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

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index f5df5b3b4a5e4aa908d2a21bb0a61f7a379be476..3ae64117d2b337bb3ac5742be32362e989e5b668 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -16,6 +16,7 @@
 #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 {
 
@@ -368,7 +369,8 @@ int32_t RTPSenderAudio::SendAudio(
                            _rtpSender->Timestamp(), "seqnum",
                            _rtpSender->SequenceNumber());
     return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
-                                     -1, kAllowRetransmission,
+                                     TickTime::MillisecondTimestamp(),
+                                     kAllowRetransmission,
                                      RtpPacketSender::kHighPriority);
   }
 
@@ -476,9 +478,9 @@ RTPSenderAudio::SendTelephoneEventPacket(bool ended,
                              "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(),

[mflodman, 2015/12/04 15:25:19]

There is a risk this DTMF packet will get the same capture time as the real
audio packet on line 371. Will that be a problem?

[stefan-webrtc, 2015/12/04 15:30:22]

Not that I'm aware. The capture time is ignored for audio except for when we
compute how long the oldest packet has been in the pacer, and when we compute
the send-side delay (which isn't used by audio). This shouldn't affect sending
of dtmf or audio.

[mflodman, 2015/12/04 16:04:58]

Great, I couldn't find anything either but wanted to double check.

+            kAllowRetransmission, RtpPacketSender::kHighPriority);
         sendCount--;
 
     }while (sendCount > 0 && retVal == 0);