diff --git a/webrtc/modules/audio_processing/agc/analog_agc.c b/webrtc/modules/audio_processing/agc/analog_agc.c
index 0376dae..32929dd 100644
--- a/webrtc/modules/audio_processing/agc/analog_agc.c
+++ b/webrtc/modules/audio_processing/agc/analog_agc.c
@@ -19,7 +19,7 @@
 
 #include <assert.h>
 #include <stdlib.h>
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif
 #include "webrtc/modules/audio_processing/agc/analog_agc.h"
@@ -139,10 +139,10 @@
             L = 8;
         } else
         {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
             fprintf(stt->fpt,
                     "AGC->add_mic, frame %d: Invalid number of samples\n\n",
-                    (stt->fcount + 1));
+                    stt->fcount + 1);
 #endif
             return -1;
         }
@@ -160,10 +160,10 @@
             L = 16;
         } else
         {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
             fprintf(stt->fpt,
                     "AGC->add_mic, frame %d: Invalid number of samples\n\n",
-                    (stt->fcount + 1));
+                    stt->fcount + 1);
 #endif
             return -1;
         }
@@ -177,10 +177,10 @@
             L = 16;
         } else
         {
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
             fprintf(stt->fpt,
                     "AGC->add_mic, frame %d: Invalid sample rate\n\n",
-                    (stt->fcount + 1));
+                    stt->fcount + 1);
 #endif
             return -1;
         }
@@ -343,7 +343,7 @@
     {
         if ((samples != 80) && (samples != 160))
         {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
             fprintf(stt->fpt,
                     "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
                     stt->fcount);
@@ -355,7 +355,7 @@
     {
         if ((samples != 160) && (samples != 320))
         {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
             fprintf(stt->fpt,
                     "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
                     stt->fcount);
@@ -367,7 +367,7 @@
     {
         if ((samples != 160) && (samples != 320))
         {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
             fprintf(stt->fpt,
                     "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
                     stt->fcount);
@@ -377,7 +377,7 @@
         subFrames = 160;
     } else
     {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
         fprintf(stt->fpt,
                 "AGC->add_far_end, frame %d: Invalid sample rate\n\n",
                 stt->fcount + 1);
@@ -657,10 +657,12 @@
             stt->micVol = *inMicLevel;
         }
 
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
         fprintf(stt->fpt,
-                "\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold, micVol:\n",
-                stt->fcount, stt->micVol);
+                "\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold,"
+                " micVol: %d\n",
+                stt->fcount,
+                stt->micVol);
 #endif
 
         stt->activeSpeech = 0;
@@ -771,14 +773,18 @@
 
     if (inMicLevelTmp > stt->maxAnalog)
     {
-#ifdef AGC_DEBUG //test log
-        fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n", stt->fcount);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+        fprintf(stt->fpt,
+                "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n",
+                stt->fcount);
 #endif
         return -1;
     } else if (inMicLevelTmp < stt->minLevel)
     {
-#ifdef AGC_DEBUG //test log
-        fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel\n", stt->fcount);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+        fprintf(stt->fpt,
+                "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel\n",
+                stt->fcount);
 #endif
         return -1;
     }
@@ -813,9 +819,10 @@
 #ifdef MIC_LEVEL_FEEDBACK
         //stt->numBlocksMicLvlSat = 0;
 #endif
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
         fprintf(stt->fpt,
-                "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel by manual decrease, raise vol\n",
+                "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel by manual"
+                " decrease, raise vol\n",
                 stt->fcount);
 #endif
     }
@@ -871,10 +878,11 @@
         }
         inMicLevelTmp = stt->micVol;
 
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
         fprintf(stt->fpt,
                 "\tAGC->ProcessAnalog, frame %d: saturated, micVol = %d\n",
-                stt->fcount, stt->micVol);
+                stt->fcount,
+                stt->micVol);
 #endif
 
