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

webrtc/modules/rtp_rtcp/source/rtp_sender.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.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 
 #include <stdlib.h>  // srand
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 
 namespace webrtc {
 
 // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
-const size_t kMaxPaddingLength = 224;
-const int kSendSideDelayWindowMs = 1000;
+static const size_t kMaxPaddingLength = 224;
+static const int kSendSideDelayWindowMs = 1000;
+static const uint32_t kAbsSendTimeFraction = 18;
 
 namespace {
 
 const size_t kRtpHeaderLength = 12;
 
 const char* FrameTypeToString(FrameType frame_type) {
   switch (frame_type) {
     case kEmptyFrame:
       return "empty";
     case kAudioFrameSpeech: return "audio_speech";
     case kAudioFrameCN: return "audio_cn";
     case kVideoFrameKey: return "video_key";
     case kVideoFrameDelta: return "video_delta";
   }
   return "";
 }
 
+// TODO(holmer): Merge this with the implementation in
+// remote_bitrate_estimator_abs_send_time.cc.
+uint32_t ConvertMsTo24Bits(int64_t time_ms) {
+  uint32_t time_24_bits =
+      static_cast<uint32_t>(
+          ((static_cast<uint64_t>(time_ms) << kAbsSendTimeFraction) + 500) /
+          1000) &
+      0x00FFFFFF;
+  return time_24_bits;
+}
 }  // namespace
 
 class BitrateAggregator {
  public:
   explicit BitrateAggregator(BitrateStatisticsObserver* bitrate_callback)
       : callback_(bitrate_callback),
         total_bitrate_observer_(*this),
         retransmit_bitrate_observer_(*this),
         ssrc_(0) {}
 
@@ skipping 1531 matching lines @@
       return;
     case ExtensionStatus::kOk:
       break;
     default:
       RTC_NOTREACHED();
   }
 
   // Update absolute send time field (convert ms to 24-bit unsigned with 18 bit
   // fractional part).
   ByteWriter<uint32_t, 3>::WriteBigEndian(rtp_packet + offset + 1,
-                                          ((now_ms << 18) / 1000) & 0x00ffffff);
+                                          ConvertMsTo24Bits(now_ms));
 }
 
 uint16_t RTPSender::UpdateTransportSequenceNumber(
     uint8_t* rtp_packet,
     size_t rtp_packet_length,
     const RTPHeader& rtp_header) const {
   size_t offset;
   CriticalSectionScoped cs(send_critsect_.get());
 
   switch (VerifyExtension(kRtpExtensionTransportSequenceNumber, rtp_packet,
@@ skipping 283 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc