From 85f6349c78804ea662606fc859d2f376bfdfe606 Mon Sep 17 00:00:00 2001
From: wm4 <wm4@nowhere>
Date: Mon, 18 Nov 2013 18:48:00 +0100
Subject: audio/filter: rename af_tools.c to tools.c

This always bothered me.
---
 audio/filter/af_tools.c | 123 ------------------------------------------------
 audio/filter/tools.c    | 123 ++++++++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 123 insertions(+), 123 deletions(-)
 delete mode 100644 audio/filter/af_tools.c
 create mode 100644 audio/filter/tools.c

(limited to 'audio')

diff --git a/audio/filter/af_tools.c b/audio/filter/af_tools.c
deleted file mode 100644
index c5f423ebf9..0000000000
--- a/audio/filter/af_tools.c
+++ /dev/null
@@ -1,123 +0,0 @@
-/*
- * This file is part of MPlayer.
- *
- * MPlayer 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.
- *
- * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <math.h>
-#include <string.h>
-
-#include "mpvcore/mp_common.h"
-#include "af.h"
-
-/* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if
- * fail. input <= -200dB will become 0 gain. */
-int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
-{
-  int i = 0;
-  // Sanity check
-  if(!in || !out)
-    return AF_ERROR;
-
-  for(i=0;i<n;i++){
-    if(in[i]<=-200)
-      out[i]=0.0;
-    else
-      out[i]=pow(10.0,MPCLAMP(in[i],mi,ma)/k);
-  }
-  return AF_OK;
-}
-
-/* Convert from gain value to dB. Returns AF_OK if of and AF_ERROR if
- * fail. gain=0 will become -200 dB. k is just a multiplier. */
-int af_to_dB(int n, float* in, float* out, float k)
-{
-  int i = 0;
-  // Sanity check
-  if(!in || !out)
-    return AF_ERROR;
-
-  for(i=0;i<n;i++){
-    if(in[i] == 0.0)
-      out[i]=-200.0;
-    else
-      out[i]=k*log10(in[i]);
-  }
-  return AF_OK;
-}
-
-/* Convert from ms to sample time */
-int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
-{
-  int i = 0;
-  // Sanity check
-  if(!in || !out)
-    return AF_ERROR;
-
-  for(i=0;i<n;i++)
-    out[i]=(int)((float)rate * MPCLAMP(in[i],mi,ma)/1000.0);
-
-  return AF_OK;
-}
-
-/* Convert from sample time to ms */
-int af_to_ms(int n, int* in, float* out, int rate)
-{
-  int i = 0;
-  // Sanity check
-  if(!in || !out || !rate)
-    return AF_ERROR;
-
-  for(i=0;i<n;i++)
-    out[i]=1000.0 * (float)in[i]/((float)rate);
-
-  return AF_OK;
-}
-
-/*
- * test if output format matches
- *  af: audio filter
- *  out: needed format, will be overwritten by available
- *       format if they do not match
- *  returns: AF_FALSE if formats do not match, AF_OK if they match
- *
- * compares the format, rate and nch values of af->data with out
- * Note: logically, *out=*af->data always happens, because out contains the
- *       format only, no actual audio data or memory allocations. *out always
- *       contains the parameters from af->data after the function returns.
- */
-int af_test_output(struct af_instance* af, struct mp_audio* out)
-{
-  if((af->data->format != out->format) ||
-     (af->data->bps    != out->bps)    ||
-     (af->data->rate   != out->rate)   ||
-     !mp_chmap_equals(&af->data->channels, &out->channels)){
-    *out = *af->data;
-    return AF_FALSE;
-  }
-  return AF_OK;
-}
-
-/* Soft clipping, the sound of a dream, thanks to Jon Wattes
-   post to Musicdsp.org */
-float af_softclip(float a)
-{
-    if (a >= M_PI/2)
-	return 1.0;
-    else if (a <= -M_PI/2)
-	return -1.0;
-    else
-	return sin(a);
-}
diff --git a/audio/filter/tools.c b/audio/filter/tools.c
new file mode 100644
index 0000000000..c5f423ebf9
--- /dev/null
+++ b/audio/filter/tools.c
@@ -0,0 +1,123 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer 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.
+ *
+ * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <math.h>
+#include <string.h>
+
+#include "mpvcore/mp_common.h"
+#include "af.h"
+
+/* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if
+ * fail. input <= -200dB will become 0 gain. */
+int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
+{
+  int i = 0;
+  // Sanity check
+  if(!in || !out)
+    return AF_ERROR;
+
+  for(i=0;i<n;i++){
+    if(in[i]<=-200)
+      out[i]=0.0;
+    else
+      out[i]=pow(10.0,MPCLAMP(in[i],mi,ma)/k);
+  }
+  return AF_OK;
+}
+
+/* Convert from gain value to dB. Returns AF_OK if of and AF_ERROR if
+ * fail. gain=0 will become -200 dB. k is just a multiplier. */
+int af_to_dB(int n, float* in, float* out, float k)
+{
+  int i = 0;
+  // Sanity check
+  if(!in || !out)
+    return AF_ERROR;
+
+  for(i=0;i<n;i++){
+    if(in[i] == 0.0)
+      out[i]=-200.0;
+    else
+      out[i]=k*log10(in[i]);
+  }
+  return AF_OK;
+}
+
+/* Convert from ms to sample time */
+int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
+{
+  int i = 0;
+  // Sanity check
+  if(!in || !out)
+    return AF_ERROR;
+
+  for(i=0;i<n;i++)
+    out[i]=(int)((float)rate * MPCLAMP(in[i],mi,ma)/1000.0);
+
+  return AF_OK;
+}
+
+/* Convert from sample time to ms */
+int af_to_ms(int n, int* in, float* out, int rate)
+{
+  int i = 0;
+  // Sanity check
+  if(!in || !out || !rate)
+    return AF_ERROR;
+
+  for(i=0;i<n;i++)
+    out[i]=1000.0 * (float)in[i]/((float)rate);
+
+  return AF_OK;
+}
+
+/*
+ * test if output format matches
+ *  af: audio filter
+ *  out: needed format, will be overwritten by available
+ *       format if they do not match
+ *  returns: AF_FALSE if formats do not match, AF_OK if they match
+ *
+ * compares the format, rate and nch values of af->data with out
+ * Note: logically, *out=*af->data always happens, because out contains the
+ *       format only, no actual audio data or memory allocations. *out always
+ *       contains the parameters from af->data after the function returns.
+ */
+int af_test_output(struct af_instance* af, struct mp_audio* out)
+{
+  if((af->data->format != out->format) ||
+     (af->data->bps    != out->bps)    ||
+     (af->data->rate   != out->rate)   ||
+     !mp_chmap_equals(&af->data->channels, &out->channels)){
+    *out = *af->data;
+    return AF_FALSE;
+  }
+  return AF_OK;
+}
+
+/* Soft clipping, the sound of a dream, thanks to Jon Wattes
+   post to Musicdsp.org */
+float af_softclip(float a)
+{
+    if (a >= M_PI/2)
+	return 1.0;
+    else if (a <= -M_PI/2)
+	return -1.0;
+    else
+	return sin(a);
+}
-- 
cgit v1.2.3