Merge changes from team/russell/codec_resample

This commit imports libresample for use in Asterisk.  It also adds a new codec
module, codec_resample.  This module uses libresample to re-sample signed linear
audio between 8 kHz and 16 kHz.

It also provides an alternative for converting between 16 kHz G.722 and 8 kHz
signed linear when using G.722, which will likely be useful as some people have
complained about volume issues when the current codec_g722 converts to 8 kHz 
signed linear.  But, to test this, you will have to disable the g722-to-slin and
g722-to-slin16 translators in codec_g722.c.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95501 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 28 insertions, 0 deletions

diff --git a/main/libresample/src/filterkit.h b/main/libresample/src/filterkit.h
new file mode 100644
index 000000000..9df0ae869
--- /dev/null
+++ b/main/libresample/src/filterkit.h
@@ -0,0 +1,28 @@
+/**********************************************************************
+
+  resamplesubs.c
+
+  Real-time library interface by Dominic Mazzoni
+
+  Based on resample-1.7:
+    http://www-ccrma.stanford.edu/~jos/resample/
+
+  License: LGPL - see the file LICENSE.txt for more information
+
+**********************************************************************/
+
+/* Definitions */
+#include "resample_defs.h"
+
+/*
+ * FilterUp() - Applies a filter to a given sample when up-converting.
+ * FilterUD() - Applies a filter to a given sample when up- or down-
+ */
+
+float lrsFilterUp(float Imp[], float ImpD[], UWORD Nwing, BOOL Interp,
+                  float *Xp, double Ph, int Inc);
+
+float lrsFilterUD(float Imp[], float ImpD[], UWORD Nwing, BOOL Interp,
+                  float *Xp, double Ph, int Inc, double dhb);
+
+void lrsLpFilter(double c[], int N, double frq, double Beta, int Num);