Fix bug in how send timestamps are converted to 24 bits.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.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 46 matching lines @@
                               const uint8_t* packet) {
   return packet + rtp_header.headerLength;
 }
 
 size_t GetPayloadDataLength(const RTPHeader& rtp_header,
                             const size_t packet_length) {
   return packet_length - rtp_header.headerLength - rtp_header.paddingLength;
 }
 
 uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
-  return 0x00fffffful & ((time_ms << 18) / 1000);
+  return (((time_ms << 18) + 500) / 1000) & 0x00ffffff;
 }
 
 class LoopbackTransportTest : public webrtc::Transport {
  public:
   LoopbackTransportTest()
       : packets_sent_(0),
         last_sent_packet_len_(0),
         total_bytes_sent_(0),
         last_sent_packet_(nullptr) {}
 
@@ skipping 1342 matching lines @@
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
       transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[1]->data()),
       transport_.sent_packets_[1]->size(), true, &map, kSeqNum + 1,
       hdr.rotation);
 }
 }  // namespace webrtc