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

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.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.
  */
 
@@ skipping 217 matching lines @@
   int EnableOpusDtx() override;
 
   int DisableOpusDtx() override;
 
   int UnregisterReceiveCodec(uint8_t payload_type) override;
 
   int EnableNack(size_t max_nack_list_size) override;
 
   void DisableNack() override;
 
+  void SetEventLog(RtcEventLog* event_log) override;
+
   std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
 
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const override;
 
  private:
   struct InputData {
     uint32_t input_timestamp;
     const int16_t* audio;
     uint16_t length_per_channel;
     uint8_t audio_channel;
@@ skipping 73 matching lines @@
   bool first_10ms_data_ GUARDED_BY(acm_crit_sect_);
 
   bool first_frame_ GUARDED_BY(acm_crit_sect_);
   uint32_t last_timestamp_ GUARDED_BY(acm_crit_sect_);
   uint32_t last_rtp_timestamp_ GUARDED_BY(acm_crit_sect_);
 
   CriticalSectionWrapper* callback_crit_sect_;
   AudioPacketizationCallback* packetization_callback_
       GUARDED_BY(callback_crit_sect_);
   ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
+
+  CriticalSectionWrapper* event_log_ptr_crit_sect_;
+  RtcEventLog* event_log_ GUARDED_BY(event_log_ptr_crit_sect_);
 };
 
 }  // namespace acm2
 
 class AudioCodingImpl : public AudioCoding {
  public:
   AudioCodingImpl(const Config& config);
   ~AudioCodingImpl() override;
 
   bool RegisterSendCodec(AudioEncoder* send_codec) override;
@@ skipping 59 matching lines @@
   int playout_frequency_hz_;
   // TODO(henrik.lundin): All members below this line are temporary and should
   // be removed after refactoring is completed.
   rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
   CodecInst current_send_codec_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_