Files
platform-external-webrtc/webrtc/call/BUILD.gn
aleloi a8eb756a34 Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.

Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.

transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.

NOTRY=True

BUG=webrtc:5589, webrtc:5878, webrtc:6785

Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
2016-11-28 15:02:19 +00:00

66 lines
1.8 KiB
Plaintext

# 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")
rtc_static_library("call") {
sources = [
"bitrate_allocator.cc",
"call.cc",
"flexfec_receive_stream.cc",
"flexfec_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 = [
"../api:call_api",
]
deps = [
"..:webrtc_common",
"../api:transport_api",
"../audio",
"../base:rtc_task_queue",
"../logging:rtc_event_log_impl",
"../modules/congestion_controller",
"../modules/rtp_rtcp",
"../system_wrappers",
"../video",
]
}
if (rtc_include_tests) {
rtc_source_set("call_tests") {
testonly = true
sources = [
"bitrate_allocator_unittest.cc",
"bitrate_estimator_tests.cc",
"call_unittest.cc",
"flexfec_receive_stream_unittest.cc",
"packet_injection_tests.cc",
]
deps = [
":call",
"../base:rtc_base_approved",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer",
"../test:test_common",
"//testing/gmock",
"//testing/gtest",
]
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" ]
}
}
}