_OK;
 }
 
diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c
index 94e161130f..2560658f45 100644
--- a/audio/filter/af_drc.c
+++ b/audio/filter/af_drc.c
@@ -272,10 +272,18 @@ static void method2_float(af_drc_t *s, struct mp_audio *c)
   s->idx = (s->idx + 1) % NSAMPLES;
 }
 
-static int filter(struct af_instance* af, struct mp_audio* data, int flags)
+static int filter(struct af_instance *af, struct mp_audio *data)
 {
   af_drc_t *s = af->priv;
 
+  if (!data)
+    return 0;
+
+  if (af_make_writeable(af, data) < 0) {
+    talloc_free(data);
+    return -1;
+  }
+
   if(af->data->format == (AF_FORMAT_S16))
   {
     if (s->method == 2)
@@ -290,6 +298,7 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
     else
         method1_float(s, data);
   }
+  af_add_output_frame(af, data);
   return 0;
 }
 
@@ -297,7 +306,7 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
 static int af_open(struct af_instance* af){
   int i = 0;
   af->control=control;
-  af->filter=filter;
+  af->filter_frame = filter;
   af_drc_t *priv = af->priv;
 
   priv->mul = MUL_INIT;
diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c
index 4e96223ee7..f5d47c8e3a 100644
--- a/audio/filter/af_equalizer.c
+++ b/audio/filter/af_equalizer.c
@@ -141,14 +141,20 @@ 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(struct af_instance* af, struct mp_audio* data)
 {
   struct mp_audio*       c      = data;                         // Current working data
+  if (!c)
+    return 0;
   af_equalizer_t*  s    = (af_equalizer_t*)af->priv;    // Setup
   uint32_t         ci   = af->data->nch;                // Index for channels
   uint32_t         nch  = af->data->nch;                // Number of channels
 
+  if (af_make_writeable(af, data) < 0) {
+    talloc_free(data);
+    return -1;
+  }
+
   while(ci--){
     float*      g   = s->g[ci];      // Gain factor
     float*      in  = ((float*)c->planes[0])+ci;
@@ -177,13 +183,14 @@ static int filter(struct af_instance* af, struct mp_audio* data, int flags)
       out+=nch;
     }
   }
+  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;
   af_equalizer_t *priv = af->priv;
   for(int i=0;i<AF_NCH;i++){
       for(int j=0;j<KM;j++){
-- 
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