Expanded error message for unexpected packet

webrtc/video/end_to_end_tests.cc

 /*
  *  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.
  */
 #include <algorithm>
@@ skipping 772 matching lines @@
       }
 
       if (header.timestamp == retransmitted_timestamp_) {
         EXPECT_EQ(retransmission_ssrc_, header.ssrc);
         EXPECT_EQ(retransmission_payload_type_, header.payloadType);
         frame_retransmitted_ = true;
         return SEND_PACKET;
       }
 
       EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc)
-          << "Payload type " << static_cast<int>(header.payloadType)
-          << " not expected.";
+          << "Unexpected packet length " << length
+          << ", header_length " << header.headerLength
+          << ", padding_length " << header.paddingLength
+          << ", timestamp " << header.timestamp
+          << ", expected timestamp " << retransmitted_timestamp_
+          << ", payload type " << static_cast<int>(header.payloadType);
       EXPECT_EQ(payload_type_, header.payloadType);
 
       // Found the final packet of the frame to inflict loss to, drop this and
       // expect a retransmission.
       if (header.markerBit && ++marker_bits_observed_ == kDroppedFrameNumber) {
+        // This should be the only dropped packet.
+        EXPECT_EQ(0u, retransmitted_timestamp_);
         retransmitted_timestamp_ = header.timestamp;
         return DROP_PACKET;
       }
 
       return SEND_PACKET;
     }
 
     void FrameCallback(VideoFrame* frame) override {
       rtc::CritScope lock(&crit_);
       if (frame->timestamp() == retransmitted_timestamp_) {
@@ skipping 2747 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc