Support for Signal calls.

Merge in RedPhone

// FREEBIE

jni/webrtc/modules/audio_processing/aec-tmp/aec_resampler.h

@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2012 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_AEC_RESAMPLER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
+
+#include "webrtc/modules/audio_processing/aec/aec_core.h"
+
+enum {
+  kResamplingDelay = 1
+};
+enum {
+  kResamplerBufferSize = FRAME_LEN * 4
+};
+
+// Unless otherwise specified, functions return 0 on success and -1 on error
+int WebRtcAec_CreateResampler(void** resampInst);
+int WebRtcAec_InitResampler(void* resampInst, int deviceSampleRateHz);
+int WebRtcAec_FreeResampler(void* resampInst);
+
+// Estimates skew from raw measurement.
+int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst);
+
+// Resamples input using linear interpolation.
+void WebRtcAec_ResampleLinear(void* resampInst,
+                              const float* inspeech,
+                              int size,
+                              float skew,
+                              float* outspeech,
+                              int* size_out);
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_