Re-enable RestartingSendStreamPreservesRtpState.

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 2719 matching lines @@
 
       EXPECT_TRUE(configured_ssrcs_[ssrc])
           << "Received SSRC that wasn't configured: " << ssrc;
 
       std::map<uint32_t, uint16_t>::iterator it =
           last_observed_sequence_number_.find(header.ssrc);
       if (it == last_observed_sequence_number_.end()) {
         last_observed_sequence_number_[ssrc] = sequence_number;
         last_observed_timestamp_[ssrc] = timestamp;
       } else {
+        // We shouldn't get replays of previous sequence numbers.
+        EXPECT_NE(sequence_number, last_observed_sequence_number_[ssrc]);
         // Verify sequence numbers are reasonably close.
         uint32_t extended_sequence_number = sequence_number;
         // Check for roll-over.
         if (sequence_number < last_observed_sequence_number_[ssrc])
           extended_sequence_number += 0xFFFFu + 1;
         EXPECT_LE(
             extended_sequence_number - last_observed_sequence_number_[ssrc],
             kMaxSequenceNumberGap)
             << "Gap in sequence numbers ("
             << last_observed_sequence_number_[ssrc] << " -> " << sequence_number
@@ skipping 123 matching lines @@
         << "Timed out waiting for all SSRCs to send packets.";
   }
 
   send_transport.StopSending();
   receive_transport.StopSending();
 
   Stop();
   DestroyStreams();
 }
 
-TEST_F(EndToEndTest, DISABLED_RestartingSendStreamPreservesRtpState) {
+TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpState) {
   TestRtpStatePreservation(false);
 }
 
 TEST_F(EndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) {
   TestRtpStatePreservation(true);
 }
 
 TEST_F(EndToEndTest, RespectsNetworkState) {
   // TODO(pbos): Remove accepted downtime packets etc. when signaling network
   // down blocks until no more packets will be sent.
@@ skipping 297 matching lines @@
   EXPECT_TRUE(default_receive_config.rtp.rtx.empty())
       << "Enabling RTX requires rtpmap: rtx negotiation.";
   EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
       << "Enabling RTP extensions require negotiation.";
 
   VerifyEmptyNackConfig(default_receive_config.rtp.nack);
   VerifyEmptyFecConfig(default_receive_config.rtp.fec);
 }
 
 }  // namespace webrtc