1 files changed, 110 insertions, 0 deletions
diff --git a/audio/filter/af_tools.c b/audio/filter/af_tools.c
new file mode 100644
index 0000000000..0d5dc6c573
--- /dev/null
+++ b/audio/filter/af_tools.c
@@ -0,0 +1,110 @@
+/*
+ * 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 "af.h"
+
+/* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if
+   fail */
+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,clamp(in[i],mi,ma)/k);
+  }
+  return AF_OK;
+}
+
+/* Convert from gain value to dB. Returns AF_OK if of and AF_ERROR if
+   fail */
+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 * clamp(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;
+}
+
+/* Helper function for testing the output format */
+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)   ||
+     (af->data->nch    != out->nch)){
+    memcpy(out,af->data,sizeof(struct mp_audio));
+    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);
+}