Increase the allowed number of probe packets in test to please msan.

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 685 matching lines @@
     uint16_t last_sequence_number_;
   } test;
 
   RunBaseTest(&test);
 }
 
 // This test drops second RTP packet with a marker bit set, makes sure it's
 // retransmitted and renders. Retransmission SSRCs are also checked.
 void EndToEndTest::DecodesRetransmittedFrame(bool enable_rtx, bool enable_red) {
   // Must be set high enough to allow the bitrate probing to finish.
-  static const int kMinProbePackets = 30;
+  static const int kMinProbePackets = 100;
   static const int kDroppedFrameNumber = kMinProbePackets + 1;
   class RetransmissionObserver : public test::EndToEndTest,
                                  public I420FrameCallback {
    public:
     RetransmissionObserver(bool enable_rtx, bool enable_red)
         : EndToEndTest(kDefaultTimeoutMs),
           payload_type_(GetPayloadType(false, enable_red)),
           retransmission_ssrc_(enable_rtx ? kSendRtxSsrcs[0]
                                           : kVideoSendSsrcs[0]),
           retransmission_payload_type_(GetPayloadType(enable_rtx, enable_red)),
@@ skipping 2820 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc