Revert of Revert Make the new jitter buffer the default jitter buffer.

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 1259 matching lines @@
   } test(rtp_history_ms);
 
   RunBaseTest(&test);
 }
 
 TEST_P(EndToEndTest, ReceivesPliAndRecoversWithNack) {
   ReceivesPliAndRecovers(1000);
 }
 
 TEST_P(EndToEndTest, ReceivesPliAndRecoversWithoutNack) {
-  // This test makes no sense for the new video jitter buffer.
-  if (GetParam() == new_jb_enabled)
-    return;
   ReceivesPliAndRecovers(0);
 }
 
 TEST_P(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
   class PacketInputObserver : public PacketReceiver {
    public:
     explicit PacketInputObserver(PacketReceiver* receiver)
         : receiver_(receiver), delivered_packet_(false, false) {}
 
     bool Wait() { return delivered_packet_.Wait(kDefaultTimeoutMs); }
@@ skipping 1801 matching lines @@
     std::map<std::string, bool> send_stats_filled_;
 
     std::vector<uint32_t> expected_receive_ssrcs_;
     std::set<uint32_t> expected_send_ssrcs_;
     std::string expected_cname_;
 
     rtc::Event check_stats_event_;
     ReceiveStreamRenderer receive_stream_renderer_;
   } test;
 
-  // TODO(philipel): Implement statistics for the new video jitter buffer.
-  if (GetParam() == new_jb_enabled)
-    return;
-
   RunBaseTest(&test);
 }
 
 class RtcpXrObserver : public test::EndToEndTest {
  public:
   RtcpXrObserver(bool enable_rrtr, bool enable_target_bitrate)
       : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
         enable_rrtr_(enable_rrtr),
         enable_target_bitrate_(enable_target_bitrate),
         sent_rtcp_sr_(0),
@@ skipping 1072 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc