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.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
 
 #include <assert.h>
 #include <stdlib.h>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/metrics.h"
 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/video/rtc_event_log.h"
 
 namespace webrtc {
 
 namespace acm2 {
 
 enum {
   kACMToneEnd = 999
 };
 
 // Maximum number of bytes in one packet (PCM16B, 20 ms packets, stereo).
@@ skipping 98 matching lines @@
       expected_in_ts_(0xD87F3F9F),
       receiver_(config),
       bitrate_logger_("WebRTC.Audio.TargetBitrateInKbps"),
       previous_pltype_(255),
       aux_rtp_header_(NULL),
       receiver_initialized_(false),
       first_10ms_data_(false),
       first_frame_(true),
       callback_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
       packetization_callback_(NULL),
-      vad_callback_(NULL) {
+      vad_callback_(NULL),
+      event_log_(nullptr) {
   if (InitializeReceiverSafe() < 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "Cannot initialize receiver");
   }
   WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created");
 }
 
 AudioCodingModuleImpl::~AudioCodingModuleImpl() {
   if (aux_rtp_header_ != NULL) {
     delete aux_rtp_header_;
@@ skipping 570 matching lines @@
 // Get 10 milliseconds of raw audio data to play out.
 // Automatic resample to the requested frequency.
 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,
                                            AudioFrame* audio_frame) {
   // GetAudio always returns 10 ms, at the requested sample rate.
   if (receiver_.GetAudio(desired_freq_hz, audio_frame) != 0) {
     WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
                  "PlayoutData failed, RecOut Failed");
     return -1;
   }
+  if (event_log_)
+    event_log_->LogDebugEvent(RtcEventLog::DebugEvent::kAudioPlayout);
 
   audio_frame->id_ = id_;
   return 0;
 }
 
 /////////////////////////////////////////
 //   Statistics
 //
 
 // TODO(turajs) change the return value to void. Also change the corresponding
@@ skipping 215 matching lines @@
 }
 
 int AudioCodingModuleImpl::EnableNack(size_t max_nack_list_size) {
   return receiver_.EnableNack(max_nack_list_size);
 }
 
 void AudioCodingModuleImpl::DisableNack() {
   receiver_.DisableNack();
 }
 
+void AudioCodingModuleImpl::SetEventLog(RtcEventLog* event_log) {
+  event_log_ = event_log;
+}
+
 std::vector<uint16_t> AudioCodingModuleImpl::GetNackList(
     int64_t round_trip_time_ms) const {
   return receiver_.GetNackList(round_trip_time_ms);
 }
 
 int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
@@ skipping 289 matching lines @@
       *channels = 1;
       break;
 #endif
     default:
       FATAL() << "Codec type " << codec_type << " not supported.";
   }
   return true;
 }
 
 }  // namespace webrtc