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

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.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 18 matching lines @@
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
   if (!incoming_payload_callback)
     incoming_payload_callback = NullObjectRtpData();
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
       clock, NullObjectRtpAudioFeedback(), incoming_messages_callback,
       rtp_payload_registry,
-      RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
+      RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback),
+      false /*is_audio*/);
 }
 
 RtpReceiver* RtpReceiver::CreateAudioReceiver(
     Clock* clock,
     RtpAudioFeedback* incoming_audio_feedback,
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
   if (!incoming_audio_feedback)
     incoming_audio_feedback = NullObjectRtpAudioFeedback();
   if (!incoming_payload_callback)
     incoming_payload_callback = NullObjectRtpData();
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
       clock, incoming_audio_feedback, incoming_messages_callback,
       rtp_payload_registry,
       RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback,
-                                               incoming_audio_feedback));
+                                               incoming_audio_feedback),
+      true /*is_audio*/);
 }
 
 RtpReceiverImpl::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)
     : clock_(clock),
       rtp_payload_registry_(rtp_payload_registry),
       rtp_media_receiver_(rtp_media_receiver),
+      is_audio_(is_audio),
       cb_rtp_feedback_(incoming_messages_callback),
       critical_section_rtp_receiver_(
           CriticalSectionWrapper::CreateCriticalSection()),
       last_receive_time_(0),
       last_received_payload_length_(0),
       ssrc_(0),
       num_csrcs_(0),
       current_remote_csrc_(),
       last_received_timestamp_(0),
       last_received_frame_time_ms_(-1),
@@ skipping 98 matching lines @@
     return false;
   }
 
   WebRtcRTPHeader webrtc_rtp_header;
   memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
   webrtc_rtp_header.header = rtp_header;
   CheckCSRC(webrtc_rtp_header);
 
   size_t payload_data_length = payload_length - rtp_header.paddingLength;
 
+
   bool is_first_packet_in_frame = false;
   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
     if (HaveReceivedFrame()) {
       is_first_packet_in_frame =
           last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
           last_received_timestamp_ != rtp_header.timestamp;
     } else {
       is_first_packet_in_frame = true;
     }
   }
 
   int32_t ret_val = rtp_media_receiver_->ParseRtpPacket(
       &webrtc_rtp_header, payload_specific, is_red, payload, payload_length,
       clock_->TimeInMilliseconds(), is_first_packet_in_frame);
 
   if (ret_val < 0) {
     return false;
   }
 
+  bool is_first_packet_in_first_frame = false;
   {
     CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
 
+    is_first_packet_in_first_frame = (last_receive_time_ == 0);
     last_receive_time_ = clock_->TimeInMilliseconds();
     last_received_payload_length_ = payload_data_length;
 
     if (in_order) {
       if (last_received_timestamp_ != rtp_header.timestamp) {
         last_received_timestamp_ = rtp_header.timestamp;
         last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
       }
       last_received_sequence_number_ = rtp_header.sequenceNumber;
     }
   }
+
+  if (is_first_packet_in_first_frame) {
+    LOG(LS_INFO) << "First RTP packet of the first " <<
+      (is_audio_ ? "audio" : "video") << " frame received";

[pthatcher1, 2016/03/05 01:17:05]

Nit: I think I'd prefer the text to start with "Received": "Received first RTP
packet of the first X frame."

[skvlad, 2016/03/07 19:36:53]

Done.

+  }
+
   return true;
 }
 
 TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
   return rtp_media_receiver_->GetTelephoneEventHandler();
 }
 
 bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
   CriticalSectionScoped lock(critical_section_rtp_receiver_.get());
   if (!HaveReceivedFrame())
@@ skipping 246 matching lines @@
     // implementations might have CSRC 0 as a valid value.
     if (num_csrcs_diff > 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
     } else if (num_csrcs_diff < 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
     }
   }
 }
 
 }  // namespace webrtc