Added log messages for important call setup events: first packet sent/received

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h

 /*
  *  Copyright (c) 2013 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 @@
 
 class RtpReceiverImpl : public RtpReceiver {
  public:
   // Callbacks passed in here may not be NULL (use Null Object callbacks if you
   // want callbacks to do nothing). This class takes ownership of the media
   // receiver but nothing else.
   RtpReceiverImpl(Clock* clock,
                   RtpAudioFeedback* incoming_audio_messages_callback,
                   RtpFeedback* incoming_messages_callback,
                   RTPPayloadRegistry* rtp_payload_registry,
-                  RTPReceiverStrategy* rtp_media_receiver);
+                  RTPReceiverStrategy* rtp_media_receiver,
+                  bool is_audio);
 
   virtual ~RtpReceiverImpl();
 
   int32_t RegisterReceivePayload(const char payload_name[RTP_PAYLOAD_NAME_SIZE],
                                  const int8_t payload_type,
                                  const uint32_t frequency,
                                  const size_t channels,
                                  const uint32_t rate) override;
 
   int32_t DeRegisterReceivePayload(const int8_t payload_type) override;
@@ skipping 27 matching lines @@
   void CheckSSRCChanged(const RTPHeader& rtp_header);
   void CheckCSRC(const WebRtcRTPHeader& rtp_header);
   int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
                               const int8_t first_payload_byte,
                               bool* is_red,
                               PayloadUnion* payload);
 
   Clock* clock_;
   RTPPayloadRegistry* rtp_payload_registry_;
   rtc::scoped_ptr<RTPReceiverStrategy> rtp_media_receiver_;
+  bool is_audio_;
 
   RtpFeedback* cb_rtp_feedback_;
 
   rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtp_receiver_;
   int64_t last_receive_time_;
   size_t last_received_payload_length_;
 
   // SSRCs.
   uint32_t ssrc_;
   uint8_t num_csrcs_;
   uint32_t current_remote_csrc_[kRtpCsrcSize];
 
   uint32_t last_received_timestamp_;
   int64_t last_received_frame_time_ms_;
   uint16_t last_received_sequence_number_;
 
   NACKMethod nack_method_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_