- Begin adding support for Google WebRTC

- This revision adds support for WebRTC's AEC and Noise Suppression

branches/2.0/doubango/bindings/_common/MediaSessionMgr.cxx

@@ -156,6 +156,11 @@ uint32_t MediaSessionMgr::defaultsGetEchoTail()
 	return tmedia_defaults_get_echo_tail();
 }
 
+bool MediaSessionMgr::defaultsSetEchoSkew(uint32_t echo_skew)
+{
+	return tmedia_defaults_set_echo_skew(echo_skew) == 0;
+}
+
 bool MediaSessionMgr::defaultsSetEchoSuppEnabled(bool echo_supp_enabled)
 {
 	return tmedia_defaults_set_echo_supp_enabled(echo_supp_enabled) == 0;

branches/2.0/doubango/bindings/_common/MediaSessionMgr.h

@@ -60,6 +60,7 @@ public:
 	static tmedia_bandwidth_level_t defaultsGetBandwidthLevel();
 	static bool defaultsSetEchoTail(uint32_t echo_tail);
 	static uint32_t defaultsGetEchoTail();
+	static bool defaultsSetEchoSkew(uint32_t echo_skew);
 	static bool defaultsSetEchoSuppEnabled(bool echo_supp_enabled);
 	static bool defaultsGetEchoSuppEnabled();
 	static bool defaultsSetAgcEnabled(bool agc_enabled);

branches/2.0/doubango/bindings/csharp/MediaSessionMgr.cs

@@ -96,6 +96,11 @@ public class MediaSessionMgr : IDisposable {
     return ret;
   }
 
+  public static bool defaultsSetEchoSkew(uint echo_skew) {
+    bool ret = tinyWRAPPINVOKE.MediaSessionMgr_defaultsSetEchoSkew(echo_skew);
+    return ret;
+  }
+
   public static bool defaultsSetEchoSuppEnabled(bool echo_supp_enabled) {
     bool ret = tinyWRAPPINVOKE.MediaSessionMgr_defaultsSetEchoSuppEnabled(echo_supp_enabled);
     return ret;

branches/2.0/doubango/bindings/csharp/tinyWRAPPINVOKE.cs

@@ -290,6 +290,9 @@ class tinyWRAPPINVOKE {
   [DllImport("tinyWRAP", EntryPoint="CSharp_MediaSessionMgr_defaultsGetEchoTail")]
   public static extern uint MediaSessionMgr_defaultsGetEchoTail();
 
+  [DllImport("tinyWRAP", EntryPoint="CSharp_MediaSessionMgr_defaultsSetEchoSkew")]
+  public static extern bool MediaSessionMgr_defaultsSetEchoSkew(uint jarg1);
+
   [DllImport("tinyWRAP", EntryPoint="CSharp_MediaSessionMgr_defaultsSetEchoSuppEnabled")]
   public static extern bool MediaSessionMgr_defaultsSetEchoSuppEnabled(bool jarg1);
 

branches/2.0/doubango/bindings/csharp/tinyWRAP_wrap.cxx

@@ -1566,6 +1566,18 @@ SWIGEXPORT unsigned int SWIGSTDCALL CSharp_MediaSessionMgr_defaultsGetEchoTail()
 }
 
 
+SWIGEXPORT unsigned int SWIGSTDCALL CSharp_MediaSessionMgr_defaultsSetEchoSkew(unsigned int jarg1) {
+  unsigned int jresult ;
+  uint32_t arg1 ;
+  bool result;
+  
+  arg1 = (uint32_t)jarg1; 
+  result = (bool)MediaSessionMgr::defaultsSetEchoSkew(arg1);
+  jresult = result; 
+  return jresult;
+}
+
+
 SWIGEXPORT unsigned int SWIGSTDCALL CSharp_MediaSessionMgr_defaultsSetEchoSuppEnabled(unsigned int jarg1) {
   unsigned int jresult ;
   bool arg1 ;

branches/2.0/doubango/bindings/java/MediaSessionMgr.java

@@ -79,6 +79,10 @@ public class MediaSessionMgr {
     return tinyWRAPJNI.MediaSessionMgr_defaultsGetEchoTail();
   }
 
+  public static boolean defaultsSetEchoSkew(long echo_skew) {
+    return tinyWRAPJNI.MediaSessionMgr_defaultsSetEchoSkew(echo_skew);
+  }
+
   public static boolean defaultsSetEchoSuppEnabled(boolean echo_supp_enabled) {
     return tinyWRAPJNI.MediaSessionMgr_defaultsSetEchoSuppEnabled(echo_supp_enabled);
   }

branches/2.0/doubango/bindings/java/android/MediaSessionMgr.java

@@ -79,6 +79,10 @@ public class MediaSessionMgr {
     return tinyWRAPJNI.MediaSessionMgr_defaultsGetEchoTail();
   }
 
+  public static boolean defaultsSetEchoSkew(long echo_skew) {
+    return tinyWRAPJNI.MediaSessionMgr_defaultsSetEchoSkew(echo_skew);
+  }
+
   public static boolean defaultsSetEchoSuppEnabled(boolean echo_supp_enabled) {
     return tinyWRAPJNI.MediaSessionMgr_defaultsSetEchoSuppEnabled(echo_supp_enabled);
   }

branches/2.0/doubango/bindings/java/android/commercial.sh

@@ -6,6 +6,6 @@ export COMMERCIAL_PARAMS="LICENSE=lgpl FFMPEG=yes H264=no THEORA=yes OPENCORE_AM
 
 for IS_NEON in no yes
 do
-../bindings/java/android/buildAll.sh $@ NEON=$IS_NEON $COMMERCIAL_PARAMS
+../bindings/java/android/buildAll.sh $@ NEON=$IS_NEON $COMMERCIAL_PARAMS HAVE_WEBRTC=$IS_NEON HAVE_WEBRTC_DENOISE=$IS_NEON
 done
 

branches/2.0/doubango/bindings/java/android/droid-makefile

@@ -42,6 +42,10 @@ ifeq ($(SPEEX), yes)
 SPEEX_LDFLAGS := -lspeex
 endif
 
+ifeq ($(HAVE_WEBRTC), yes)
+	WEBRTC_LDFLAGS := -lwebrtc_aec_$(MARCH) -lwebrtc_apm_utility_$(MARCH) -lwebrtc_ns_$(MARCH) -lwebrtc_system_wrappers_$(MARCH)
+endif
+
 ifeq ($(ILBC), yes)
 ILBC_LDFLAGS := -liLBC
 endif
@@ -68,7 +72,7 @@ else
 	_LT=Bstatic
 endif
 
-LDFLAGS := $(LDFLAGS_LIB) -L$(THIRDPARTIES_LIB) -L$(THIRDPARTIES_MARCH_LIB) $(FFMPEG_LDFLAGS) $(SPEEX_DSP_LDFLAGS) $(SPEEX_LDFLAGS) $(OPENCORE_ARM_LDFLAGS) $(BV16_LDFLAGS) $(ILBC_LDFLAGS) $(LIBGSM_LDFLAGS) $(G729_LDFLAGS)\
+LDFLAGS := $(LDFLAGS_LIB) -L$(THIRDPARTIES_LIB) -L$(THIRDPARTIES_MARCH_LIB) $(FFMPEG_LDFLAGS) $(WEBRTC_LDFLAGS) $(SPEEX_DSP_LDFLAGS) $(SPEEX_LDFLAGS) $(OPENCORE_ARM_LDFLAGS) $(BV16_LDFLAGS) $(ILBC_LDFLAGS) $(LIBGSM_LDFLAGS) $(G729_LDFLAGS)\
 	-Wl,-Bdynamic -lm -lstdc++ -lgcc -llog \
 	-Wl,-$(_LT) -ltinySAK_$(MARCH) -ltinyHTTP_$(MARCH) -ltinyXCAP_$(MARCH) -ltinyIPSec_$(MARCH) -ltinySIGCOMP_$(MARCH) -ltinySMS_$(MARCH) -ltinyNET_$(MARCH) -ltinySDP_$(MARCH) -ltinyRTP_$(MARCH) -ltinyMEDIA_$(MARCH) -ltinyMSRP_$(MARCH) -ltinyDAV_$(MARCH) -ltinySIP_$(MARCH)
 

branches/2.0/doubango/bindings/java/android/gpl.sh

@@ -6,6 +6,6 @@ export GPL_PARAMS="LICENSE=gpl NO_FREE=yes FFMPEG=yes H264=yes THEORA=yes OPENCO
 
 for IS_NEON in no yes
 do
-../bindings/java/android/buildAll.sh $@ NEON=$IS_NEON $GPL_PARAMS
+../bindings/java/android/buildAll.sh $@ NEON=$IS_NEON $GPL_PARAMS HAVE_WEBRTC=$IS_NEON HAVE_WEBRTC_DENOISE=$IS_NEON
 done
 

branches/2.0/doubango/bindings/java/android/lgpl.sh

@@ -6,6 +6,6 @@ export LGPL_PARAMS="LICENSE=lgpl FFMPEG=yes H264=no THEORA=yes OPENCORE_AMR=yes
 
 for IS_NEON in no yes
 do
-../bindings/java/android/buildAll.sh $@ NEON=$IS_NEON $LGPL_PARAMS
+../bindings/java/android/buildAll.sh $@ NEON=$IS_NEON $LGPL_PARAMS HAVE_WEBRTC=$IS_NEON HAVE_WEBRTC_DENOISE=$IS_NEON
 done
 

branches/2.0/doubango/bindings/java/android/minimal.sh

@@ -6,5 +6,5 @@ export MINIMAL_PARAMS="FFMPEG=no H264=no THEORA=no OPENCORE_AMR=no SPEEX_DSP=yes
 
 for IS_NEON in no yes
 do
-../bindings/java/android/buildAll.sh $@ NEON=$IS_NEON $MINIMAL_PARAMS
+../bindings/java/android/buildAll.sh $@ NEON=$IS_NEON $MINIMAL_PARAMS HAVE_WEBRTC=$IS_NEON HAVE_WEBRTC_DENOISE=$IS_NEON
 done

branches/2.0/doubango/bindings/java/android/tinyWRAPJNI.java

@@ -45,6 +45,7 @@ class tinyWRAPJNI {
   public final static native int MediaSessionMgr_defaultsGetBandwidthLevel();
   public final static native boolean MediaSessionMgr_defaultsSetEchoTail(long jarg1);
   public final static native long MediaSessionMgr_defaultsGetEchoTail();
+  public final static native boolean MediaSessionMgr_defaultsSetEchoSkew(long jarg1);
   public final static native boolean MediaSessionMgr_defaultsSetEchoSuppEnabled(boolean jarg1);
   public final static native boolean MediaSessionMgr_defaultsGetEchoSuppEnabled();
   public final static native boolean MediaSessionMgr_defaultsSetAgcEnabled(boolean jarg1);

branches/2.0/doubango/bindings/java/android/tinyWRAP_wrap.cxx

@@ -2464,6 +2464,20 @@ SWIGEXPORT jlong JNICALL Java_org_doubango_tinyWRAP_tinyWRAPJNI_MediaSessionMgr_
 }
 
