Files
platform-external-webrtc/modules/pacing/BUILD.gn
Karl Wiberg 918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00

121 lines
3.3 KiB
Plaintext

# 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.
import("../../webrtc.gni")
rtc_static_library("pacing") {
sources = [
"bitrate_prober.cc",
"bitrate_prober.h",
"paced_sender.cc",
"paced_sender.h",
"pacer.h",
"packet_queue.cc",
"packet_queue.h",
"packet_queue_interface.cc",
"packet_queue_interface.h",
"packet_router.cc",
"packet_router.h",
"round_robin_packet_queue.cc",
"round_robin_packet_queue.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" ]
}
deps = [
":interval_budget",
"..:module_api",
"../../:typedefs",
"../../:webrtc_common",
"../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../logging:rtc_event_pacing",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base/experiments:alr_experiment",
"../../system_wrappers",
"../../system_wrappers:field_trial_api",
"../../system_wrappers:runtime_enabled_features_api",
"../congestion_controller/goog_cc:alr_detector",
"../remote_bitrate_estimator",
"../rtp_rtcp",
"../rtp_rtcp:rtp_rtcp_format",
"../utility",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
}
rtc_source_set("interval_budget") {
sources = [
"interval_budget.cc",
"interval_budget.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" ]
}
deps = [
# "../../:typedefs",
"../../:webrtc_common",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
]
}
if (rtc_include_tests) {
rtc_source_set("pacing_unittests") {
testonly = true
sources = [
"bitrate_prober_unittest.cc",
"interval_budget_unittest.cc",
"paced_sender_unittest.cc",
"packet_router_unittest.cc",
]
deps = [
":interval_budget",
":pacing",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:rtc_base_tests_utils",
"../../rtc_base/experiments:alr_experiment",
"../../system_wrappers",
"../../system_wrappers:field_trial_api",
"../../system_wrappers:runtime_enabled_features_api",
"../../test:field_trial",
"../../test:test_support",
"../rtp_rtcp",
"../rtp_rtcp:mock_rtp_rtcp",
"../rtp_rtcp:rtp_rtcp_format",
]
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" ]
}
}
rtc_source_set("mock_paced_sender") {
testonly = true
sources = [
"mock/mock_paced_sender.h",
]
deps = [
":pacing",
"../../system_wrappers",
"../../test:test_support",
]
}
}