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diff --git a/src/modules/audio_processing/agc/main/source/digital_agc.h b/src/modules/audio_processing/agc/main/source/digital_agc.h
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+++ b/src/modules/audio_processing/agc/main/source/digital_agc.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_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
+
+#ifdef AGC_DEBUG
+#include <stdio.h>
+#endif
+#include "typedefs.h"
+#include "signal_processing_library.h"
+
+// the 32 most significant bits of A(19) * B(26) >> 13
+#define AGC_MUL32(A, B)             (((B)>>13)*(A) + ( ((0x00001FFF & (B))*(A)) >> 13 ))
+// C + the 32 most significant bits of A * B
+#define AGC_SCALEDIFF32(A, B, C)    ((C) + ((B)>>16)*(A) + ( ((0x0000FFFF & (B))*(A)) >> 16 ))
+
+typedef struct
+{
+    WebRtc_Word32 downState[8];
+    WebRtc_Word16 HPstate;
+    WebRtc_Word16 counter;
+    WebRtc_Word16 logRatio; // log( P(active) / P(inactive) ) (Q10)
+    WebRtc_Word16 meanLongTerm; // Q10
+    WebRtc_Word32 varianceLongTerm; // Q8
+    WebRtc_Word16 stdLongTerm; // Q10
+    WebRtc_Word16 meanShortTerm; // Q10
+    WebRtc_Word32 varianceShortTerm; // Q8
+    WebRtc_Word16 stdShortTerm; // Q10
+} AgcVad_t; // total = 54 bytes
+
+typedef struct
+{
+    WebRtc_Word32 capacitorSlow;
+    WebRtc_Word32 capacitorFast;
+    WebRtc_Word32 gain;
+    WebRtc_Word32 gainTable[32];
+    WebRtc_Word16 gatePrevious;
+    WebRtc_Word16 agcMode;
+    AgcVad_t      vadNearend;
+    AgcVad_t      vadFarend;
+#ifdef AGC_DEBUG
+    FILE*         logFile;
+    int           frameCounter;
+#endif
+} DigitalAgc_t;
+
+WebRtc_Word32 WebRtcAgc_InitDigital(DigitalAgc_t *digitalAgcInst, WebRtc_Word16 agcMode);
+
+WebRtc_Word32 WebRtcAgc_ProcessDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inNear,
+                             const WebRtc_Word16 *inNear_H, WebRtc_Word16 *out,
+                             WebRtc_Word16 *out_H, WebRtc_UWord32 FS,
+                             WebRtc_Word16 lowLevelSignal);
+
+WebRtc_Word32 WebRtcAgc_AddFarendToDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inFar,
+                                 WebRtc_Word16 nrSamples);
+
+void WebRtcAgc_InitVad(AgcVad_t *vadInst);
+
+WebRtc_Word16 WebRtcAgc_ProcessVad(AgcVad_t *vadInst, // (i) VAD state
+                            const WebRtc_Word16 *in, // (i) Speech signal
+                            WebRtc_Word16 nrSamples); // (i) number of samples
+
+WebRtc_Word32 WebRtcAgc_CalculateGainTable(WebRtc_Word32 *gainTable, // Q16
+                                 WebRtc_Word16 compressionGaindB, // Q0 (in dB)
+                                 WebRtc_Word16 targetLevelDbfs,// Q0 (in dB)
+                                 WebRtc_UWord8 limiterEnable, WebRtc_Word16 analogTarget);
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_