 
+SWIGEXPORT jboolean JNICALL Java_org_doubango_tinyWRAP_tinyWRAPJNI_MediaSessionMgr_1defaultsSetEchoSkew(JNIEnv *jenv, jclass jcls, jlong jarg1) {
+  jboolean jresult = 0 ;
+  uint32_t arg1 ;
+  bool result;
+  
+  (void)jenv;
+  (void)jcls;
+  arg1 = (uint32_t)jarg1; 
+  result = (bool)MediaSessionMgr::defaultsSetEchoSkew(arg1);
+  jresult = (jboolean)result; 
+  return jresult;
+}
+
+
 SWIGEXPORT jboolean JNICALL Java_org_doubango_tinyWRAP_tinyWRAPJNI_MediaSessionMgr_1defaultsSetEchoSuppEnabled(JNIEnv *jenv, jclass jcls, jboolean jarg1) {
   jboolean jresult = 0 ;
   bool arg1 ;

branches/2.0/doubango/bindings/java/tinyWRAPJNI.java

@@ -45,6 +45,7 @@ class tinyWRAPJNI {
   public final static native int MediaSessionMgr_defaultsGetBandwidthLevel();
   public final static native boolean MediaSessionMgr_defaultsSetEchoTail(long jarg1);
   public final static native long MediaSessionMgr_defaultsGetEchoTail();
+  public final static native boolean MediaSessionMgr_defaultsSetEchoSkew(long jarg1);
   public final static native boolean MediaSessionMgr_defaultsSetEchoSuppEnabled(boolean jarg1);
   public final static native boolean MediaSessionMgr_defaultsGetEchoSuppEnabled();
   public final static native boolean MediaSessionMgr_defaultsSetAgcEnabled(boolean jarg1);

branches/2.0/doubango/bindings/java/tinyWRAP_wrap.cxx

@@ -2464,6 +2464,20 @@ SWIGEXPORT jlong JNICALL Java_org_doubango_tinyWRAP_tinyWRAPJNI_MediaSessionMgr_
 }
 
 
+SWIGEXPORT jboolean JNICALL Java_org_doubango_tinyWRAP_tinyWRAPJNI_MediaSessionMgr_1defaultsSetEchoSkew(JNIEnv *jenv, jclass jcls, jlong jarg1) {
+  jboolean jresult = 0 ;
+  uint32_t arg1 ;
+  bool result;
+  
+  (void)jenv;
+  (void)jcls;
+  arg1 = (uint32_t)jarg1; 
+  result = (bool)MediaSessionMgr::defaultsSetEchoSkew(arg1);
+  jresult = (jboolean)result; 
+  return jresult;
+}
+
+
 SWIGEXPORT jboolean JNICALL Java_org_doubango_tinyWRAP_tinyWRAPJNI_MediaSessionMgr_1defaultsSetEchoSuppEnabled(JNIEnv *jenv, jclass jcls, jboolean jarg1) {
   jboolean jresult = 0 ;
   bool arg1 ;

branches/2.0/doubango/bindings/perl/tinyWRAP.pm

@@ -202,6 +202,7 @@ sub DESTROY {
 *defaultsGetBandwidthLevel = *tinyWRAPc::MediaSessionMgr_defaultsGetBandwidthLevel;
 *defaultsSetEchoTail = *tinyWRAPc::MediaSessionMgr_defaultsSetEchoTail;
 *defaultsGetEchoTail = *tinyWRAPc::MediaSessionMgr_defaultsGetEchoTail;
+*defaultsSetEchoSkew = *tinyWRAPc::MediaSessionMgr_defaultsSetEchoSkew;
 *defaultsSetEchoSuppEnabled = *tinyWRAPc::MediaSessionMgr_defaultsSetEchoSuppEnabled;
 *defaultsGetEchoSuppEnabled = *tinyWRAPc::MediaSessionMgr_defaultsGetEchoSuppEnabled;
 *defaultsSetAgcEnabled = *tinyWRAPc::MediaSessionMgr_defaultsSetAgcEnabled;

branches/2.0/doubango/bindings/perl/tinyWRAP_wrap.cxx

@@ -3455,6 +3455,34 @@ XS(_wrap_MediaSessionMgr_defaultsGetEchoTail) {
 }
 
 
+XS(_wrap_MediaSessionMgr_defaultsSetEchoSkew) {
+  {
+    uint32_t arg1 ;
+    unsigned int val1 ;
+    int ecode1 = 0 ;
+    int argvi = 0;
+    bool result;
+    dXSARGS;
+    
+    if ((items < 1) || (items > 1)) {
+      SWIG_croak("Usage: MediaSessionMgr_defaultsSetEchoSkew(echo_skew);");
+    }
+    ecode1 = SWIG_AsVal_unsigned_SS_int SWIG_PERL_CALL_ARGS_2(ST(0), &val1);
+    if (!SWIG_IsOK(ecode1)) {
+      SWIG_exception_fail(SWIG_ArgError(ecode1), "in method '" "MediaSessionMgr_defaultsSetEchoSkew" "', argument " "1"" of type '" "uint32_t""'");
+    } 
+    arg1 = static_cast< uint32_t >(val1);
+    result = (bool)MediaSessionMgr::defaultsSetEchoSkew(arg1);
+    ST(argvi) = SWIG_From_bool  SWIG_PERL_CALL_ARGS_1(static_cast< bool >(result)); argvi++ ;
+    
+    XSRETURN(argvi);
+  fail:
+    
+    SWIG_croak_null();
+  }
+}
+
+
 XS(_wrap_MediaSessionMgr_defaultsSetEchoSuppEnabled) {
   {
     bool arg1 ;
@@ -21269,6 +21297,7 @@ static swig_command_info swig_commands[] = {
 {"tinyWRAPc::MediaSessionMgr_defaultsGetBandwidthLevel", _wrap_MediaSessionMgr_defaultsGetBandwidthLevel},
 {"tinyWRAPc::MediaSessionMgr_defaultsSetEchoTail", _wrap_MediaSessionMgr_defaultsSetEchoTail},
 {"tinyWRAPc::MediaSessionMgr_defaultsGetEchoTail", _wrap_MediaSessionMgr_defaultsGetEchoTail},
+{"tinyWRAPc::MediaSessionMgr_defaultsSetEchoSkew", _wrap_MediaSessionMgr_defaultsSetEchoSkew},
 {"tinyWRAPc::MediaSessionMgr_defaultsSetEchoSuppEnabled", _wrap_MediaSessionMgr_defaultsSetEchoSuppEnabled},
 {"tinyWRAPc::MediaSessionMgr_defaultsGetEchoSuppEnabled", _wrap_MediaSessionMgr_defaultsGetEchoSuppEnabled},
 {"tinyWRAPc::MediaSessionMgr_defaultsSetAgcEnabled", _wrap_MediaSessionMgr_defaultsSetAgcEnabled},

branches/2.0/doubango/bindings/python/tinyWRAP.py

@@ -161,6 +161,8 @@ class MediaSessionMgr(_object):
     if _newclass:defaultsSetEchoTail = staticmethod(_tinyWRAP.MediaSessionMgr_defaultsSetEchoTail)
     __swig_getmethods__["defaultsGetEchoTail"] = lambda x: _tinyWRAP.MediaSessionMgr_defaultsGetEchoTail
     if _newclass:defaultsGetEchoTail = staticmethod(_tinyWRAP.MediaSessionMgr_defaultsGetEchoTail)
+    __swig_getmethods__["defaultsSetEchoSkew"] = lambda x: _tinyWRAP.MediaSessionMgr_defaultsSetEchoSkew
+    if _newclass:defaultsSetEchoSkew = staticmethod(_tinyWRAP.MediaSessionMgr_defaultsSetEchoSkew)
     __swig_getmethods__["defaultsSetEchoSuppEnabled"] = lambda x: _tinyWRAP.MediaSessionMgr_defaultsSetEchoSuppEnabled
     if _newclass:defaultsSetEchoSuppEnabled = staticmethod(_tinyWRAP.MediaSessionMgr_defaultsSetEchoSuppEnabled)
     __swig_getmethods__["defaultsGetEchoSuppEnabled"] = lambda x: _tinyWRAP.MediaSessionMgr_defaultsGetEchoSuppEnabled
@@ -210,6 +212,10 @@ def MediaSessionMgr_defaultsGetEchoTail():
   return _tinyWRAP.MediaSessionMgr_defaultsGetEchoTail()
 MediaSessionMgr_defaultsGetEchoTail = _tinyWRAP.MediaSessionMgr_defaultsGetEchoTail
 
+def MediaSessionMgr_defaultsSetEchoSkew(*args):
+  return _tinyWRAP.MediaSessionMgr_defaultsSetEchoSkew(*args)
+MediaSessionMgr_defaultsSetEchoSkew = _tinyWRAP.MediaSessionMgr_defaultsSetEchoSkew
+
 def MediaSessionMgr_defaultsSetEchoSuppEnabled(*args):
   return _tinyWRAP.MediaSessionMgr_defaultsSetEchoSuppEnabled(*args)
 MediaSessionMgr_defaultsSetEchoSuppEnabled = _tinyWRAP.MediaSessionMgr_defaultsSetEchoSuppEnabled

branches/2.0/doubango/bindings/python/tinyWRAP_wrap.cxx

@@ -6159,6 +6159,28 @@ fail:
 }
 
 
+SWIGINTERN PyObject *_wrap_MediaSessionMgr_defaultsSetEchoSkew(PyObject *SWIGUNUSEDPARM(self), PyObject *args) {
+  PyObject *resultobj = 0;
+  uint32_t arg1 ;
+  unsigned int val1 ;
+  int ecode1 = 0 ;
+  PyObject * obj0 = 0 ;
+  bool result;
+  
+  if (!PyArg_ParseTuple(args,(char *)"O:MediaSessionMgr_defaultsSetEchoSkew",&obj0)) SWIG_fail;
+  ecode1 = SWIG_AsVal_unsigned_SS_int(obj0, &val1);
+  if (!SWIG_IsOK(ecode1)) {
+    SWIG_exception_fail(SWIG_ArgError(ecode1), "in method '" "MediaSessionMgr_defaultsSetEchoSkew" "', argument " "1"" of type '" "uint32_t""'");
+  } 
+  arg1 = static_cast< uint32_t >(val1);
+  result = (bool)MediaSessionMgr::defaultsSetEchoSkew(arg1);
+  resultobj = SWIG_From_bool(static_cast< bool >(result));
+  return resultobj;
+fail:
+  return NULL;
+}
+
+
 SWIGINTERN PyObject *_wrap_MediaSessionMgr_defaultsSetEchoSuppEnabled(PyObject *SWIGUNUSEDPARM(self), PyObject *args) {
   PyObject *resultobj = 0;
   bool arg1 ;
@@ -20763,6 +20785,7 @@ static PyMethodDef SwigMethods[] = {
 	 { (char *)"MediaSessionMgr_defaultsGetBandwidthLevel", _wrap_MediaSessionMgr_defaultsGetBandwidthLevel, METH_VARARGS, NULL},
 	 { (char *)"MediaSessionMgr_defaultsSetEchoTail", _wrap_MediaSessionMgr_defaultsSetEchoTail, METH_VARARGS, NULL},
 	 { (char *)"MediaSessionMgr_defaultsGetEchoTail", _wrap_MediaSessionMgr_defaultsGetEchoTail, METH_VARARGS, NULL},
+	 { (char *)"MediaSessionMgr_defaultsSetEchoSkew", _wrap_MediaSessionMgr_defaultsSetEchoSkew, METH_VARARGS, NULL},
 	 { (char *)"MediaSessionMgr_defaultsSetEchoSuppEnabled", _wrap_MediaSessionMgr_defaultsSetEchoSuppEnabled, METH_VARARGS, NULL},
 	 { (char *)"MediaSessionMgr_defaultsGetEchoSuppEnabled", _wrap_MediaSessionMgr_defaultsGetEchoSuppEnabled, METH_VARARGS, NULL},
 	 { (char *)"MediaSessionMgr_defaultsSetAgcEnabled", _wrap_MediaSessionMgr_defaultsSetAgcEnabled, METH_VARARGS, NULL},

