1 files changed, 35 insertions, 0 deletions
diff --git a/src/modules/audio_processing/aec/aec_resampler.h b/src/modules/audio_processing/aec/aec_resampler.h
new file mode 100644
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+++ b/src/modules/audio_processing/aec/aec_resampler.h
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+/*
+ *  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_AEC_RESAMPLER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
+
+#include "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.
+// Returns size of resampled array.
+int WebRtcAec_ResampleLinear(void *resampInst,
+                             const short *inspeech,
+                             int size,
+                             float skew,
+                             short *outspeech);
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_