Revert of Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 8287668258986fe10a57738fc6cc9cd9a93c142c..943a04572be088689cb655578001dd533703c0b7 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -19,7 +19,6 @@
 #include "webrtc/base/logging.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
-#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
@@ -59,87 +58,6 @@
 
 const int kTelephoneEventAttenuationdB = 10;
 
-class RtcEventLogProxy final : public webrtc::RtcEventLog {
- public:
-  RtcEventLogProxy() : event_log_(nullptr) {}
-
-  bool StartLogging(const std::string& file_name,
-                    int64_t max_size_bytes) override {
-    RTC_NOTREACHED();
-    return false;
-  }
-
-  bool StartLogging(rtc::PlatformFile log_file,
-                    int64_t max_size_bytes) override {
-    RTC_NOTREACHED();
-    return false;
-  }
-
-  void StopLogging() override { RTC_NOTREACHED(); }
-
-  void LogVideoReceiveStreamConfig(
-      const webrtc::VideoReceiveStream::Config& config) override {
-    rtc::CritScope lock(&crit_);
-    if (event_log_) {
-      event_log_->LogVideoReceiveStreamConfig(config);
-    }
-  }
-
-  void LogVideoSendStreamConfig(
-      const webrtc::VideoSendStream::Config& config) override {
-    rtc::CritScope lock(&crit_);
-    if (event_log_) {
-      event_log_->LogVideoSendStreamConfig(config);
-    }
-  }
-
-  void LogRtpHeader(webrtc::PacketDirection direction,
-                    webrtc::MediaType media_type,
-                    const uint8_t* header,
-                    size_t packet_length) override {
-    rtc::CritScope lock(&crit_);
-    if (event_log_) {
-      event_log_->LogRtpHeader(direction, media_type, header, packet_length);
-    }
-  }
-
-  void LogRtcpPacket(webrtc::PacketDirection direction,
-                     webrtc::MediaType media_type,
-                     const uint8_t* packet,
-                     size_t length) override {
-    rtc::CritScope lock(&crit_);
-    if (event_log_) {
-      event_log_->LogRtcpPacket(direction, media_type, packet, length);
-    }
-  }
-
-  void LogAudioPlayout(uint32_t ssrc) override {
-    rtc::CritScope lock(&crit_);
-    if (event_log_) {
-      event_log_->LogAudioPlayout(ssrc);
-    }
-  }
-
-  void LogBwePacketLossEvent(int32_t bitrate,
-                             uint8_t fraction_loss,
-                             int32_t total_packets) override {
-    rtc::CritScope lock(&crit_);
-    if (event_log_) {
-      event_log_->LogBwePacketLossEvent(bitrate, fraction_loss, total_packets);
-    }
-  }
-
-  void SetEventLog(RtcEventLog* event_log) {
-    rtc::CritScope lock(&crit_);
-    event_log_ = event_log;
-  }
-
- private:
-  rtc::CriticalSection crit_;
-  RtcEventLog* event_log_ GUARDED_BY(crit_);
-  RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogProxy);
-};
-
 class TransportFeedbackProxy : public TransportFeedbackObserver {
  public:
   TransportFeedbackProxy() : feedback_observer_(nullptr) {
@@ -562,9 +480,11 @@
 MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted(
     int32_t id,
     AudioFrame* audioFrame) {
-  unsigned int ssrc;
-  RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
-  event_log_proxy_->LogAudioPlayout(ssrc);
+  if (event_log_) {
+    unsigned int ssrc;
+    RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
+    event_log_->LogAudioPlayout(ssrc);
+  }
   // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
   bool muted;
   if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame,
@@ -751,8 +671,9 @@
 int32_t Channel::CreateChannel(Channel*& channel,
                                int32_t channelId,
                                uint32_t instanceId,
+                               RtcEventLog* const event_log,
                                const Config& config) {
-  return CreateChannel(channel, channelId, instanceId, config,
+  return CreateChannel(channel, channelId, instanceId, event_log, config,
                        CreateBuiltinAudioDecoderFactory());
 }
 
@@ -760,13 +681,15 @@
     Channel*& channel,
     int32_t channelId,
     uint32_t instanceId,
+    RtcEventLog* const event_log,
     const Config& config,
     const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                "Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
                instanceId);
 
-  channel = new Channel(channelId, instanceId, config, decoder_factory);
+  channel =
+      new Channel(channelId, instanceId, event_log, config, decoder_factory);
   if (channel == NULL) {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                  "Channel::CreateChannel() unable to allocate memory for"
@@ -825,11 +748,12 @@
 
 Channel::Channel(int32_t channelId,
                  uint32_t instanceId,
+                 RtcEventLog* const event_log,
                  const Config& config,
                  const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
     : _instanceId(instanceId),
       _channelId(channelId),
-      event_log_proxy_(new RtcEventLogProxy()),
+      event_log_(event_log),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_payload_registry_(
           new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
@@ -932,7 +856,7 @@
         seq_num_allocator_proxy_.get();
     configuration.transport_feedback_callback = feedback_observer_proxy_.get();
   }
-  configuration.event_log = &(*event_log_proxy_);
+  configuration.event_log = event_log;
 
   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
   _rtpRtcpModule->SetSendingMediaStatus(false);
@@ -3084,10 +3008,6 @@
   }
 }
 
-void Channel::SetRtcEventLog(RtcEventLog* event_log) {
-  event_log_proxy_->SetEventLog(event_log);
-}
-
 int Channel::RegisterExternalMediaProcessing(ProcessingTypes type,
                                              VoEMediaProcess& processObject) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),