branches/2.0/doubango/thirdparties/android/include/webrtc/echo_cancellation.h

@@ -0,0 +1,260 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_
+
+#include "typedefs.h"
+
+// Errors
+#define AEC_UNSPECIFIED_ERROR           12000
+#define AEC_UNSUPPORTED_FUNCTION_ERROR  12001
+#define AEC_UNINITIALIZED_ERROR         12002
+#define AEC_NULL_POINTER_ERROR          12003
+#define AEC_BAD_PARAMETER_ERROR         12004
+
+// Warnings
+#define AEC_BAD_PARAMETER_WARNING       12050
+
+enum {
+    kAecNlpConservative = 0,
+    kAecNlpModerate,
+    kAecNlpAggressive
+};
+
+enum {
+    kAecFalse = 0,
+    kAecTrue
+};
+
+typedef struct {
+    WebRtc_Word16 nlpMode;        // default kAecNlpModerate
+    WebRtc_Word16 skewMode;       // default kAecFalse
+    WebRtc_Word16 metricsMode;    // default kAecFalse
+    //float realSkew;
+} AecConfig;
+
+typedef struct {
+    WebRtc_Word16 instant;
+    WebRtc_Word16 average;
+    WebRtc_Word16 max;
+    WebRtc_Word16 min;
+} AecLevel;
+
+typedef struct {
+    AecLevel rerl;
+    AecLevel erl;
+    AecLevel erle;
+    AecLevel aNlp;
+} AecMetrics;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * Allocates the memory needed by the AEC. The memory needs to be initialized
+ * separately using the WebRtcAec_Init() function.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void **aecInst               Pointer to the AEC instance to be created
+ *                              and initilized
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return          0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_Create(void **aecInst);
+
+/*
+ * This function releases the memory allocated by WebRtcAec_Create().
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void         *aecInst        Pointer to the AEC instance
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_Free(void *aecInst);
+
+/*
+ * Initializes an AEC instance.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ * WebRtc_Word32  sampFreq      Sampling frequency of data
+ * WebRtc_Word32  scSampFreq    Soundcard sampling frequency
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return          0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_Init(void *aecInst,
+                             WebRtc_Word32 sampFreq,
+                             WebRtc_Word32 scSampFreq);
+
+/*
+ * Inserts an 80 or 160 sample block of data into the farend buffer.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ * WebRtc_Word16  *farend       In buffer containing one frame of
+ *                              farend signal for L band
+ * WebRtc_Word16  nrOfSamples   Number of samples in farend buffer
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst,
+                                     const WebRtc_Word16 *farend,
+                                     WebRtc_Word16 nrOfSamples);
+
+/*
+ * Runs the echo canceller on an 80 or 160 sample blocks of data.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void          *aecInst       Pointer to the AEC instance
+ * WebRtc_Word16 *nearend       In buffer containing one frame of
+ *                              nearend+echo signal for L band
+ * WebRtc_Word16 *nearendH      In buffer containing one frame of
+ *                              nearend+echo signal for H band
+ * WebRtc_Word16 nrOfSamples    Number of samples in nearend buffer
+ * WebRtc_Word16 msInSndCardBuf Delay estimate for sound card and
+ *                              system buffers
+ * WebRtc_Word16 skew           Difference between number of samples played
+ *                              and recorded at the soundcard (for clock skew
+ *                              compensation)
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word16  *out          Out buffer, one frame of processed nearend
+ *                              for L band
+ * WebRtc_Word16  *outH         Out buffer, one frame of processed nearend
+ *                              for H band
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_Process(void *aecInst,
+                                const WebRtc_Word16 *nearend,
+                                const WebRtc_Word16 *nearendH,
+                                WebRtc_Word16 *out,
+                                WebRtc_Word16 *outH,
+                                WebRtc_Word16 nrOfSamples,
+                                WebRtc_Word16 msInSndCardBuf,
+                                WebRtc_Word32 skew);
+
+/*
+ * This function enables the user to set certain parameters on-the-fly.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ * AecConfig      config        Config instance that contains all
+ *                              properties to be set
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config);
+
+/*
+ * Gets the on-the-fly paramters.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * AecConfig      *config       Pointer to the config instance that
+ *                              all properties will be written to
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config);
+
+/*
+ * Gets the current echo status of the nearend signal.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word16  *status       0: Almost certainly nearend single-talk
+ *                              1: Might not be neared single-talk
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status);
+
+/*
+ * Gets the current echo metrics for the session.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * AecMetrics     *metrics      Struct which will be filled out with the
+ *                              current echo metrics.
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics);
+
+/*
+ * Gets the last error code.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32  return        11000-11100: error code
+ */
+WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst);
+
+/*
+ * Gets a version string.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * char           *versionStr   Pointer to a string array
+ * WebRtc_Word16  len           The maximum length of the string
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word8   *versionStr   Pointer to a string array
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len);
+
+#ifdef __cplusplus
+}
+#endif
+#endif  /* WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_ */

branches/2.0/doubango/thirdparties/android/include/webrtc/noise_suppression.h

@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_
+
+#include "typedefs.h"
+
+typedef struct NsHandleT NsHandle;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * This function returns the version number of the code.
+ *
+ * Input:
+ *      - version       : Pointer to a character array where the version
+ *                        info is stored.
+ *      - length        : Length of version.
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error (probably length is not sufficient)
+ */
+int WebRtcNs_get_version(char *version, short length);
+
+
+/*
+ * This function creates an instance to the noise reduction structure
+ *
+ * Input:
+ *      - NS_inst       : Pointer to noise reduction instance that should be
+ *                        created
+ *
+ * Output:
+ *      - NS_inst       : Pointer to created noise reduction instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNs_Create(NsHandle **NS_inst);
+
+
+/*
+ * This function frees the dynamic memory of a specified Noise Reduction
+ * instance.
+ *
+ * Input:
+ *      - NS_inst       : Pointer to NS instance that should be freed
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNs_Free(NsHandle *NS_inst);
+
+
+/*
+ * This function initializes a NS instance
+ *
+ * Input:
+ *      - NS_inst       : Instance that should be initialized
+ *      - fs            : sampling frequency
+ *
+ * Output:
+ *      - NS_inst       : Initialized instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNs_Init(NsHandle *NS_inst, WebRtc_UWord32 fs);
+
+/*
+ * This changes the aggressiveness of the noise suppression method.
+ *
+ * Input:
+ *      - NS_inst       : Instance that should be initialized
+ *      - mode          : 0: Mild, 1: Medium , 2: Aggressive
+ *
+ * Output:
+ *      - NS_inst       : Initialized instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNs_set_policy(NsHandle *NS_inst, int mode);
+
+
+/*
+ * This functions does Noise Suppression for the inserted speech frame. The
+ * input and output signals should always be 10ms (80 or 160 samples).
+ *
+ * Input
+ *      - NS_inst       : VAD Instance. Needs to be initiated before call.
+ *      - spframe       : Pointer to speech frame buffer for L band
+ *      - spframe_H     : Pointer to speech frame buffer for H band
+ *      - fs            : sampling frequency
+ *
+ * Output:
+ *      - NS_inst       : Updated NS instance
+ *      - outframe      : Pointer to output frame for L band
+ *      - outframe_H    : Pointer to output frame for H band
+ *
+ * Return value         :  0 - OK
+ *                        -1 - Error
+ */
+int WebRtcNs_Process(NsHandle *NS_inst,
+                     short *spframe,
+                     short *spframe_H,
+                     short *outframe,
+                     short *outframe_H);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_

branches/2.0/doubango/thirdparties/android/include/webrtc/noise_suppression_x.h

@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
+
+#include "signal_processing_library.h"
+
+typedef struct NsxHandleT NsxHandle;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * This function returns the version number of the code.
+ *
+ * Input:
+ *      - version       : Pointer to a character array where the version
+ *                        info is stored.
+ *      - length        : Length of version.
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error (probably length is not sufficient)
+ */
+int WebRtcNsx_get_version(char *version, short length);
+
+
+/*
+ * This function creates an instance to the noise reduction structure
+ *
+ * Input:
+ *      - nsxInst       : Pointer to noise reduction instance that should be
+ *                       created
+ *
+ * Output:
+ *      - nsxInst       : Pointer to created noise reduction instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNsx_Create(NsxHandle **nsxInst);
+
+
+/*
+ * This function frees the dynamic memory of a specified Noise Suppression
+ * instance.
+ *
+ * Input:
+ *      - nsxInst       : Pointer to NS instance that should be freed
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNsx_Free(NsxHandle *nsxInst);
+
+
+/*
+ * This function initializes a NS instance
+ *
+ * Input:
+ *      - nsxInst       : Instance that should be initialized
+ *      - fs            : sampling frequency
+ *
+ * Output:
+ *      - nsxInst       : Initialized instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNsx_Init(NsxHandle *nsxInst, WebRtc_UWord32 fs);
+
+/*
+ * This changes the aggressiveness of the noise suppression method.
+ *
+ * Input:
+ *      - nsxInst       : Instance that should be initialized
+ *      - mode          : 0: Mild, 1: Medium , 2: Aggressive
+ *
+ * Output:
+ *      - nsxInst       : Initialized instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNsx_set_policy(NsxHandle *nsxInst, int mode);
+
+/*
+ * This functions does noise suppression for the inserted speech frame. The
+ * input and output signals should always be 10ms (80 or 160 samples).
+ *
+ * Input
+ *      - nsxInst       : NSx instance. Needs to be initiated before call.
+ *      - speechFrame   : Pointer to speech frame buffer for L band
+ *      - speechFrameHB : Pointer to speech frame buffer for H band
+ *      - fs            : sampling frequency
+ *
+ * Output:
+ *      - nsxInst       : Updated NSx instance
+ *      - outFrame      : Pointer to output frame for L band
+ *      - outFrameHB    : Pointer to output frame for H band
+ *
+ * Return value         :  0 - OK
+ *                        -1 - Error
+ */
+int WebRtcNsx_Process(NsxHandle *nsxInst,
+                      short *speechFrame,
+                      short *speechFrameHB,
+                      short *outFrame,
+                      short *outFrameHB);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_

branches/2.0/doubango/thirdparties/android/include/webrtc/typedefs.h

@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ *
+ * This file contains type definitions used in all WebRtc APIs.
+ * 
+ */
+
+/* Reserved words definitions */
+#define WEBRTC_EXTERN extern
+#define G_CONST const
+#define WEBRTC_INLINE extern __inline
+
+#ifndef WEBRTC_TYPEDEFS_H
+#define WEBRTC_TYPEDEFS_H
+
+/* Define WebRtc preprocessor identifiers based on the current build platform */
+#if defined(WIN32)
+    // Windows & Windows Mobile
+    #if !defined(WEBRTC_TARGET_PC)
+        #define WEBRTC_TARGET_PC
+    #endif
+#elif defined(__APPLE__)
+    // Mac OS X
+    #if defined(__LITTLE_ENDIAN__ ) //TODO: is this used?
+        #if !defined(WEBRTC_TARGET_MAC_INTEL)
+            #define WEBRTC_TARGET_MAC_INTEL
+        #endif  
+    #else
+        #if !defined(WEBRTC_TARGET_MAC)
+            #define WEBRTC_TARGET_MAC
+        #endif  
+    #endif
+#else
+    // Linux etc.
+    #if !defined(WEBRTC_TARGET_PC)
+        #define WEBRTC_TARGET_PC
+    #endif
+#endif
+
+#if defined(WEBRTC_TARGET_PC)
+
+#if !defined(_MSC_VER)
+  #include <stdint.h>
+#else
+    // Define C99 equivalent types.
+    // Since MSVC doesn't include these headers, we have to write our own
+    // version to provide a compatibility layer between MSVC and the WebRTC
+    // headers.
+    typedef signed char         int8_t;
+    typedef signed short        int16_t;
+    typedef signed int          int32_t;
+    typedef signed long long    int64_t;
+    typedef unsigned char       uint8_t;
+    typedef unsigned short      uint16_t;
+    typedef unsigned int        uint32_t;
+    typedef unsigned long long  uint64_t;
+#endif
+
+#if defined(WIN32)
+    typedef __int64             WebRtc_Word64;
+    typedef unsigned __int64    WebRtc_UWord64;
+#else
+    typedef int64_t             WebRtc_Word64;
+    typedef uint64_t            WebRtc_UWord64;
+#endif
+    typedef int32_t             WebRtc_Word32;
+    typedef uint32_t            WebRtc_UWord32;
+    typedef int16_t             WebRtc_Word16;
+    typedef uint16_t            WebRtc_UWord16;
+    typedef char                WebRtc_Word8;
+    typedef uint8_t             WebRtc_UWord8;
+
+    /* Define endian for the platform */
+    #define WEBRTC_LITTLE_ENDIAN
+
+#elif defined(WEBRTC_TARGET_MAC_INTEL)
+    #include <stdint.h>
+
+    typedef int64_t             WebRtc_Word64;
+    typedef uint64_t            WebRtc_UWord64;
+    typedef int32_t             WebRtc_Word32;
+    typedef uint32_t            WebRtc_UWord32;
+    typedef int16_t             WebRtc_Word16;
+    typedef char                WebRtc_Word8;
+    typedef uint16_t            WebRtc_UWord16;
+    typedef uint8_t             WebRtc_UWord8;
+
+    /* Define endian for the platform */
+    #define WEBRTC_LITTLE_ENDIAN
+
+#else
+
+    #error "No platform defined for WebRtc type definitions (webrtc_typedefs.h)"
+
+#endif
+
+
+#endif // WEBRTC_TYPEDEFS_H

branches/2.0/doubango/thirdparties/win32/include/webrtc/echo_cancellation.h

@@ -0,0 +1,260 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_
+
+#include "typedefs.h"
+
+// Errors
+#define AEC_UNSPECIFIED_ERROR           12000
+#define AEC_UNSUPPORTED_FUNCTION_ERROR  12001
+#define AEC_UNINITIALIZED_ERROR         12002
+#define AEC_NULL_POINTER_ERROR          12003
+#define AEC_BAD_PARAMETER_ERROR         12004
+
+// Warnings
+#define AEC_BAD_PARAMETER_WARNING       12050
+
+enum {
+    kAecNlpConservative = 0,
+    kAecNlpModerate,
+    kAecNlpAggressive
+};
+
+enum {
+    kAecFalse = 0,
+    kAecTrue
+};
+
+typedef struct {
+    WebRtc_Word16 nlpMode;        // default kAecNlpModerate
+    WebRtc_Word16 skewMode;       // default kAecFalse
+    WebRtc_Word16 metricsMode;    // default kAecFalse
+    //float realSkew;
+} AecConfig;
+
+typedef struct {
+    WebRtc_Word16 instant;
+    WebRtc_Word16 average;
+    WebRtc_Word16 max;
+    WebRtc_Word16 min;
+} AecLevel;
+
+typedef struct {
+    AecLevel rerl;
+    AecLevel erl;
+    AecLevel erle;
+    AecLevel aNlp;
+} AecMetrics;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * Allocates the memory needed by the AEC. The memory needs to be initialized
+ * separately using the WebRtcAec_Init() function.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void **aecInst               Pointer to the AEC instance to be created
+ *                              and initilized
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return          0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_Create(void **aecInst);
+
+/*
+ * This function releases the memory allocated by WebRtcAec_Create().
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void         *aecInst        Pointer to the AEC instance
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_Free(void *aecInst);
+
+/*
+ * Initializes an AEC instance.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ * WebRtc_Word32  sampFreq      Sampling frequency of data
+ * WebRtc_Word32  scSampFreq    Soundcard sampling frequency
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32 return          0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_Init(void *aecInst,
+                             WebRtc_Word32 sampFreq,
+                             WebRtc_Word32 scSampFreq);
+
+/*
+ * Inserts an 80 or 160 sample block of data into the farend buffer.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ * WebRtc_Word16  *farend       In buffer containing one frame of
+ *                              farend signal for L band
+ * WebRtc_Word16  nrOfSamples   Number of samples in farend buffer
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst,
+                                     const WebRtc_Word16 *farend,
+                                     WebRtc_Word16 nrOfSamples);
+
+/*
+ * Runs the echo canceller on an 80 or 160 sample blocks of data.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void          *aecInst       Pointer to the AEC instance
+ * WebRtc_Word16 *nearend       In buffer containing one frame of
+ *                              nearend+echo signal for L band
+ * WebRtc_Word16 *nearendH      In buffer containing one frame of
+ *                              nearend+echo signal for H band
+ * WebRtc_Word16 nrOfSamples    Number of samples in nearend buffer
+ * WebRtc_Word16 msInSndCardBuf Delay estimate for sound card and
+ *                              system buffers
+ * WebRtc_Word16 skew           Difference between number of samples played
+ *                              and recorded at the soundcard (for clock skew
+ *                              compensation)
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word16  *out          Out buffer, one frame of processed nearend
+ *                              for L band
+ * WebRtc_Word16  *outH         Out buffer, one frame of processed nearend
+ *                              for H band
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_Process(void *aecInst,
+                                const WebRtc_Word16 *nearend,
+                                const WebRtc_Word16 *nearendH,
+                                WebRtc_Word16 *out,
+                                WebRtc_Word16 *outH,
+                                WebRtc_Word16 nrOfSamples,
+                                WebRtc_Word16 msInSndCardBuf,
+                                WebRtc_Word32 skew);
+
+/*
+ * This function enables the user to set certain parameters on-the-fly.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ * AecConfig      config        Config instance that contains all
+ *                              properties to be set
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config);
+
+/*
+ * Gets the on-the-fly paramters.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * AecConfig      *config       Pointer to the config instance that
+ *                              all properties will be written to
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config);
+
+/*
+ * Gets the current echo status of the nearend signal.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word16  *status       0: Almost certainly nearend single-talk
+ *                              1: Might not be neared single-talk
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status);
+
+/*
+ * Gets the current echo metrics for the session.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * AecMetrics     *metrics      Struct which will be filled out with the
+ *                              current echo metrics.
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics);
+
+/*
+ * Gets the last error code.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * void           *aecInst      Pointer to the AEC instance
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word32  return        11000-11100: error code
+ */
+WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst);
+
+/*
+ * Gets a version string.
+ *
+ * Inputs                       Description
+ * -------------------------------------------------------------------
+ * char           *versionStr   Pointer to a string array
+ * WebRtc_Word16  len           The maximum length of the string
+ *
+ * Outputs                      Description
+ * -------------------------------------------------------------------
+ * WebRtc_Word8   *versionStr   Pointer to a string array
+ * WebRtc_Word32  return         0: OK
+ *                              -1: error
+ */
+WebRtc_Word32 WebRtcAec_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len);
+
+#ifdef __cplusplus
+}
+#endif
+#endif  /* WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_ */

branches/2.0/doubango/thirdparties/win32/include/webrtc/noise_suppression.h

@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_
+
+#include "typedefs.h"
+
+typedef struct NsHandleT NsHandle;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * This function returns the version number of the code.
+ *
+ * Input:
+ *      - version       : Pointer to a character array where the version
+ *                        info is stored.
+ *      - length        : Length of version.
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error (probably length is not sufficient)
+ */
+int WebRtcNs_get_version(char *version, short length);
+
+
+/*
+ * This function creates an instance to the noise reduction structure
+ *
+ * Input:
+ *      - NS_inst       : Pointer to noise reduction instance that should be
+ *                        created
+ *
+ * Output:
+ *      - NS_inst       : Pointer to created noise reduction instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNs_Create(NsHandle **NS_inst);
+
+
+/*
+ * This function frees the dynamic memory of a specified Noise Reduction
+ * instance.
+ *
+ * Input:
+ *      - NS_inst       : Pointer to NS instance that should be freed
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNs_Free(NsHandle *NS_inst);
+
+
+/*
+ * This function initializes a NS instance
+ *
+ * Input:
+ *      - NS_inst       : Instance that should be initialized
+ *      - fs            : sampling frequency
+ *
+ * Output:
+ *      - NS_inst       : Initialized instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNs_Init(NsHandle *NS_inst, WebRtc_UWord32 fs);
+
+/*
+ * This changes the aggressiveness of the noise suppression method.
+ *
+ * Input:
+ *      - NS_inst       : Instance that should be initialized
+ *      - mode          : 0: Mild, 1: Medium , 2: Aggressive
+ *
+ * Output:
+ *      - NS_inst       : Initialized instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNs_set_policy(NsHandle *NS_inst, int mode);
+
+
+/*
+ * This functions does Noise Suppression for the inserted speech frame. The
+ * input and output signals should always be 10ms (80 or 160 samples).
+ *
+ * Input
+ *      - NS_inst       : VAD Instance. Needs to be initiated before call.
+ *      - spframe       : Pointer to speech frame buffer for L band
+ *      - spframe_H     : Pointer to speech frame buffer for H band
+ *      - fs            : sampling frequency
+ *
+ * Output:
+ *      - NS_inst       : Updated NS instance
+ *      - outframe      : Pointer to output frame for L band
+ *      - outframe_H    : Pointer to output frame for H band
+ *
+ * Return value         :  0 - OK
+ *                        -1 - Error
+ */
+int WebRtcNs_Process(NsHandle *NS_inst,
+                     short *spframe,
+                     short *spframe_H,
+                     short *outframe,
+                     short *outframe_H);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_

branches/2.0/doubango/thirdparties/win32/include/webrtc/noise_suppression_x.h

@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
+
+#include "signal_processing_library.h"
+
+typedef struct NsxHandleT NsxHandle;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * This function returns the version number of the code.
+ *
+ * Input:
+ *      - version       : Pointer to a character array where the version
+ *                        info is stored.
+ *      - length        : Length of version.
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error (probably length is not sufficient)
+ */
+int WebRtcNsx_get_version(char *version, short length);
+
+
+/*
+ * This function creates an instance to the noise reduction structure
+ *
+ * Input:
+ *      - nsxInst       : Pointer to noise reduction instance that should be
+ *                       created
+ *
+ * Output:
+ *      - nsxInst       : Pointer to created noise reduction instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNsx_Create(NsxHandle **nsxInst);
+
+
+/*
+ * This function frees the dynamic memory of a specified Noise Suppression
+ * instance.
+ *
+ * Input:
+ *      - nsxInst       : Pointer to NS instance that should be freed
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNsx_Free(NsxHandle *nsxInst);
+
+
+/*
+ * This function initializes a NS instance
+ *
+ * Input:
+ *      - nsxInst       : Instance that should be initialized
+ *      - fs            : sampling frequency
+ *
+ * Output:
+ *      - nsxInst       : Initialized instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNsx_Init(NsxHandle *nsxInst, WebRtc_UWord32 fs);
+
+/*
+ * This changes the aggressiveness of the noise suppression method.
+ *
+ * Input:
+ *      - nsxInst       : Instance that should be initialized
+ *      - mode          : 0: Mild, 1: Medium , 2: Aggressive
+ *
+ * Output:
+ *      - nsxInst       : Initialized instance
+ *
+ * Return value         :  0 - Ok
+ *                        -1 - Error
+ */
+int WebRtcNsx_set_policy(NsxHandle *nsxInst, int mode);
+
+/*
+ * This functions does noise suppression for the inserted speech frame. The
+ * input and output signals should always be 10ms (80 or 160 samples).
+ *
+ * Input
+ *      - nsxInst       : NSx instance. Needs to be initiated before call.
+ *      - speechFrame   : Pointer to speech frame buffer for L band
+ *      - speechFrameHB : Pointer to speech frame buffer for H band
+ *      - fs            : sampling frequency
+ *
+ * Output:
+ *      - nsxInst       : Updated NSx instance
+ *      - outFrame      : Pointer to output frame for L band
+ *      - outFrameHB    : Pointer to output frame for H band
+ *
+ * Return value         :  0 - OK
+ *                        -1 - Error
+ */
+int WebRtcNsx_Process(NsxHandle *nsxInst,
+                      short *speechFrame,
+                      short *speechFrameHB,
+                      short *outFrame,
+                      short *outFrameHB);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_

branches/2.0/doubango/thirdparties/win32/include/webrtc/typedefs.h

@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ *
+ * This file contains type definitions used in all WebRtc APIs.
+ * 
+ */
+
+/* Reserved words definitions */
+#define WEBRTC_EXTERN extern
+#define G_CONST const
+#define WEBRTC_INLINE extern __inline
+
+#ifndef WEBRTC_TYPEDEFS_H
+#define WEBRTC_TYPEDEFS_H
+
+/* Define WebRtc preprocessor identifiers based on the current build platform */
+#if defined(WIN32)
+    // Windows & Windows Mobile
+    #if !defined(WEBRTC_TARGET_PC)
+        #define WEBRTC_TARGET_PC
+    #endif
+#elif defined(__APPLE__)
+    // Mac OS X
+    #if defined(__LITTLE_ENDIAN__ ) //TODO: is this used?
+        #if !defined(WEBRTC_TARGET_MAC_INTEL)
+            #define WEBRTC_TARGET_MAC_INTEL
+        #endif  
+    #else
+        #if !defined(WEBRTC_TARGET_MAC)
+            #define WEBRTC_TARGET_MAC
+        #endif  
+    #endif
+#else
+    // Linux etc.
+    #if !defined(WEBRTC_TARGET_PC)
+        #define WEBRTC_TARGET_PC
+    #endif
+#endif
+
+#if defined(WEBRTC_TARGET_PC)
+
+#if !defined(_MSC_VER)
+  #include <stdint.h>
+#else
+    // Define C99 equivalent types.
+    // Since MSVC doesn't include these headers, we have to write our own
+    // version to provide a compatibility layer between MSVC and the WebRTC
+    // headers.
+    typedef signed char         int8_t;
+    typedef signed short        int16_t;
+    typedef signed int          int32_t;
+    typedef signed long long    int64_t;
+    typedef unsigned char       uint8_t;
+    typedef unsigned short      uint16_t;
+    typedef unsigned int        uint32_t;
+    typedef unsigned long long  uint64_t;
+#endif
+
+#if defined(WIN32)
+    typedef __int64             WebRtc_Word64;
+    typedef unsigned __int64    WebRtc_UWord64;
+#else
+    typedef int64_t             WebRtc_Word64;
+    typedef uint64_t            WebRtc_UWord64;
+#endif
+    typedef int32_t             WebRtc_Word32;
+    typedef uint32_t            WebRtc_UWord32;
+    typedef int16_t             WebRtc_Word16;
+    typedef uint16_t            WebRtc_UWord16;
+    typedef char                WebRtc_Word8;
+    typedef uint8_t             WebRtc_UWord8;
+
+    /* Define endian for the platform */
+    #define WEBRTC_LITTLE_ENDIAN
+
+#elif defined(WEBRTC_TARGET_MAC_INTEL)
+    #include <stdint.h>
+
+    typedef int64_t             WebRtc_Word64;
+    typedef uint64_t            WebRtc_UWord64;
+    typedef int32_t             WebRtc_Word32;
+    typedef uint32_t            WebRtc_UWord32;
+    typedef int16_t             WebRtc_Word16;
+    typedef char                WebRtc_Word8;
+    typedef uint16_t            WebRtc_UWord16;
+    typedef uint8_t             WebRtc_UWord8;
+
+    /* Define endian for the platform */
+    #define WEBRTC_LITTLE_ENDIAN
+
+#else
+
+    #error "No platform defined for WebRtc type definitions (webrtc_typedefs.h)"
+
+#endif
+
+
+#endif // WEBRTC_TYPEDEFS_H

branches/2.0/doubango/tinyDAV/droid-makefile

@@ -82,6 +82,18 @@ else
 	SPEEX_CFLAGS := -DHAVE_LIB_SPEEX=0
 endif
 
+# Google WebRTC (Default: enabled)
+ifeq ($(HAVE_WEBRTC), yes)
+	WEBRTC_CFLAGS := -DHAVE_WEBRTC=1
+	WEBRTC_LDFLAGS := -lwebrtc_aec_$(MARCH) -lwebrtc_apm_utility_$(MARCH) -lwebrtc_ns_$(MARCH) -lwebrtc_system_wrappers_$(MARCH)
+else
+	WEBRTC_CFLAGS := -DHAVE_WEBRTC=0
+endif
+
+ifeq ($(HAVE_WEBRTC_DENOISE), yes)
+	WEBRTC_CFLAGS += -DHAVE_WEBRTC_DENOISE=1
+endif
+
 
 # ILBC (Default: disabled)
 ifeq ($(ILBC), yes)
@@ -122,12 +134,12 @@ endif
 ################################
 
 CFLAGS := $(CFLAGS_LIB) -I$(THIRDPARTIES_INC) $(ILBC_CFLAGS) $(LIBGSM_CFLAGS) \
-	$(FFMPEG_CFLAGS) $(SPEEX_CFLAGS) $(SPEEX_DSP_CFLAGS) $(OPENCORE_AMR_CFLAGS) $(BV16_CFLAGS) $(G729_CFLAGS) \
+	$(FFMPEG_CFLAGS) $(SPEEX_CFLAGS) $(SPEEX_DSP_CFLAGS) $(WEBRTC_CFLAGS) $(OPENCORE_AMR_CFLAGS) $(BV16_CFLAGS) $(G729_CFLAGS) \
 	-I../tinySAK/src -I../tinyNET/src -I../tinySDP/include -I../tinyRTP/include -I../tinyMEDIA/include -I../tinyMSRP/include -I./include \
 	-DJB_HISTORY_SIZE=500
 	
 LDFLAGS := $(LDFLAGS_LIB) -L$(THIRDPARTIES_LIB) -L$(THIRDPARTIES_MARCH_LIB) \
-	 $(FFMPEG_LDFLAGS) $(SPEEX_LDFLAGS) $(SPEEX_DSP_LDFLAGS) $(OPENCORE_AMR_LDFLAGS) $(ILBC_LDFLAGS) $(LIBGSM_LDFLAGS) $(BV16_LDFLAGS) $(G729_LDFLAGS)\
+	 $(FFMPEG_LDFLAGS) $(SPEEX_LDFLAGS) $(SPEEX_DSP_LDFLAGS) $(WEBRTC_LDFLAGS) $(OPENCORE_AMR_LDFLAGS) $(ILBC_LDFLAGS) $(LIBGSM_LDFLAGS) $(BV16_LDFLAGS) $(G729_LDFLAGS)\
 	 -ltinySAK_$(MARCH) -ltinyNET_$(MARCH) -ltinySDP_$(MARCH) -ltinyRTP_$(MARCH) -ltinyMEDIA_$(MARCH) -ltinyMSRP_$(MARCH) -lm -lgcc
 
 all: $(APP)
@@ -143,7 +155,8 @@ OBJS += src/audio/tdav_consumer_audio.o \
     src/audio/tdav_session_audio.o \
     src/audio/tdav_speex_denoise.o \
     src/audio/tdav_speex_jitterbuffer.o \
-    src/audio/tdav_speex_resampler.o
+    src/audio/tdav_speex_resampler.o \
+    src/audio/tdav_webrtc_denoise.o
     