         if (stt->micVol < stt->minOutput)
@@ -1011,10 +1019,13 @@
 #ifdef MIC_LEVEL_FEEDBACK
                     //stt->numBlocksMicLvlSat = 0;
 #endif
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
                     fprintf(stt->fpt,
-                            "\tAGC->ProcessAnalog, frame %d: measure > 2ndUpperLim, micVol = %d, maxLevel = %d\n",
-                            stt->fcount, stt->micVol, stt->maxLevel);
+                            "\tAGC->ProcessAnalog, frame %d: measure >"
+                            " 2ndUpperLim, micVol = %d, maxLevel = %d\n",
+                            stt->fcount,
+                            stt->micVol,
+                            stt->maxLevel);
 #endif
                 }
             } else if (stt->Rxx160_LPw32 > stt->upperLimit)
@@ -1054,10 +1065,13 @@
 #ifdef MIC_LEVEL_FEEDBACK
                     //stt->numBlocksMicLvlSat = 0;
 #endif
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
                     fprintf(stt->fpt,
-                            "\tAGC->ProcessAnalog, frame %d: measure > UpperLim, micVol = %d, maxLevel = %d\n",
-                            stt->fcount, stt->micVol, stt->maxLevel);
+                            "\tAGC->ProcessAnalog, frame %d: measure >"
+                            " UpperLim, micVol = %d, maxLevel = %d\n",
+                            stt->fcount,
+                            stt->micVol,
+                            stt->maxLevel);
 #endif
                 }
             } else if (stt->Rxx160_LPw32 < stt->lowerSecondaryLimit)
@@ -1113,10 +1127,12 @@
                         fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
                     }
 #endif
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
                     fprintf(stt->fpt,
-                            "\tAGC->ProcessAnalog, frame %d: measure < 2ndLowerLim, micVol = %d\n",
-                            stt->fcount, stt->micVol);
+                            "\tAGC->ProcessAnalog, frame %d: measure <"
+                            " 2ndLowerLim, micVol = %d\n",
+                            stt->fcount,
+                            stt->micVol);
 #endif
                 }
             } else if (stt->Rxx160_LPw32 < stt->lowerLimit)
@@ -1172,10 +1188,11 @@
                         fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
                     }
 #endif
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
                     fprintf(stt->fpt,
                             "\tAGC->ProcessAnalog, frame %d: measure < LowerLim, micVol = %d\n",
-                            stt->fcount, stt->micVol);
+                            stt->fcount,
+                            stt->micVol);
 #endif
 
                 }
@@ -1272,9 +1289,10 @@
     {
         if ((samples != 80) && (samples != 160))
         {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
             fprintf(stt->fpt,
-                    "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
+                    "AGC->Process, frame %d: Invalid number of samples\n\n",
+                    stt->fcount);
 #endif
             return -1;
         }
@@ -1283,9 +1301,10 @@
     {
         if ((samples != 160) && (samples != 320))
         {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
             fprintf(stt->fpt,
-                    "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
+                    "AGC->Process, frame %d: Invalid number of samples\n\n",
+                    stt->fcount);
 #endif
             return -1;
         }
@@ -1294,18 +1313,20 @@
     {
         if ((samples != 160) && (samples != 320))
         {
-#ifdef AGC_DEBUG //test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
             fprintf(stt->fpt,
-                    "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
+                    "AGC->Process, frame %d: Invalid number of samples\n\n",
+                    stt->fcount);
 #endif
             return -1;
         }
         subFrames = 160;
     } else
     {
-#ifdef AGC_DEBUG// test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
         fprintf(stt->fpt,
-                "AGC->Process, frame %d: Invalid sample rate\n\n", stt->fcount);
+                "AGC->Process, frame %d: Invalid sample rate\n\n",
+                stt->fcount);
 #endif
         return -1;
     }
@@ -1341,7 +1362,7 @@
         }
     }
 
-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
     stt->fcount++;
 #endif
 
@@ -1350,8 +1371,10 @@
         if (WebRtcAgc_ProcessDigital(&stt->digitalAgc, &in_near[i], &in_near_H[i], &out[i], &out_H[i],
                            stt->fs, stt->lowLevelSignal) == -1)
         {
-#ifdef AGC_DEBUG//test log
-            fprintf(stt->fpt, "AGC->Process, frame %d: Error from DigAGC\n\n", stt->fcount);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+            fprintf(stt->fpt,
+                    "AGC->Process, frame %d: Error from DigAGC\n\n",
+                    stt->fcount);
 #endif
             return -1;
         }
@@ -1364,8 +1387,14 @@
                 return -1;
             }
         }
-#ifdef AGC_DEBUG//test log
-        fprintf(stt->agcLog, "%5d\t%d\t%d\t%d\n", stt->fcount, inMicLevelTmp, *outMicLevel, stt->maxLevel, stt->micVol);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+        fprintf(stt->agcLog,
+                "%5d\t%d\t%d\t%d\t%d\n",
+                stt->fcount,
+                inMicLevelTmp,
+                *outMicLevel,
+                stt->maxLevel,
+                stt->micVol);
 #endif
 
