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

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.
  */
 
@@ skipping 451 matching lines @@
   }
   if (payload_type_ == payload_type) {
     if (!audio_configured_) {
       *video_type = video_->VideoCodecType();
     }
     return 0;
   }
   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);
   RtpUtility::Payload* payload = it->second;
   assert(payload);
   if (!payload->audio && !audio_configured_) {
     video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
     *video_type = payload->typeSpecific.Video.videoCodecType;
     video_->SetMaxConfiguredBitrateVideo(payload->typeSpecific.Video.maxRate);
   }
@@ skipping 22 matching lines @@
   {
     // Drop this packet if we're not sending media packets.
     CriticalSectionScoped cs(send_critsect_.get());
     ssrc = ssrc_;
     if (!sending_media_) {
       return 0;
     }
   }
   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;
   }
 
   int32_t ret_val;
   if (audio_configured_) {
     TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", capture_timestamp,
                             "Send", "type", FrameTypeToString(frame_type));
     assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
            frame_type == kEmptyFrame);
 
@@ skipping 192 matching lines @@
     RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length);
     RTPHeader header;
     if (!rtp_parser.Parse(header)) {
       assert(false);
       return -1;
     }
     // Convert from TickTime to Clock since capture_time_ms is based on
     // TickTime.
     int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
     paced_sender_->InsertPacket(
-        RtpPacketSender::kHighPriority, header.ssrc, header.sequenceNumber,
+        RtpPacketSender::kNormalPriority, header.ssrc, header.sequenceNumber,
         corrected_capture_tims_ms, length - header.headerLength, true);
 
     return length;
   }
   int rtx = kRtxOff;
   {
     CriticalSectionScoped lock(send_critsect_.get());
     rtx = rtx_;
   }
   if (!PrepareAndSendPacket(data_buffer, length, capture_time_ms,
@@ skipping 259 matching lines @@
   uint8_t pt_fec;
   video_->GenericFECStatus(fec_enabled, pt_red, pt_fec);
   return fec_enabled &&
       header.payloadType == pt_red &&
       buffer[header.headerLength] == pt_fec;
 }
 
 size_t RTPSender::TimeToSendPadding(size_t bytes) {
   if (bytes == 0)
     return 0;
+  if (audio_configured_)
+    return 0;
   {
     CriticalSectionScoped cs(send_critsect_.get());
     if (!sending_media_)
       return 0;
   }
   size_t bytes_sent = TrySendRedundantPayloads(bytes);
   if (bytes_sent < bytes)
     bytes_sent += SendPadData(bytes - bytes_sent, false, 0, 0);
   return bytes_sent;
 }
@@ skipping 885 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc