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

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 732 matching lines @@
     } else {
       video_stream->SetSyncChannel(voice_engine(), -1);
     }
   }
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
                                                  const uint8_t* packet,
                                                  size_t length) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
-  // TODO(pbos): Figure out what channel needs it actually.
-  //             Do NOT broadcast! Also make sure it's a valid packet.
+  // TODO(pbos): Make sure it's a valid packet.
   //             Return DELIVERY_UNKNOWN_SSRC if it can be determined that
   //             there's no receiver of the packet.
   received_rtcp_bytes_ += length;
   bool rtcp_delivered = false;
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     ReadLockScoped read_lock(*receive_crit_);
     for (VideoReceiveStream* stream : video_receive_streams_) {
-      if (stream->DeliverRtcp(packet, length)) {
+      if (stream->DeliverRtcp(packet, length))
         rtcp_delivered = true;
-        if (event_log_)
-          event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet,
-                                    length);
-      }
+    }
+  }
+  if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
+    ReadLockScoped read_lock(*receive_crit_);
+    for (auto& kv : audio_receive_ssrcs_) {
+      if (kv.second->DeliverRtcp(packet, length))
+        rtcp_delivered = true;
     }
   }
   if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
     ReadLockScoped read_lock(*send_crit_);
     for (VideoSendStream* stream : video_send_streams_) {
-      if (stream->DeliverRtcp(packet, length)) {
+      if (stream->DeliverRtcp(packet, length))
         rtcp_delivered = true;
-        if (event_log_)
-          event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet,
-                                    length);
-      }
     }
   }
+  if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
+    ReadLockScoped read_lock(*send_crit_);
+    for (auto& kv : audio_send_ssrcs_) {
+      if (kv.second->DeliverRtcp(packet, length))
+        rtcp_delivered = true;
+    }
+  }
+
+  if (event_log_ && rtcp_delivered)
+    event_log_->LogRtcpPacket(kIncomingPacket, media_type, packet, length);
+
   return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                 const uint8_t* packet,
                                                 size_t length,
                                                 const PacketTime& packet_time) {
   TRACE_EVENT0("webrtc", "Call::DeliverRtp");
   // Minimum RTP header size.
   if (length < 12)
@@ skipping 42 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc