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

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index dc544fbe69fe01534aae27c20e843c65dd3c0802..d2e745efdb502b4f43ac8cd9c9b373d3d774bd8b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -469,7 +469,8 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
   std::map<int8_t, RtpUtility::Payload*>::iterator it =
       payload_type_map_.find(payload_type);
   if (it == payload_type_map_.end()) {
-    LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
+    LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type)
+                    << " not registered.";
     return -1;
   }
   SetSendPayloadType(payload_type);
@@ -512,7 +513,8 @@ int32_t RTPSender::SendOutgoingData(FrameType frame_type,
   }
   RtpVideoCodecTypes video_type = kRtpVideoGeneric;
   if (CheckPayloadType(payload_type, &video_type) != 0) {
-    LOG(LS_ERROR) << "Don't send data with unknown payload type.";
+    LOG(LS_ERROR) << "Don't send data with unknown payload type: "
+                  << static_cast<int>(payload_type) << ".";
     return -1;
   }
 
@@ -1005,6 +1007,8 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer,
 size_t RTPSender::TimeToSendPadding(size_t bytes) {
   if (bytes == 0)
     return 0;
+  if (audio_configured_)
+    return 0;

[stefan-webrtc, 2015/12/04 10:31:43]

Added this to make sure we don't send padding on the audio stream since I'm not
sure how well neteq would handle padding-only packets.

   {
     CriticalSectionScoped cs(send_critsect_.get());
     if (!sending_media_)