Initial implementation of RtpTransportControllerReceive and related interfaces.

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 1673 matching lines @@
   if (header->payload_type_frequency < 0)
     return false;
   bool in_order = IsPacketInOrder(*header);
   rtp_receive_statistics_->IncomingPacket(
       *header, length, IsPacketRetransmitted(*header, in_order));
   rtp_payload_registry_->SetIncomingPayloadType(*header);
 
   return ReceivePacket(received_packet, length, *header, in_order);
 }
 
+// TODO(nisse): Delete, as soon as the VoENetwork code is gone.
 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()");
 
   RTPHeader header;
   if (!rtp_header_parser_->Parse(received_packet, length, &header)) {
     WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVoice, _channelId,
                  "Incoming packet: invalid RTP header");
@@ skipping 720 matching lines @@
 int Channel::GetRemoteSSRC(unsigned int& ssrc) {
   ssrc = rtp_receiver_->SSRC();
   return 0;
 }
 
 int Channel::SetSendAudioLevelIndicationStatus(bool enable, unsigned char id) {
   _includeAudioLevelIndication = enable;
   return SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
 }
 
-int Channel::SetReceiveAudioLevelIndicationStatus(bool enable,
-                                                  unsigned char id) {
-  rtp_header_parser_->DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel);
-  if (enable &&
-      !rtp_header_parser_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
-                                                      id)) {
-    return -1;
-  }
-  return 0;
-}
-
 void Channel::EnableSendTransportSequenceNumber(int id) {
   int ret =
       SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id);
   RTC_DCHECK_EQ(0, ret);
 }
 
-void Channel::EnableReceiveTransportSequenceNumber(int id) {
-  rtp_header_parser_->DeregisterRtpHeaderExtension(
-      kRtpExtensionTransportSequenceNumber);
-  bool ret = rtp_header_parser_->RegisterRtpHeaderExtension(
-      kRtpExtensionTransportSequenceNumber, id);
-  RTC_DCHECK(ret);
-}
-
 void Channel::RegisterSenderCongestionControlObjects(
     RtpTransportControllerSendInterface* transport,
     RtcpBandwidthObserver* bandwidth_observer) {
   RtpPacketSender* rtp_packet_sender = transport->packet_sender();
   TransportFeedbackObserver* transport_feedback_observer =
       transport->transport_feedback_observer();
   PacketRouter* packet_router = transport->packet_router();
 
   RTC_DCHECK(rtp_packet_sender);
   RTC_DCHECK(transport_feedback_observer);
@@ skipping 639 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