TrackMediaInfoMap added.

webrtc/api/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("../build/webrtc.gni")
 if (is_android) {
@@ skipping 83 matching lines @@
     "rtpsender.cc",
     "rtpsender.h",
     "rtpsenderinterface.h",
     "sctputils.cc",
     "sctputils.h",
     "statscollector.cc",
     "statscollector.h",
     "statstypes.cc",
     "statstypes.h",
     "streamcollection.h",
+    "trackmediainfomap.cc",
+    "trackmediainfomap.h",
     "videocapturertracksource.cc",
     "videocapturertracksource.h",
     "videosourceproxy.h",
     "videotrack.cc",
     "videotrack.h",
     "videotracksource.cc",
     "videotracksource.h",
     "webrtcsdp.cc",
     "webrtcsdp.h",
     "webrtcsession.cc",
@@ skipping 126 matching lines @@
       "rtpsenderreceiver_unittest.cc",
       "sctputils_unittest.cc",
       "statscollector_unittest.cc",
       "test/fakeaudiocapturemodule.cc",
       "test/fakeaudiocapturemodule.h",
       "test/fakeaudiocapturemodule_unittest.cc",
       "test/fakeconstraints.h",
       "test/fakedatachannelprovider.h",
       "test/fakeperiodicvideocapturer.h",
       "test/fakertccertificategenerator.h",
+      "test/fakertpsenderreceiver.cc",
+      "test/fakertpsenderreceiver.h",
       "test/fakevideotrackrenderer.h",
       "test/mock_datachannel.h",
       "test/mock_peerconnection.h",
       "test/mock_webrtcsession.h",
       "test/mockpeerconnectionobservers.h",
       "test/peerconnectiontestwrapper.cc",
       "test/peerconnectiontestwrapper.h",
       "test/rtcstatsobtainer.h",
       "test/testsdpstrings.h",
+      "trackmediainfomap_unittest.cc",
       "videocapturertracksource_unittest.cc",
       "videotrack_unittest.cc",
       "webrtcsdp_unittest.cc",
       "webrtcsession_unittest.cc",
     ]
 
     if (rtc_enable_sctp) {
       defines = [ "HAVE_SCTP" ]
     }
 
@@ skipping 63 matching lines @@
     public_deps = [
       ":audio_mixer_api",
     ]
 
     deps = [
       "//testing/gmock",
       "//webrtc/test:test_support",
     ]
   }
 }