      ### video
 OBJS += src/video/tdav_converter_video.o \

branches/2.0/doubango/tinyDAV/include/tinydav/audio/tdav_webrtc_denoise.h

@@ -0,0 +1,79 @@
+/*
+* Copyright (C) 2011 Doubango Telecom <http://www.doubango.org>
+*
+* Contact: Mamadou Diop <diopmamadou(at)doubango.org>
+*	
+* This file is part of Open Source Doubango Framework.
+*
+* DOUBANGO 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 3 of the License, or
+* (at your option) any later version.
+*	
+* DOUBANGO 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 DOUBANGO.
+*
+*/
+
+/**@file tdav_webrtc_denoise.h
+ * @brief Google WebRTC Denoiser (Noise suppression, AGC, AEC) Plugin
+ *
+ * @author Mamadou Diop <diopmamadou(at)doubango.org>
+ *
+
+ */
+#ifndef TINYDAV_WEBRTC_DENOISE_H
+#define TINYDAV_WEBRTC_DENOISE_H
+
+#include "tinydav_config.h"
+
+#if HAVE_WEBRTC && (!defined(HAVE_WEBRTC_DENOISE) || HAVE_WEBRTC_DENOISE)
+
+#include "tinymedia/tmedia_denoise.h"
+
+#include <webrtc/echo_cancellation.h>
+#include <webrtc/noise_suppression.h>
+
+#if !defined(PREFER_SPEEX_DENOISER)
+#	define PREFER_SPEEX_DENOISER	0
+#endif
+
+#if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER /* Speex denoiser works better than WebRTC's denoiser */
+#	include <speex/speex_preprocess.h>
+#endif
+
+TDAV_BEGIN_DECLS
+
+/** Speex denoiser*/
+typedef struct tdav_webrtc_denoise_s
+{
+	TMEDIA_DECLARE_DENOISE;
+
+	void *AEC_inst;
+#if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER
+	SpeexPreprocessState *SpeexDenoiser_proc;
+#else
+	NsHandle *NS_inst;
+#endif
+	
+	uint32_t sound_card_buffer_len;
+	uint32_t echo_skew;
+	uint32_t frame_size;
+	uint32_t sampling_rate;
+
+	WebRtc_Word16 *temp_rec_out;
+}
+tdav_webrtc_denoise_t;
+
+const tmedia_denoise_plugin_def_t *tdav_webrtc_denoise_plugin_def_t;
+
+TDAV_END_DECLS
+
+#endif /* #if HAVE_WEBRTC */
+
+#endif /* TINYDAV_WEBRTC_DENOISE_H */

branches/2.0/doubango/tinyDAV/src/audio/tdav_speex_denoise.c

@@ -53,7 +53,7 @@ static int tdav_speex_denoise_open(tmedia_denoise_t* self, uint32_t frame_size,
 	if(!denoiser->echo_state && TMEDIA_DENOISE(denoiser)->echo_supp_enabled){
 		TSK_DEBUG_INFO("Init Aec frame_size[%d] filter_length[%d] SampleRate[%d]",
 			frame_size* sizeof(spx_int16_t),TMEDIA_DENOISE(denoiser)->echo_tail*frame_size,sampling_rate);
-		if((denoiser->echo_state = speex_echo_state_init(frame_size, TMEDIA_DENOISE(denoiser)->echo_tail*frame_size))){
+		if((denoiser->echo_state = speex_echo_state_init(frame_size, TMEDIA_DENOISE(denoiser)->echo_tail))){
 			speex_echo_ctl(denoiser->echo_state, SPEEX_ECHO_SET_SAMPLING_RATE, &sampling_rate);
 		}
 	}

branches/2.0/doubango/tinyDAV/src/audio/tdav_webrtc_denoise.c

@@ -0,0 +1,362 @@
+/*
+* Copyright (C) 2011 Doubango Telecom <http://www.doubango.org>
+*
+* Contact: Mamadou Diop <diopmamadou(at)doubango.org>
+*	
+* This file is part of Open Source Doubango Framework.
+*
+* DOUBANGO 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 3 of the License, or
+* (at your option) any later version.
+*	
+* DOUBANGO 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 DOUBANGO.
+*
+*/
+
+/**@file tdav_webrtc_denoise.c
+* @brief Google WebRTC Denoiser (Noise suppression, AGC, AEC) Plugin
+*
+* @author Mamadou Diop <diopmamadou(at)doubango.org>
+*
+
+*/
+#include "tinydav/audio/tdav_webrtc_denoise.h"
+
+#if HAVE_WEBRTC && (!defined(HAVE_WEBRTC_DENOISE) || HAVE_WEBRTC_DENOISE)
+
+#include "tsk_memory.h"
+#include "tsk_debug.h"
+
+#include "tinymedia/tmedia_defaults.h"
+
+#include <string.h>
+
+#define WEBRTC_MAX_ECHO_TAIL		500
+
+#if !defined(WEBRTC_HAVE_SPL)
+#	define WEBRTC_HAVE_SPL				0 // To avoid linking against libspl
+#endif
+
+#if !WEBRTC_HAVE_SPL
+
+#define WEBRTC_SPL_MAX_SEED_USED    0x80000000L
+
+static WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32 *seed)
+{
+    seed[0] = (seed[0] * ((WebRtc_Word32)69069) + 1) & (WEBRTC_SPL_MAX_SEED_USED - 1);
+    return seed[0];
+}
+
+static WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32 *seed)
+{
+    return (WebRtc_Word16)(WebRtcSpl_IncreaseSeed(seed) >> 16);
+}
+
+// Creates an array of uniformly distributed variables
+WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
+                                   WebRtc_Word16 vector_length,
+                                   WebRtc_UWord32* seed)
+{
+    int i;
+    for (i = 0; i < vector_length; i++)
+    {
+        vector[i] = WebRtcSpl_RandU(seed);
+    }
+    return vector_length;
+}
+
+#endif /* WEBRTC_HAVE_SPL */
+
+
+static int tdav_webrtc_denoise_set(tmedia_denoise_t* self, const tmedia_param_t* param)
+{
+	/* tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self; */
+	return tmedia_denoise_set(self, param);
+}
+
+static int tdav_webrtc_denoise_open(tmedia_denoise_t* self, uint32_t frame_size, uint32_t sampling_rate)
+{
+	tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self;
+	int ret;
+
+	if(!denoiser){
+		TSK_DEBUG_ERROR("Invalid parameter");
+		return -1;
+	}
+
+	if(denoiser->AEC_inst || 
+#if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER
+		denoiser->SpeexDenoiser_proc
+#else
+		denoiser->NS_inst
+#endif
+		){
+		TSK_DEBUG_ERROR("Denoiser already initialized");
+		return -2;
+	}
+	
+	denoiser->sound_card_buffer_len = TSK_MIN(TMEDIA_DENOISE(denoiser)->echo_tail, WEBRTC_MAX_ECHO_TAIL);
+	denoiser->echo_skew = TMEDIA_DENOISE(denoiser)->echo_skew;
+	denoiser->frame_size = frame_size;
+	denoiser->sampling_rate = sampling_rate;
+	
+	//
+	//	AEC instance
+	//
+	if((ret = WebRtcAec_Create(&denoiser->AEC_inst))){
+		TSK_DEBUG_ERROR("WebRtcAec_Create failed with error code = %d", ret);
+		return ret;
+	}
+	if((ret = WebRtcAec_Init(denoiser->AEC_inst, denoiser->sampling_rate, denoiser->sampling_rate))){
+		TSK_DEBUG_ERROR("WebRtcAec_Init failed with error code = %d", ret);
+		return ret;
+	}
+	
+	//
+	//	Noise Suppression instance
+	//
+	if(TMEDIA_DENOISE(denoiser)->noise_supp_enabled){
+#if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER
+		if((denoiser->SpeexDenoiser_proc = speex_preprocess_state_init(denoiser->frame_size, denoiser->sampling_rate))){
+			int i = 1;
+			speex_preprocess_ctl(denoiser->SpeexDenoiser_proc, SPEEX_PREPROCESS_SET_DENOISE, &i);
+			i = TMEDIA_DENOISE(denoiser)->noise_supp_level;
+			speex_preprocess_ctl(denoiser->SpeexDenoiser_proc, SPEEX_PREPROCESS_SET_NOISE_SUPPRESS, &i);
+		}
+#else
+		if((ret = WebRtcNs_Create(&denoiser->NS_inst))){
+			TSK_DEBUG_ERROR("WebRtcNs_Create failed with error code = %d", ret);
+			return ret;
+		}
+		if((ret = WebRtcNs_Init(denoiser->NS_inst, denoiser->sampling_rate))){
+			TSK_DEBUG_ERROR("WebRtcNs_Init failed with error code = %d", ret);
+			return ret;
+		}
+#endif
+	}
+
+	// allocate temp buffer for record processing
+	if(!(denoiser->temp_rec_out = tsk_realloc(denoiser->temp_rec_out, denoiser->frame_size * sizeof(WebRtc_Word16)))){
+		TSK_DEBUG_ERROR("Failed to allocate new buffer");
+		return -3;
+	}
+
+	TSK_DEBUG_INFO("WebRTC denoiser opened");
+
+	return ret;
+}
+
+static int tdav_webrtc_denoise_echo_playback(tmedia_denoise_t* self, const void* echo_frame)
+{
+	tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self;
+	if(denoiser->AEC_inst){
+		int ret;
+		const WebRtc_Word16 *pEchoFrame = (const WebRtc_Word16 *)echo_frame;
+		switch(denoiser->sampling_rate){
+			case 8000:
+				{
+					if((ret = WebRtcAec_BufferFarend(denoiser->AEC_inst, pEchoFrame, denoiser->frame_size))){
+						TSK_DEBUG_ERROR("WebRtcAec_BufferFarend failed with error code = %d", ret);
+						return ret;
+					}
+					break;
+				}
+				// 32Hz and 16Khz work but produce very ugly results
+			case 16000:
+			case 32000:
+				{
+					uint32_t i;
+					for(i = 0; i<denoiser->frame_size; i+=denoiser->frame_size/2){
+						if((ret = WebRtcAec_BufferFarend(denoiser->AEC_inst, &pEchoFrame[i], denoiser->frame_size/2))){
+							TSK_DEBUG_ERROR("WebRtcAec_BufferFarend failed with error code = %d", ret);
+							return ret;
+						}
+					}
+					break;
+				}
+			default:
+				{
+					TSK_DEBUG_ERROR("%d Hz not supported by WebRTC AEC", denoiser->sampling_rate);
+					return -2;
+				}
+		}
+	}
+	return 0;
+}
+
+
+
+static int tdav_webrtc_denoise_process_record(tmedia_denoise_t* self, void* audio_frame, tsk_bool_t* silence_or_noise)
+{
+	tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self;
+	
+	*silence_or_noise = tsk_false;
+
+	if(denoiser->AEC_inst){
+		int ret;
+		WebRtc_Word16 *pAudioFrame = audio_frame;
+		
+		// Noise suppression
+#if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER
+		speex_preprocess_run(denoiser->SpeexDenoiser_proc, pAudioFrame);
+#else
+
+		// WebRTC NoiseSupp only accept 10ms frames
+		// Our encoder will always output 20ms frames ==> execute 2x noise_supp
+		if(
+				(ret = WebRtcNs_Process(denoiser->NS_inst, pAudioFrame, tsk_null, denoiser->temp_rec_out, tsk_null)) ||
+				(ret = WebRtcNs_Process(denoiser->NS_inst, &pAudioFrame[denoiser->frame_size/2], tsk_null, &denoiser->temp_rec_out[denoiser->frame_size/2], tsk_null))
+			)
+		{
+			TSK_DEBUG_ERROR("WebRtcNs_Process with error code = %d", ret);
+			return ret;
+		}
+#endif
+
+		// AEC
+		switch(denoiser->sampling_rate){
+			case 8000:
+				{
+					if((ret = WebRtcAec_Process(denoiser->AEC_inst, denoiser->temp_rec_out, tsk_null, pAudioFrame, tsk_null, denoiser->frame_size, denoiser->sound_card_buffer_len, denoiser->echo_skew))){
+						TSK_DEBUG_ERROR("WebRtcAec_Process with error code = %d", ret);
+						return ret;
+					}
+					break;
+				}
+				// 32Hz and 16Khz work but produce very ugly results
+			case 16000:
+			case 3200:
+				{
+					uint32_t i;
+					for(i = 0; i<denoiser->frame_size; i+=denoiser->frame_size/2){
+						if((ret = WebRtcAec_Process(denoiser->AEC_inst, &denoiser->temp_rec_out[i], tsk_null, &pAudioFrame[i], tsk_null, denoiser->frame_size/2, denoiser->sound_card_buffer_len, denoiser->echo_skew))){
+							TSK_DEBUG_ERROR("WebRtcAec_Process with error code = %d", ret);
+							return ret;
+						}
+
+					}
+					break;
+				}
+			default:
+				{
+					TSK_DEBUG_ERROR("%d Hz not supported by WebRTC AEC", denoiser->sampling_rate);
+					return -2;
+				}
+		}
+	}
+
+	return 0;
+}
+
+WebRtc_Word16 test[];
+static int tdav_webrtc_denoise_process_playback(tmedia_denoise_t* self, void* audio_frame)
+{
+	tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self;
+
+	// Not mandatory but we could denoise the audio data sent from an encoder without denoiser
+	// All Doubango clients support noise suppression
+	return 0;
+}
+
+static int tdav_webrtc_denoise_close(tmedia_denoise_t* self)
+{
+	tdav_webrtc_denoise_t *denoiser = (tdav_webrtc_denoise_t *)self;
+
+	if(denoiser->AEC_inst){
+		WebRtcAec_Free(denoiser->AEC_inst);
+		denoiser->AEC_inst = tsk_null;
+	}
+#if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER
+	if(denoiser->SpeexDenoiser_proc){
+		speex_preprocess_state_destroy(denoiser->SpeexDenoiser_proc);
+		denoiser->SpeexDenoiser_proc = tsk_null;
+	}
+#else
+	if(denoiser->NS_inst){
+		WebRtcNs_Free(denoiser->NS_inst);
+		denoiser->NS_inst = tsk_null;
+	}
+#endif
+	TSK_FREE(denoiser->temp_rec_out);
+
+	return 0;
+}
+
+
+//
+//	Speex denoiser Plugin definition
+//
+
+/* constructor */
+static tsk_object_t* tdav_webrtc_denoise_ctor(tsk_object_t * self, va_list * app)
+{
+	tdav_webrtc_denoise_t *denoise = self;
+	if(denoise){
+		/* init base */
+		tmedia_denoise_init(TMEDIA_DENOISE(denoise));
+		/* init self */
+	}
+	return self;
+}
+/* destructor */
+static tsk_object_t* tdav_webrtc_denoise_dtor(tsk_object_t * self)
+{ 
+	tdav_webrtc_denoise_t *denoise = self;
+	if(denoise){
+		/* deinit base */
+		tmedia_denoise_deinit(TMEDIA_DENOISE(denoise));
+		/* deinit self */
+		if(denoise->AEC_inst){
+			WebRtcAec_Free(denoise->AEC_inst);
+			denoise->AEC_inst = tsk_null;
+		}
+		
+#if HAVE_SPEEX_DSP && PREFER_SPEEX_DENOISER
+		if(denoise->SpeexDenoiser_proc){
+			speex_preprocess_state_destroy(denoise->SpeexDenoiser_proc);
+			denoise->SpeexDenoiser_proc = tsk_null;
+		}
+#else
+		if(denoise->NS_inst){
+			WebRtcNs_Free(denoise->NS_inst);
+			denoise->NS_inst = tsk_null;
+		}
+#endif
+		TSK_FREE(denoise->temp_rec_out);
+	}
+
+	return self;
+}
+/* object definition */
+static const tsk_object_def_t tdav_webrtc_denoise_def_s = 
+{
+	sizeof(tdav_webrtc_denoise_t),
+	tdav_webrtc_denoise_ctor, 
+	tdav_webrtc_denoise_dtor,
+	tsk_null, 
+};
+/* plugin definition*/
+static const tmedia_denoise_plugin_def_t tdav_webrtc_denoise_plugin_def_s = 
+{
+	&tdav_webrtc_denoise_def_s,
+
+	"Audio Denoiser based on Google WebRTC",
+
+	tdav_webrtc_denoise_set,
+	tdav_webrtc_denoise_open,
+	tdav_webrtc_denoise_echo_playback,
+	tdav_webrtc_denoise_process_record,
+	tdav_webrtc_denoise_process_playback,
+	tdav_webrtc_denoise_close,
+};
+const tmedia_denoise_plugin_def_t *tdav_webrtc_denoise_plugin_def_t = &tdav_webrtc_denoise_plugin_def_s;
+
+
+#endif /* HAVE_WEBRTC */

