Switch voice transport to use Call and Stream instead of VoENetwork.

webrtc/voice_engine/channel.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 1468 matching lines @@
         "disabled");
     return 0;
   }
   _externalTransport = false;
   _transportPtr = NULL;
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "DeRegisterExternalTransport() all transport is disabled");
   return 0;
 }
 
-int32_t Channel::ReceivedRTPPacket(const int8_t* data,
+int32_t Channel::ReceivedRTPPacket(const uint8_t* received_packet,
                                    size_t length,
                                    const PacketTime& packet_time) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::ReceivedRTPPacket()");
 
   // Store playout timestamp for the received RTP packet
   UpdatePlayoutTimestamp(false);
 
-  const uint8_t* received_packet = reinterpret_cast<const uint8_t*>(data);
   RTPHeader header;
   if (!rtp_header_parser_->Parse(received_packet, length, &header)) {
     WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVoice, _channelId,
                  "Incoming packet: invalid RTP header");
     return -1;
   }
   header.payload_type_frequency =
       rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
   if (header.payload_type_frequency < 0)
     return -1;
@@ skipping 69 matching lines @@
   StreamStatistician* statistician =
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
     return false;
   // Check if this is a retransmission.
   int64_t min_rtt = 0;
   _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
   return !in_order && statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 
-int32_t Channel::ReceivedRTCPPacket(const int8_t* data, size_t length) {
+int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::ReceivedRTCPPacket()");
   // Store playout timestamp for the received RTCP packet
   UpdatePlayoutTimestamp(true);
 
   // Deliver RTCP packet to RTP/RTCP module for parsing
-  if (_rtpRtcpModule->IncomingRtcpPacket((const uint8_t*)data, length) == -1) {
+  if (_rtpRtcpModule->IncomingRtcpPacket(data, length) == -1) {
     _engineStatisticsPtr->SetLastError(
         VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
         "Channel::IncomingRTPPacket() RTCP packet is invalid");
   }
 
   int64_t rtt = GetRTT(true);
   if (rtt == 0) {
     // Waiting for valid RTT.
     return 0;
   }
@@ skipping 1943 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc