Add probe logging to RtcEventLog.

webrtc/voice_engine/channel.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.
  */
 
@@ skipping 90 matching lines @@
     rtc::CritScope lock(&crit_);
     if (event_log_) {
       event_log_->LogAudioSendStreamConfig(config);
     }
   }
 
   void LogRtpHeader(webrtc::PacketDirection direction,
                     webrtc::MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length) override {
+    LogRtpHeader(direction, media_type, header, packet_length,
+                 PacedPacketInfo::kNotAProbe);
+  }
+
+  void LogRtpHeader(webrtc::PacketDirection direction,
+                    webrtc::MediaType media_type,
+                    const uint8_t* header,
+                    size_t packet_length,
+                    int probe_cluster_id) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
-      event_log_->LogRtpHeader(direction, media_type, header, packet_length);
+      event_log_->LogRtpHeader(direction, media_type, header, packet_length,
+                               probe_cluster_id);
     }
   }
 
   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);
@@ skipping 26 matching lines @@
   }
 
   void LogAudioNetworkAdaptation(
       const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override {
     rtc::CritScope lock(&crit_);
     if (event_log_) {
       event_log_->LogAudioNetworkAdaptation(config);
     }
   }
 
+  void LogProbeClusterCreated(int id,
+                              int bitrate_bps,
+                              int min_probes,
+                              int min_bytes) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogProbeClusterCreated(id, bitrate_bps, min_probes,
+                                         min_bytes);
+    }
+  };
+
+  void LogProbeResultSuccess(int id, int bitrate_bps) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogProbeResultSuccess(id, bitrate_bps);
+    }
+  };
+
+  void LogProbeResultFailure(int id,
+                             ProbeFailureReason failure_reason) override {
+    rtc::CritScope lock(&crit_);
+    if (event_log_) {
+      event_log_->LogProbeResultFailure(id, failure_reason);
+    }
+  };
+
   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);
 };
@@ skipping 2870 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc