Move video send/receive stream headers to webrtc/call.

webrtc/BUILD.gn

 # Copyright (c) 2014 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.
 
 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
 
 import("//build/config/linux/pkg_config.gni")
 import("//build/config/sanitizers/sanitizers.gni")
 import("webrtc.gni")
 import("//third_party/protobuf/proto_library.gni")
 if (is_android) {
   import("//build/config/android/config.gni")
   import("//build/config/android/rules.gni")
 }
 
 # Contains the defines and includes in common.gypi that are duplicated both as
 # target_defaults and direct_dependent_settings.
 config("common_inherited_config") {
   defines = []
   cflags = []
   ldflags = []
   if (build_with_mozilla) {
     defines += [ "WEBRTC_MOZILLA_BUILD" ]
   }
+

[aleloi, 2017/08/22 09:04:03]

Blank line added by 'git cl format'

[kwiberg-webrtc, 2017/08/22 11:21:15]

Acknowledged.

   # Some tests need to declare their own trace event handlers. If this define is
   # not set, the first time TRACE_EVENT_* is called it will store the return
   # value for the current handler in an static variable, so that subsequent
   # changes to the handler for that TRACE_EVENT_* will be ignored.
   # So when tests are included, we set this define, making it possible to use
   # different event handlers in different tests.
   if (rtc_include_tests) {
     defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=1" ]
   } else {
     defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=0" ]
@@ skipping 189 matching lines @@
   if (use_libfuzzer || use_drfuzz || use_afl) {
     # Used in Chromium's overrides to disable logging
     defines += [ "WEBRTC_UNSAFE_FUZZER_MODE" ]
   }
 }
 
 config("common_objc") {
   libs = [ "Foundation.framework" ]
 }
 
-rtc_source_set("video_stream_api") {
-  sources = [
-    "video_receive_stream.h",
-    "video_send_stream.h",
-  ]
-  deps = [
-    ":webrtc_common",
-    "api:transport_api",
-    "common_video:common_video",
-    "rtc_base:rtc_base_approved",
-  ]
-}
-
 if (!build_with_chromium) {
   # Target to build all the WebRTC production code.
   rtc_static_library("webrtc") {
     # Only the root target should depend on this.
     visibility = [ "//:default" ]
 
     sources = []
     complete_static_lib = true
     defines = []
 
@@ skipping 285 matching lines @@
       ]
 
       deps = [
         "examples:AppRTCMobile_javalib",
         "sdk/android:libjingle_peerconnection_java",
         "//base:base_java_test_support",
       ]
     }
   }
 }