         /* update queue */
@@ -1441,8 +1470,10 @@
     if (WebRtcAgc_CalculateGainTable(&(stt->digitalAgc.gainTable[0]), stt->compressionGaindB,
                            stt->targetLevelDbfs, stt->limiterEnable, stt->analogTarget) == -1)
     {
-#ifdef AGC_DEBUG//test log
-        fprintf(stt->fpt, "AGC->set_config, frame %d: Error from calcGainTable\n\n", stt->fcount);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+        fprintf(stt->fpt,
+                "AGC->set_config, frame %d: Error from calcGainTable\n\n",
+                stt->fcount);
 #endif
         return -1;
     }
@@ -1498,7 +1529,7 @@
         return -1;
     }
 
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
     stt->fpt = fopen("./agc_test_log.txt", "wt");
     stt->agcLog = fopen("./agc_debug_log.txt", "wt");
     stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt");
@@ -1515,7 +1546,7 @@
     Agc_t *stt;
 
     stt = (Agc_t *)state;
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
     fclose(stt->fpt);
     fclose(stt->agcLog);
     fclose(stt->digitalAgc.logFile);
@@ -1553,13 +1584,13 @@
      *            2 - Digital Automatic Gain Control [-targetLevelDbfs (default -3 dBOv)]
      *            3 - Fixed Digital Gain [compressionGaindB (default 8 dB)]
      */
-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
     stt->fcount = 0;
     fprintf(stt->fpt, "AGC->Init\n");
 #endif
     if (agcMode < kAgcModeUnchanged || agcMode > kAgcModeFixedDigital)
     {
-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
         fprintf(stt->fpt, "AGC->Init: error, incorrect mode\n\n");
 #endif
         return -1;
@@ -1616,10 +1647,12 @@
     stt->numBlocksMicLvlSat = 0;
     stt->micLvlSat = 0;
 #endif
-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
     fprintf(stt->fpt,
             "AGC->Init: minLevel = %d, maxAnalog = %d, maxLevel = %d\n",
-            stt->minLevel, stt->maxAnalog, stt->maxLevel);
+            stt->minLevel,
+            stt->maxAnalog,
+            stt->maxLevel);
 #endif
 
     /* Minimum output volume is 4% higher than the available lowest volume level */
@@ -1687,13 +1720,13 @@
     /* Only positive values are allowed that are not too large */
     if ((minLevel >= maxLevel) || (maxLevel & 0xFC000000))
     {
-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
         fprintf(stt->fpt, "minLevel, maxLevel value(s) are invalid\n\n");
 #endif
         return -1;
     } else
     {
-#ifdef AGC_DEBUG//test log
+#ifdef WEBRTC_AGC_DEBUG_DUMP
         fprintf(stt->fpt, "\n");
 #endif
         return 0;
diff --git a/webrtc/modules/audio_processing/agc/analog_agc.h b/webrtc/modules/audio_processing/agc/analog_agc.h
index b036f44..19b324f 100644
--- a/webrtc/modules/audio_processing/agc/analog_agc.h
+++ b/webrtc/modules/audio_processing/agc/analog_agc.h
@@ -15,9 +15,8 @@
 #include "webrtc/modules/audio_processing/agc/include/gain_control.h"
 #include "webrtc/typedefs.h"
 
-//#define AGC_DEBUG
 //#define MIC_LEVEL_FEEDBACK
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif
 
@@ -122,10 +121,10 @@
     AgcVad_t            vadMic;
     DigitalAgc_t        digitalAgc;
 
-#ifdef AGC_DEBUG
-    FILE*               fpt;
-    FILE*               agcLog;
-    int32_t             fcount;
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+    FILE* fpt;
+    FILE* agcLog;
+    int32_t fcount;
 #endif
 
     int16_t             lowLevelSignal;
diff --git a/webrtc/modules/audio_processing/agc/digital_agc.c b/webrtc/modules/audio_processing/agc/digital_agc.c
index e74bb4c..da087ca 100644
--- a/webrtc/modules/audio_processing/agc/digital_agc.c
+++ b/webrtc/modules/audio_processing/agc/digital_agc.c
@@ -16,7 +16,7 @@
 
 #include <assert.h>
 #include <string.h>
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif
 
@@ -274,7 +274,7 @@
     stt->gain = 65536;
     stt->gatePrevious = 0;
     stt->agcMode = agcMode;
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
     stt->frameCounter = 0;
 #endif
 
@@ -397,9 +397,14 @@
             decay = 0;
         }
     }
-#ifdef AGC_DEBUG
+#ifdef WEBRTC_AGC_DEBUG_DUMP
     stt->frameCounter++;
-    fprintf(stt->logFile, "%5.2f\t%d\t%d\t%d\t", (float)(stt->frameCounter) / 100, logratio, decay, stt->vadNearend.stdLongTerm);
+    fprintf(stt->logFile,
+            "%5.2f\t%d\t%d\t%d\t",
+            (float)(stt->frameCounter) / 100,
+            logratio,
+            decay,
+            stt->vadNearend.stdLongTerm);
 #endif
     // Find max amplitude per sub frame
     // iterate over sub frames
@@ -461,10 +466,15 @@
         frac = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp32, 19); // Q12
         tmp32 = WEBRTC_SPL_MUL((stt->gainTable[zeros-1] - stt->gainTable[zeros]), frac);
         gains[k + 1] = stt->gainTable[zeros] + WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
-#ifdef AGC_DEBUG
-        if (k == 0)
-        {
-            fprintf(stt->logFile, "%d\t%d\t%d\t%d\t%d\n", env[0], cur_level, stt->capacitorFast, stt->capacitorSlow, zeros);
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+        if (k == 0) {
+          fprintf(stt->logFile,
+                  "%d\t%d\t%d\t%d\t%d\n",
+                  env[0],
+                  cur_level,
+                  stt->capacitorFast,
+                  stt->capacitorSlow,
+                  zeros);
         }
 #endif
     }
diff --git a/webrtc/modules/audio_processing/agc/digital_agc.h b/webrtc/modules/audio_processing/agc/digital_agc.h
index 6bd086f..3dcd937 100644
--- a/webrtc/modules/audio_processing/agc/digital_agc.h
+++ b/webrtc/modules/audio_processing/agc/digital_agc.h
@@ -11,7 +11,7 @@
 #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
+#ifdef WEBRTC_AGC_DEBUG_DUMP
 #include <stdio.h>
 #endif
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
@@ -46,9 +46,9 @@
     int16_t agcMode;
     AgcVad_t      vadNearend;
     AgcVad_t      vadFarend;
-#ifdef AGC_DEBUG
-    FILE*         logFile;
-    int           frameCounter;
+#ifdef WEBRTC_AGC_DEBUG_DUMP
+    FILE* logFile;
+    int frameCounter;
 #endif
 } DigitalAgc_t;
 
diff --git a/webrtc/modules/audio_processing/audio_processing.gypi b/webrtc/modules/audio_processing/audio_processing.gypi
index 9298f11..9bbcfae 100644
--- a/webrtc/modules/audio_processing/audio_processing.gypi
+++ b/webrtc/modules/audio_processing/audio_processing.gypi
@@ -21,6 +21,7 @@
       'variables': {
         # Outputs some low-level debug files.
         'aec_debug_dump%': 0,
+        'agc_debug_dump%': 0,
 
         # Disables the usual mode where we trust the reported system delay
         # values the AEC receives. The corresponding define is set appropriately
@@ -93,6 +94,9 @@
         ['aec_untrusted_delay_for_testing==1', {
           'defines': ['WEBRTC_UNTRUSTED_DELAY',],
         }],
+        ['agc_debug_dump==1', {
+          'defines': ['WEBRTC_AGC_DEBUG_DUMP',],
+        }],
         ['enable_protobuf==1', {
           'dependencies': ['audioproc_debug_proto'],
           'defines': ['WEBRTC_AUDIOPROC_DEBUG_DUMP'],