Revert of Implement AudioReceiveStream::GetStats().

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index a69be98bbb85dc1b79e4d3e12e8af9ddc99bbe88..9a036c9ede2bd8ae32c2de92151a9603401aaeec 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -123,8 +123,7 @@
 
   VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
 
-  RtcEventLog* event_log_ = nullptr;
-  VoECodec* voe_codec_ = nullptr;
+  RtcEventLog* event_log_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
@@ -143,7 +142,8 @@
       config_(config),
       network_enabled_(true),
       receive_crit_(RWLockWrapper::CreateRWLock()),
-      send_crit_(RWLockWrapper::CreateRWLock()) {
+      send_crit_(RWLockWrapper::CreateRWLock()),
+      event_log_(nullptr) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
@@ -153,11 +153,11 @@
                   config.bitrate_config.start_bitrate_bps);
   }
   if (config.voice_engine) {
-    // Keep a reference to VoECodec, so we're sure the VoiceEngine lives for the
-    // duration of the call.
-    voe_codec_ = VoECodec::GetInterface(config.voice_engine);
-    if (voe_codec_)
-      event_log_ = voe_codec_->GetEventLog();
+    VoECodec* voe_codec = VoECodec::GetInterface(config.voice_engine);
+    if (voe_codec) {
+      event_log_ = voe_codec->GetEventLog();
+      voe_codec->Release();
+    }
   }
 
   Trace::CreateTrace();
@@ -179,9 +179,6 @@
 
   module_process_thread_->Stop();
   Trace::ReturnTrace();
-
-  if (voe_codec_)
-    voe_codec_->Release();
 }
 
 PacketReceiver* Call::Receiver() {
@@ -232,8 +229,7 @@
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
-      channel_group_->GetRemoteBitrateEstimator(false), config,
-      config_.voice_engine);
+      channel_group_->GetRemoteBitrateEstimator(false), config);
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==