Move ready to send logic from BaseChannel to RtpTransport.

webrtc/pc/BUILD.gn

 # Copyright (c) 2016 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 173 matching lines @@
   rtc_test("rtc_pc_unittests") {
     testonly = true
 
     sources = [
       "bundlefilter_unittest.cc",
       "channel_unittest.cc",
       "channelmanager_unittest.cc",
       "currentspeakermonitor_unittest.cc",
       "mediasession_unittest.cc",
       "rtcpmuxfilter_unittest.cc",
+      "rtptransport_unittest.cc",
       "srtpfilter_unittest.cc",
     ]
 
     include_dirs = [ "//third_party/libsrtp/srtp" ]
 
     configs += [ ":rtc_pc_unittests_config" ]
 
     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" ]
@@ skipping 159 matching lines @@
       "//testing/gmock",
     ]
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_support" ]
 
       shard_timeout = 900
     }
   }
 }