branches/2.0/doubango/tinyDAV/src/tdav.c

@@ -81,6 +81,9 @@
 #if HAVE_SPEEX_DSP && (!defined(HAVE_SPEEX_DENOISE) || HAVE_SPEEX_DENOISE)
 #	include "tinydav/audio/tdav_speex_denoise.h"
 #endif
+#if HAVE_WEBRTC && (!defined(HAVE_WEBRTC_DENOISE) || HAVE_WEBRTC_DENOISE)
+#	include "tinydav/audio/tdav_webrtc_denoise.h"
+#endif
 
 // Audio resampler
 #if HAVE_SPEEX_DSP && (!defined(HAVE_SPEEX_RESAMPLER) || HAVE_SPEEX_RESAMPLER)
@@ -227,6 +230,9 @@ int tdav_init()
 #if HAVE_SPEEX_DSP && (!defined(HAVE_SPEEX_DENOISE) || HAVE_SPEEX_DENOISE)
 	tmedia_denoise_plugin_register(tdav_speex_denoise_plugin_def_t);
 #endif
+#if HAVE_WEBRTC && (!defined(HAVE_WEBRTC_DENOISE) || HAVE_WEBRTC_DENOISE)
+	tmedia_denoise_plugin_register(tdav_webrtc_denoise_plugin_def_t);
+#endif
 
 	/* === Register Audio Resampler === */
 #if HAVE_SPEEX_DSP && (!defined(HAVE_SPEEX_RESAMPLER) || HAVE_SPEEX_RESAMPLER)
@@ -494,6 +500,9 @@ int tdav_deinit()
 #if HAVE_SPEEX_DSP && (!defined(HAVE_SPEEX_DENOISE) || HAVE_SPEEX_DENOISE)
 	tmedia_denoise_plugin_unregister(tdav_speex_denoise_plugin_def_t);
 #endif
+#if HAVE_WEBRTC && (!defined(HAVE_WEBRTC_DENOISE) || HAVE_WEBRTC_DENOISE)
+	tmedia_denoise_plugin_unregister(tdav_webrtc_denoise_plugin_def_t);
+#endif
 
 	/* === UnRegister Audio Resampler === */
 #if HAVE_SPEEX_DSP && (!defined(HAVE_SPEEX_RESAMPLER) || HAVE_SPEEX_RESAMPLER)

