Clang format on AGC legacy code.

webrtc/modules/audio_processing/agc/legacy/digital_agc.h

 /*
  *  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_LEGACY_DIGITAL_AGC_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
 
 #ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/typedefs.h"
 
 // the 32 most significant bits of A(19) * B(26) >> 13
-#define AGC_MUL32(A, B)             (((B)>>13)*(A) + ( ((0x00001FFF & (B))*(A)) >> 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 ))
+#define AGC_SCALEDIFF32(A, B, C) \
+  ((C) + ((B) >> 16) * (A) + (((0x0000FFFF & (B)) * (A)) >> 16))
 
-typedef struct
-{
-    int32_t downState[8];
-    int16_t HPstate;
-    int16_t counter;
-    int16_t logRatio; // log( P(active) / P(inactive) ) (Q10)
-    int16_t meanLongTerm; // Q10
-    int32_t varianceLongTerm; // Q8
-    int16_t stdLongTerm; // Q10
-    int16_t meanShortTerm; // Q10
-    int32_t varianceShortTerm; // Q8
-    int16_t stdShortTerm; // Q10
-} AgcVad;  // total = 54 bytes
+typedef struct {
+  int32_t downState[8];
+  int16_t HPstate;
+  int16_t counter;
+  int16_t logRatio;           // log( P(active) / P(inactive) ) (Q10)
+  int16_t meanLongTerm;       // Q10
+  int32_t varianceLongTerm;   // Q8
+  int16_t stdLongTerm;        // Q10
+  int16_t meanShortTerm;      // Q10
+  int32_t varianceShortTerm;  // Q8
+  int16_t stdShortTerm;       // Q10
+} AgcVad;                     // total = 54 bytes
 
-typedef struct
-{
-    int32_t capacitorSlow;
-    int32_t capacitorFast;
-    int32_t gain;
-    int32_t gainTable[32];
-    int16_t gatePrevious;
-    int16_t agcMode;
-    AgcVad vadNearend;
-    AgcVad vadFarend;
+typedef struct {
+  int32_t capacitorSlow;
+  int32_t capacitorFast;
+  int32_t gain;
+  int32_t gainTable[32];
+  int16_t gatePrevious;
+  int16_t agcMode;
+  AgcVad vadNearend;
+  AgcVad vadFarend;
 #ifdef WEBRTC_AGC_DEBUG_DUMP
-    FILE* logFile;
-    int frameCounter;
+  FILE* logFile;
+  int frameCounter;
 #endif
 } DigitalAgc;
 
 int32_t WebRtcAgc_InitDigital(DigitalAgc* digitalAgcInst, int16_t agcMode);
 
 int32_t WebRtcAgc_ProcessDigital(DigitalAgc* digitalAgcInst,
                                  const int16_t* const* inNear,
                                  size_t num_bands,
                                  int16_t* const* out,
                                  uint32_t FS,
                                  int16_t lowLevelSignal);
 
 int32_t WebRtcAgc_AddFarendToDigital(DigitalAgc* digitalAgcInst,
                                      const int16_t* inFar,
                                      size_t nrSamples);
 
 void WebRtcAgc_InitVad(AgcVad* vadInst);
 
-int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst,     // (i) VAD state
-                             const int16_t* in,   // (i) Speech signal
+int16_t WebRtcAgc_ProcessVad(AgcVad* vadInst,    // (i) VAD state
+                             const int16_t* in,  // (i) Speech signal
                              size_t nrSamples);  // (i) number of samples
 
-int32_t WebRtcAgc_CalculateGainTable(int32_t *gainTable, // Q16
-                                     int16_t compressionGaindB, // Q0 (in dB)
-                                     int16_t targetLevelDbfs,// Q0 (in dB)
+int32_t WebRtcAgc_CalculateGainTable(int32_t* gainTable,         // Q16
+                                     int16_t compressionGaindB,  // Q0 (in dB)
+                                     int16_t targetLevelDbfs,    // Q0 (in dB)
                                      uint8_t limiterEnable,
                                      int16_t analogTarget);
 
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_LEGACY_DIGITAL_AGC_H_