Tried to reproduce the issue with AudioTrackSinkInterface::OnData

webrtc/examples/peerconnection/client/conductor.cc

 /*
  *  Copyright 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/examples/peerconnection/client/conductor.h"
 
+#include <iostream>
 #include <memory>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/test/fakeconstraints.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/json.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/examples/peerconnection/client/defaults.h"
 #include "webrtc/media/engine/webrtcvideocapturerfactory.h"
 #include "webrtc/modules/video_capture/video_capture_factory.h"
 
 // Names used for a IceCandidate JSON object.
 const char kCandidateSdpMidName[] = "sdpMid";
 const char kCandidateSdpMlineIndexName[] = "sdpMLineIndex";
 const char kCandidateSdpName[] = "candidate";
 
 // Names used for a SessionDescription JSON object.
 const char kSessionDescriptionTypeName[] = "type";
 const char kSessionDescriptionSdpName[] = "sdp";
 
 #define DTLS_ON  true
 #define DTLS_OFF false
 
+class MyAudioSink : public webrtc::AudioTrackSinkInterface {
+ public:
+  MyAudioSink() { std::cout << "[Sink] Constructing new sink!" << std::endl; }
+  void OnData(const void* audio_data,
+              int bits_per_sample,
+              int sample_rate,
+              size_t number_of_channels,
+              size_t number_of_frames) {
+    std::cout << "[Sink] bits per sample: " << bits_per_sample << "\n"
+              << "[Sink] sample rate: " << sample_rate << "\n"
+              << "[Sink] #channels: " << number_of_channels << "\n"
+              << "[Sink] #frames: " << number_of_frames
+              << std::endl;
+  }
+};
+
 class DummySetSessionDescriptionObserver
     : public webrtc::SetSessionDescriptionObserver {
  public:
   static DummySetSessionDescriptionObserver* Create() {
     return
         new rtc::RefCountedObject<DummySetSessionDescriptionObserver>();
   }
   virtual void OnSuccess() {
     LOG(INFO) << __FUNCTION__;
   }
@@ skipping 439 matching lines @@
     case NEW_STREAM_ADDED: {
       webrtc::MediaStreamInterface* stream =
           reinterpret_cast<webrtc::MediaStreamInterface*>(
           data);
       webrtc::VideoTrackVector tracks = stream->GetVideoTracks();
       // Only render the first track.
       if (!tracks.empty()) {
         webrtc::VideoTrackInterface* track = tracks[0];
         main_wnd_->StartRemoteRenderer(track);
       }
+
+      webrtc::AudioTrackVector audio_tracks = stream->GetAudioTracks();
+      for (auto & audio_track : audio_tracks) {
+        audio_track->AddSink(new MyAudioSink());
+      }
+
       stream->Release();
       break;
     }
 
     case STREAM_REMOVED: {
       // Remote peer stopped sending a stream.
       webrtc::MediaStreamInterface* stream =
           reinterpret_cast<webrtc::MediaStreamInterface*>(
           data);
       stream->Release();
@@ skipping 31 matching lines @@
 }
 
 void Conductor::OnFailure(const std::string& error) {
     LOG(LERROR) << error;
 }
 
 void Conductor::SendMessage(const std::string& json_object) {
   std::string* msg = new std::string(json_object);
   main_wnd_->QueueUIThreadCallback(SEND_MESSAGE_TO_PEER, msg);
 }