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 4dd231b723c8794bc1d739ab0c05eee2d2b3196a..dfefdc1b4ec8ed1ad6fd041844c6d642c1682906 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -21,6 +21,7 @@
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
+#include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
@@ -45,6 +46,88 @@ namespace voe {
 
 const int kTelephoneEventAttenuationdB = 10;
 
+class RtcEventLogProxy final : public webrtc::RtcEventLog {
+ public:
+  RtcEventLogProxy() : event_log_(nullptr) {}
+
+  void SetBufferDuration(int64_t buffer_duration_us) override {
+    RTC_NOTREACHED();
+  }
+
+  void StartLogging(const std::string& file_name, int duration_ms) override {
+    RTC_NOTREACHED();
+  }
+
+  bool StartLogging(rtc::PlatformFile log_file,
+                    int64_t max_size_bytes) override {
+    RTC_NOTREACHED();
+  }
+
+  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) {
@@ -465,11 +548,9 @@ bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet,
 }
 
 int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame) {
-  if (event_log_) {
-    unsigned int ssrc;
-    RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
-    event_log_->LogAudioPlayout(ssrc);
-  }
+  unsigned int ssrc;
+  RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0);
+  event_log_proxy_->LogAudioPlayout(ssrc);
   // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
   if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame) ==
       -1) {
@@ -645,13 +726,12 @@ int32_t Channel::NeededFrequency(int32_t id) const {
 int32_t Channel::CreateChannel(Channel*& channel,
                                int32_t channelId,
                                uint32_t instanceId,
-                               RtcEventLog* const event_log,
                                const Config& config) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                "Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
                instanceId);
 
-  channel = new Channel(channelId, instanceId, event_log, config);
+  channel = new Channel(channelId, instanceId, config);
   if (channel == NULL) {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                  "Channel::CreateChannel() unable to allocate memory for"
@@ -708,13 +788,10 @@ void Channel::RecordFileEnded(int32_t id) {
                " shutdown");
 }
 
-Channel::Channel(int32_t channelId,
-                 uint32_t instanceId,
-                 RtcEventLog* const event_log,
-                 const Config& config)
+Channel::Channel(int32_t channelId, uint32_t instanceId, const Config& config)
     : _instanceId(instanceId),
       _channelId(channelId),
-      event_log_(event_log),
+      event_log_proxy_(new RtcEventLogProxy()),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_payload_registry_(
           new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
@@ -815,7 +892,7 @@ Channel::Channel(int32_t channelId,
         seq_num_allocator_proxy_.get();
     configuration.transport_feedback_callback = feedback_observer_proxy_.get();
   }
-  configuration.event_log = event_log;
+  configuration.event_log = &(*event_log_proxy_);
 
   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
   _rtpRtcpModule->SetSendingMediaStatus(false);
@@ -3022,6 +3099,10 @@ void Channel::DisassociateSendChannel(int channel_id) {
   }
 }
 
+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),