branches/2.0/doubango/tinyDAV/tinyDAV.vcproj

@@ -42,7 +42,7 @@
 				Name="VCCLCompilerTool"
 				Optimization="0"
 				AdditionalIncludeDirectories=""$(PSDK_DIR)include";"$(DXSDK_DIR)include";..\thirdparties\win32\include;include;..\tinyMSRP\include;..\tinyRTP\include;..\tinyMEDIA\include;..\tinySDP\include;..\tinyNET\src;..\tinyDSHOW\include;..\tinySAK\src;..\thirdparties\win32\include\BroadVoice16\bvcommon;..\thirdparties\win32\include\BroadVoice16\bv16"
-				PreprocessorDefinitions="HAVE_G729=0;HAVE_BV16=0;HAVE_OPENCORE_AMR=1;HAVE_ILBC=0;HAVE_LIBGSM=1;HAVE_TINYDSHOW=1;HAVE_DSOUND_H=1;HAVE_WAVE_API=0;HAVE_FFMPEG=1;HAVE_SPEEX_DSP=1;HAVE_SPEEX_JB=1;HAVE_LIB_SPEEX=1;G192BITSTREAM=0;DEBUG_LEVEL=DEBUG_LEVEL_INFO;WIN32;_DEBUG;_WINDOWS;_USRDLL;_WIN32_WINNT=0x0501;TINYDAV_EXPORTS"
+				PreprocessorDefinitions="HAVE_G729=0;HAVE_BV16=0;HAVE_OPENCORE_AMR=1;HAVE_ILBC=0;HAVE_LIBGSM=1;HAVE_TINYDSHOW=1;HAVE_DSOUND_H=1;HAVE_WAVE_API=0;HAVE_FFMPEG=1;HAVE_SPEEX_DSP=1;HAVE_WEBRTC=1;HAVE_SPEEX_JB=1;HAVE_LIB_SPEEX=1;G192BITSTREAM=0;DEBUG_LEVEL=DEBUG_LEVEL_INFO;WIN32;_DEBUG;_WINDOWS;_USRDLL;_WIN32_WINNT=0x0501;TINYDAV_EXPORTS"
 				MinimalRebuild="true"
 				BasicRuntimeChecks="3"
 				RuntimeLibrary="3"
@@ -64,9 +64,9 @@
 			/>
 			<Tool
 				Name="VCLinkerTool"
-				AdditionalDependencies="Winmm.lib $(OutDir)\tinySAK.lib $(OutDir)\tinyNET.lib $(OutDir)\tinyRTP.lib $(OutDir)\tinyMSRP.lib $(OutDir)\tinySDP.lib $(OutDir)\tinyMEDIA.lib $(OutDir)\tinyDSHOW.lib &quot;..\thirdparties\win32\lib\gsm\libgsm.a&quot; &quot;..\thirdparties\win32\lib\ilbc\libiLBC.a&quot; &quot;..\thirdparties\win32\lib\speex\libspeex.a&quot; &quot;..\thirdparties\win32\lib\speex\libspeexdsp.a&quot; ..\thirdparties\win32\lib\libgcc.a ..\thirdparties\win32\lib\libmingwex.a &quot;..\thirdparties\win32\lib\ffmpeg\libavcodec.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libavutil.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libswscale.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libavcore.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libx264.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libtheora.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libogg.a&quot;"
+				AdditionalDependencies="Winmm.lib $(OutDir)\tinySAK.lib $(OutDir)\tinyNET.lib $(OutDir)\tinyRTP.lib $(OutDir)\tinyMSRP.lib $(OutDir)\tinySDP.lib $(OutDir)\tinyMEDIA.lib $(OutDir)\tinyDSHOW.lib &quot;..\thirdparties\win32\lib\gsm\libgsm.a&quot; &quot;..\thirdparties\win32\lib\ilbc\libiLBC.a&quot; &quot;..\thirdparties\win32\lib\speex\libspeex.a&quot; &quot;..\thirdparties\win32\lib\speex\libspeexdsp.a&quot; ..\thirdparties\win32\lib\libgcc.a ..\thirdparties\win32\lib\libmingwex.a &quot;..\thirdparties\win32\lib\ffmpeg\libavcodec.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libavutil.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libswscale.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libavcore.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libx264.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libtheora.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libogg.a&quot; &quot;..\thirdparties\win32\lib\webrtc\aec.lib&quot; &quot;..\thirdparties\win32\lib\webrtc\apm_util.lib&quot; &quot;..\thirdparties\win32\lib\webrtc\system_wrappers.lib&quot; &quot;..\thirdparties\win32\lib\webrtc\ns.lib&quot;"
 				LinkIncremental="2"
-				IgnoreDefaultLibraryNames="MSVCRT"
+				IgnoreDefaultLibraryNames="MSVCRT;LIBCMTD;LIBCMT"
 				GenerateDebugInformation="true"
 				SubSystem="2"
 				RandomizedBaseAddress="1"
@@ -122,7 +122,7 @@
 				Name="VCCLCompilerTool"
 				EnableIntrinsicFunctions="false"
 				AdditionalIncludeDirectories="&quot;$(PSDK_DIR)include&quot;;&quot;$(DXSDK_DIR)include&quot;;..\thirdparties\win32\include;include;..\tinyMSRP\include;..\tinyRTP\include;..\tinyMEDIA\include;..\tinySDP\include;..\tinyNET\src;..\tinyDSHOW\include;..\tinySAK\src;..\thirdparties\win32\include\BroadVoice16\bvcommon;..\thirdparties\win32\include\BroadVoice16\bv16"
-				PreprocessorDefinitions="WIN32;NDEBUG;_WINDOWS;_USRDLL;TINYDAV_EXPORTS;DEBUG_LEVEL=DEBUG_LEVEL_INFOS;HAVE_G729=0;HAVE_BV16=0;HAVE_OPENCORE_AMR=1;HAVE_ILBC=0;HAVE_LIBGSM=1;HAVE_TINYDSHOW=1;HAVE_DSOUND_H=1;HAVE_WAVE_API=0;HAVE_FFMPEG=1;HAVE_SPEEX_DSP=1;HAVE_SPEEX_JB=1;HAVE_LIB_SPEEX=1;G192BITSTREAM=0;_WIN32_WINNT=0x0501"
+				PreprocessorDefinitions="WIN32;NDEBUG;_WINDOWS;_USRDLL;TINYDAV_EXPORTS;DEBUG_LEVEL=DEBUG_LEVEL_INFOS;HAVE_G729=0;HAVE_BV16=0;HAVE_OPENCORE_AMR=1;HAVE_ILBC=0;HAVE_LIBGSM=1;HAVE_TINYDSHOW=1;HAVE_DSOUND_H=1;HAVE_WAVE_API=0;HAVE_FFMPEG=1;HAVE_SPEEX_DSP=1;HAVE_WEBRTC=1;HAVE_SPEEX_JB=1;HAVE_LIB_SPEEX=1;G192BITSTREAM=0;_WIN32_WINNT=0x0501"
 				RuntimeLibrary="2"
 				EnableFunctionLevelLinking="false"
 				UsePrecompiledHeader="0"
@@ -143,9 +143,9 @@
 			/>
 			<Tool
 				Name="VCLinkerTool"
-				AdditionalDependencies="Winmm.lib $(OutDir)\tinySAK.lib $(OutDir)\tinyNET.lib $(OutDir)\tinyRTP.lib $(OutDir)\tinyMSRP.lib $(OutDir)\tinySDP.lib $(OutDir)\tinyMEDIA.lib $(OutDir)\tinyDSHOW.lib &quot;..\thirdparties\win32\lib\gsm\libgsm.a&quot; &quot;..\thirdparties\win32\lib\ilbc\libiLBC.a&quot; &quot;..\thirdparties\win32\lib\speex\libspeex.a&quot; &quot;..\thirdparties\win32\lib\speex\libspeexdsp.a&quot; ..\thirdparties\win32\lib\libgcc.a ..\thirdparties\win32\lib\libmingwex.a &quot;..\thirdparties\win32\lib\ffmpeg\libavcodec.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libavutil.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libswscale.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libavcore.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libx264.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libtheora.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libogg.a&quot; &quot;..\thirdparties\win32\lib\BroadVoice16\libbv16.a&quot; &quot;..\thirdparties\win32\lib\opencore\libopencore-amrnb.a&quot;"
+				AdditionalDependencies="Winmm.lib $(OutDir)\tinySAK.lib $(OutDir)\tinyNET.lib $(OutDir)\tinyRTP.lib $(OutDir)\tinyMSRP.lib $(OutDir)\tinySDP.lib $(OutDir)\tinyMEDIA.lib $(OutDir)\tinyDSHOW.lib &quot;..\thirdparties\win32\lib\gsm\libgsm.a&quot; &quot;..\thirdparties\win32\lib\ilbc\libiLBC.a&quot; &quot;..\thirdparties\win32\lib\speex\libspeex.a&quot; &quot;..\thirdparties\win32\lib\speex\libspeexdsp.a&quot; ..\thirdparties\win32\lib\libgcc.a ..\thirdparties\win32\lib\libmingwex.a &quot;..\thirdparties\win32\lib\ffmpeg\libavcodec.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libavutil.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libswscale.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libavcore.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libx264.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libtheora.a&quot; &quot;..\thirdparties\win32\lib\ffmpeg\libogg.a&quot; &quot;..\thirdparties\win32\lib\BroadVoice16\libbv16.a&quot; &quot;..\thirdparties\win32\lib\opencore\libopencore-amrnb.a&quot; &quot;..\thirdparties\win32\lib\webrtc\aec.lib&quot; &quot;..\thirdparties\win32\lib\webrtc\apm_util.lib&quot; &quot;..\thirdparties\win32\lib\webrtc\system_wrappers.lib&quot; &quot;..\thirdparties\win32\lib\webrtc\ns.lib&quot;"
 				LinkIncremental="1"
-				IgnoreDefaultLibraryNames="MSVCRTD"
+				IgnoreDefaultLibraryNames="MSVCRTD;LIBCMT"
 				GenerateDebugInformation="false"
 				SubSystem="2"
 				OptimizeReferences="2"
@@ -370,6 +370,10 @@
 					RelativePath=".\include\tinydav\audio\tdav_speex_resampler.h"
 					>
 				</File>
+				<File
+					RelativePath=".\include\tinydav\audio\tdav_webrtc_denoise.h"
+					>
+				</File>
 				<Filter
 					Name="android"
 					>
@@ -654,6 +658,10 @@
 					RelativePath=".\src\audio\tdav_speex_resampler.c"
 					>
 				</File>
+				<File
+					RelativePath=".\src\audio\tdav_webrtc_denoise.c"
+					>
+				</File>
 				<Filter
 					Name="android"
 					>

branches/2.0/doubango/tinyMEDIA/include/tinymedia/tmedia_defaults.h

@@ -38,7 +38,9 @@ TINYMEDIA_API tmedia_bandwidth_level_t tmedia_defaults_get_bl();
 // Denoiser: Echo suppression, AEC, Noise redution, AGC, ...
 //
 TINYMEDIA_API int tmedia_defaults_set_echo_tail(uint32_t echo_tail);
+TINYMEDIA_API int tmedia_defaults_set_echo_skew(uint32_t echo_skew);
 TINYMEDIA_API uint32_t tmedia_defaults_get_echo_tail();
+TINYMEDIA_API uint32_t tmedia_defaults_get_echo_skew();
 TINYMEDIA_API int tmedia_defaults_set_echo_supp_enabled(tsk_bool_t echo_supp_enabled);
 TINYMEDIA_API tsk_bool_t tmedia_defaults_get_echo_supp_enabled();
 TINYMEDIA_API int tmedia_defaults_set_agc_enabled(tsk_bool_t agc_enabled);

branches/2.0/doubango/tinyMEDIA/include/tinymedia/tmedia_denoise.h

@@ -46,6 +46,7 @@ typedef struct tmedia_denoise_s
 	tsk_bool_t opened;
 	
 	uint32_t echo_tail;
+	uint32_t echo_skew;
 	tsk_bool_t echo_supp_enabled;
 	tsk_bool_t agc_enabled ;
 	float agc_level;

branches/2.0/doubango/tinyMEDIA/src/tmedia_defaults.c

@@ -45,6 +45,7 @@ tmedia_bandwidth_level_t tmedia_defaults_get_bl()
 // Denoiser: Echo suppression, AEC, Noise redution, AGC, ...
 //
 static uint32_t __echo_tail = 20;
+static uint32_t __echo_skew = 0;
 static tsk_bool_t __echo_supp_enabled;
 static tsk_bool_t __agc_enabled = tsk_false;
 static float __agc_level = 8000.0f;
@@ -61,11 +62,22 @@ int tmedia_defaults_set_echo_tail(uint32_t echo_tail)
 	return 0;
 }
 
+int tmedia_defaults_set_echo_skew(uint32_t echo_skew)
+{
+	__echo_skew = echo_skew;
+	return 0;
+}
+
 uint32_t tmedia_defaults_get_echo_tail()
 {
 	return __echo_tail;
 }
 
+uint32_t tmedia_defaults_get_echo_skew()
+{
+	return __echo_skew;
+}
+
 int tmedia_defaults_set_echo_supp_enabled(tsk_bool_t echo_supp_enabled)
 {
 	__echo_supp_enabled = echo_supp_enabled;

branches/2.0/doubango/tinyMEDIA/src/tmedia_denoise.c

@@ -42,6 +42,7 @@ int tmedia_denoise_init(tmedia_denoise_t* self)
 	}
 
 	self->echo_tail = tmedia_defaults_get_echo_tail();
+	self->echo_skew = tmedia_defaults_get_echo_skew();
 	self->echo_supp_enabled = tmedia_defaults_get_echo_supp_enabled();
 	self->agc_enabled = tmedia_defaults_get_agc_enabled();
 	self->agc_level = tmedia_defaults_get_agc_level();

branches/2.0/doubango/tinyMEDIA/src/tmedia_session.c

@@ -344,7 +344,6 @@ tmedia_codecs_L_t* tmedia_session_match_codec(tmedia_session_t* self, const tsdp
 			// Guard to avoid matching a codec more than once
 			// For example, H.264 codecs without profiles (Jetsi,Tiscali PC client) to distinguish them could match more than once
 			if(matchingCodecs && tsk_list_find_object_by_pred(matchingCodecs, __pred_find_codec_by_format, codec)){
-				TSK_DEBUG_WARN("Codec already matched. Try to add valid FMTP to your codec to help matching.");
 				continue;
 			}