Added support for logging the SSRC corresponding to AudioPlayout events.

webrtc/modules/audio_coding/main/interface/audio_coding_module.h

 /*
  *  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 11 matching lines @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // forward declarations
 struct CodecInst;
 struct WebRtcRTPHeader;
 class AudioDecoder;
 class AudioEncoder;
 class AudioFrame;
-class RtcEventLog;
 class RTPFragmentationHeader;
 
 #define WEBRTC_10MS_PCM_AUDIO 960  // 16 bits super wideband 48 kHz
 
 // Callback class used for sending data ready to be packetized
 class AudioPacketizationCallback {
  public:
   virtual ~AudioPacketizationCallback() {}
 
   virtual int32_t SendData(FrameType frame_type,
@@ skipping 33 matching lines @@
       const uint8_t eventFlags,  // concealed voice flags
       const uint16_t delayMS) = 0;  // average delay in ms
 };
 
 class AudioCodingModule {
  protected:
   AudioCodingModule() {}
 
  public:
   struct Config {
-    Config()
-        : id(0),
-          neteq_config(),
-          clock(Clock::GetRealTimeClock()),
-          event_log(nullptr) {}
+    Config() : id(0), neteq_config(), clock(Clock::GetRealTimeClock()) {}
 
     int id;
     NetEq::Config neteq_config;
     Clock* clock;
-    RtcEventLog* event_log;
   };
 
   ///////////////////////////////////////////////////////////////////////////
   // Creation and destruction of a ACM.
   //
   // The second method is used for testing where a simulated clock can be
   // injected into ACM. ACM will take the ownership of the object clock and
   // delete it when destroyed.
   //
   static AudioCodingModule* Create(int id);
@@ skipping 905 matching lines @@
   virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
 
   // Returns the timing statistics for calls to Get10MsAudio.
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_