New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

webrtc/call/BUILD.gn

 # Copyright (c) 2015 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")
 
 rtc_source_set("call_interfaces") {
   sources = [
     "audio_receive_stream.h",
     "audio_send_stream.cc",
     "audio_send_stream.h",
     "audio_state.h",
     "call.h",
     "flexfec_receive_stream.h",
-    "rtp_demuxer.h",
-    "rtp_transport_controller_send_interface.h",
     "syncable.cc",
     "syncable.h",
   ]
   deps = [
+    ":rtp_interfaces",
     "..:video_stream_api",
     "..:webrtc_common",
     "../api:audio_mixer_api",
     "../api:libjingle_peerconnection_api",
     "../api:transport_api",
     "../api/audio_codecs:audio_codecs_api",
     "../base:rtc_base",
     "../base:rtc_base_approved",
   ]
 }
 
+# TODO(nisse): These RTP targets should be moved elsewhere
+# when interfaces have stabilized.
+rtc_source_set("rtp_interfaces") {
+  sources = [
+    "rtp_packet_sink_interface.h",
+    "rtp_transport_controller_send_interface.h",
+  ]
+}
+
+rtc_source_set("rtp_receiver") {
+  sources = [
+    "rtp_demuxer.cc",
+    "rtp_demuxer.h",
+    "rtx_receive_stream.cc",
+    "rtx_receive_stream.h",
+  ]
+  deps = [
+    ":rtp_interfaces",
+    "../base:rtc_base_approved",
+    "../modules/rtp_rtcp",
+  ]
+}
+
+rtc_source_set("rtp_sender") {
+  sources = [
+    "rtp_transport_controller_send.cc",
+    "rtp_transport_controller_send.h",
+  ]
+  deps = [
+    ":rtp_interfaces",
+    "../base:rtc_base_approved",
+    "../modules/congestion_controller",
+  ]
+}
+
 rtc_static_library("call") {
   sources = [
     "bitrate_allocator.cc",
     "call.cc",
     "flexfec_receive_stream_impl.cc",
     "flexfec_receive_stream_impl.h",
-    "rtp_demuxer.cc",
-    "rtp_transport_controller_send.cc",
-    "rtp_transport_controller_send.h",
-    "rtx_receive_stream.cc",
-    "rtx_receive_stream.h",
   ]
 
   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" ]
   }
 
   public_deps = [
     ":call_interfaces",
     "../api:call_api",
   ]
 
   deps = [
     ":call_interfaces",
+    ":rtp_interfaces",
+    ":rtp_receiver",
+    ":rtp_sender",
     "..:webrtc_common",
     "../api:transport_api",
     "../audio",
     "../base:rtc_task_queue",
     "../logging:rtc_event_log_api",
     "../logging:rtc_event_log_impl",
     "../modules/bitrate_controller",
     "../modules/congestion_controller",
     "../modules/pacing",
     "../modules/rtp_rtcp",
@@ skipping 16 matching lines @@
     sources = [
       "bitrate_allocator_unittest.cc",
       "bitrate_estimator_tests.cc",
       "call_unittest.cc",
       "flexfec_receive_stream_unittest.cc",
       "rtp_demuxer_unittest.cc",
       "rtx_receive_stream_unittest.cc",
     ]
     deps = [
       ":call",
+      ":rtp_interfaces",
+      ":rtp_receiver",
+      ":rtp_sender",
       "../api:mock_audio_mixer",
       "../base:rtc_base_approved",
       "../logging:rtc_event_log_api",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer",
       "../modules/bitrate_controller",
       "../modules/congestion_controller:mock_congestion_controller",
       "../modules/pacing",
       "../modules/rtp_rtcp",
       "../modules/rtp_rtcp:mock_rtp_rtcp",
@@ skipping 46 matching lines @@
       "//testing/gtest",
       "//webrtc/test:field_trial",
       "//webrtc/test:test_common",
     ]
     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" ]
     }
   }
 }