Move ownership of RtpTransportControllerSendInterface from Call to PeerConnection.

webrtc/ortc/BUILD.gn

 # Copyright (c) 2017 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.
 
 import("../webrtc.gni")
 if (is_android) {
@@ skipping 19 matching lines @@
   ]
 
   # TODO(deadbeef): Create a separate target for the common things ORTC and
   # PeerConnection code shares, so that ortc can depend on that instead of
   # libjingle_peerconnection.
   deps = [
     "../api/audio_codecs:builtin_audio_decoder_factory",
     "../api/audio_codecs:builtin_audio_encoder_factory",
     "../base:rtc_base",
     "../base:rtc_base_approved",
+    "../call",
     "../call:call_interfaces",
     "../logging:rtc_event_log_api",
     "../media:rtc_media",
     "../media:rtc_media_base",
     "../p2p:rtc_p2p",
     "../pc:libjingle_peerconnection",
     "../pc:rtc_pc",
   ]
 
   public_deps = [
@@ skipping 39 matching lines @@
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
     }
   }
 }