Prepare the AudioSendStream to be hooked up to send-side BWE.

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.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_audio.h"
 
 #include <assert.h> //assert
 #include <string.h> //memcpy
 
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
 
 namespace webrtc {
 
 static const int kDtmfFrequencyHz = 8000;
 
 RTPSenderAudio::RTPSenderAudio(Clock* clock,
                                RTPSender* rtpSender,
                                RtpAudioFeedback* audio_feedback)
     : _clock(clock),
       _rtpSender(rtpSender),
@@ skipping 332 matching lines @@
     RtpUtility::RtpHeaderParser rtp_parser(dataBuffer, packetSize);
     RTPHeader rtp_header;
     rtp_parser.Parse(rtp_header);
     _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
                                  (frameType == kAudioFrameSpeech),
                                  audio_level_dbov);
     TRACE_EVENT_ASYNC_END2("webrtc", "Audio", captureTimeStamp, "timestamp",
                            _rtpSender->Timestamp(), "seqnum",
                            _rtpSender->SequenceNumber());
     return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
-                                     -1, kAllowRetransmission,
+                                     TickTime::MillisecondTimestamp(),
+                                     kAllowRetransmission,
                                      RtpPacketSender::kHighPriority);
   }
 
     // Audio level magnitude and voice activity flag are set for each RTP packet
 int32_t
 RTPSenderAudio::SetAudioLevel(const uint8_t level_dBov)
 {
     if (level_dBov > 127)
     {
         return -1;
@@ skipping 87 matching lines @@
 
         // First byte is Event number, equals key number
         dtmfbuffer[12] = _dtmfKey;
         dtmfbuffer[13] = E|R|volume;
         ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 14, duration);
 
         TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                              "Audio::SendTelephoneEvent", "timestamp",
                              dtmfTimeStamp, "seqnum",
                              _rtpSender->SequenceNumber());
-        retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12, -1,
-                                           kAllowRetransmission,
-                                           RtpPacketSender::kHighPriority);
+        retVal = _rtpSender->SendToNetwork(
+            dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
+            kAllowRetransmission, RtpPacketSender::kHighPriority);
         sendCount--;
 
     }while (sendCount > 0 && retVal == 0);
 
     return retVal;
 }
 }  // namespace webrtc