Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.

webrtc/modules/audio_coding/main/interface/audio_coding_module.h

 /*
  *  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_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_
 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_
 
 #include <vector>
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/neteq/interface/neteq.h"
 #include "webrtc/modules/interface/module.h"
 #include "webrtc/system_wrappers/interface/clock.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // forward declarations
 struct CodecInst;
 struct WebRtcRTPHeader;
+class AudioEncoderMutable;
 class AudioFrame;
+class RtcEventLog;
 class RTPFragmentationHeader;
-class AudioEncoderMutable;
 
 #define WEBRTC_10MS_PCM_AUDIO 960  // 16 bits super wideband 48 kHz
 
 // Callback class used for sending data ready to be packetized
 class AudioPacketizationCallback {
  public:
   virtual ~AudioPacketizationCallback() {}
 
   virtual int32_t SendData(FrameType frame_type,
                            uint8_t payload_type,
@@ skipping 929 matching lines @@
   //
   // |max_nack_list_size| should be positive (none zero) and less than or
   // equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1
   // is returned. 0 is returned at success.
   //
   virtual int EnableNack(size_t max_nack_list_size) = 0;
 
   // Disable NACK.
   virtual void DisableNack() = 0;
 
+  ///////////////////////////////////////////////////////////////////////////
+  // void SetEventLog(RtcEventLog* event_log)
+  //
+  // Set an RtcEventLog object to enable logging of debug events inside the
+  // audio coding module.
+  //
+  // Input:
+  //   -event_log         : pointer to logging object.
+  virtual void SetEventLog(RtcEventLog* event_log) = 0;
+
   //
   // Get a list of packets to be retransmitted. |round_trip_time_ms| is an
   // estimate of the round-trip-time (in milliseconds). Missing packets which
   // will be playout in a shorter time than the round-trip-time (with respect
   // to the time this API is called) will not be included in the list.
   //
   // Negative |round_trip_time_ms| results is an error message and empty list
   // is returned.
   //
   virtual std::vector<uint16_t> GetNackList(
@@ skipping 171 matching lines @@
   virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
 
   // Returns the timing statistics for calls to Get10MsAudio.
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_