blob: 47129530e170fe3d6c20b83c9588211b9b5f00d2 [file] [log] [blame]
# 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("../webrtc.gni")
rtc_source_set("call_interfaces") {
sources = [
"audio_receive_stream.h",
"audio_send_stream.cc",
"audio_send_stream.h",
"audio_state.h",
"call.h",
"flexfec_receive_stream.h",
"rtp_transport_controller_send.h",
"syncable.cc",
"syncable.h",
]
deps = [
"..:webrtc_common",
"../api:audio_mixer_api",
"../api:transport_api",
"../api/audio_codecs:audio_codecs_api",
"../base:rtc_base",
"../base:rtc_base_approved",
"../modules/audio_coding:audio_encoder_interface",
]
}
rtc_static_library("call") {
sources = [
"bitrate_allocator.cc",
"call.cc",
"flexfec_receive_stream_impl.cc",
"flexfec_receive_stream_impl.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 = [
":call_interfaces",
"../api:call_api",
]
deps = [
":call_interfaces",
"..:webrtc_common",
"../api:transport_api",
"../audio",
"../base:rtc_task_queue",
"../logging:rtc_event_log_api",
"../logging:rtc_event_log_impl",
"../modules/bitrate_controller",
"../modules/congestion_controller",
"../modules/pacing",
"../modules/rtp_rtcp",
"../modules/utility",
"../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",
]
deps = [
":call",
"../base:rtc_base_approved",
"../logging:rtc_event_log_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer",
"../modules/bitrate_controller",
"../modules/pacing",
"../modules/rtp_rtcp",
"../system_wrappers",
"../test:direct_transport",
"../test:test_common",
"../test:test_support",
"../test:video_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" ]
}
}
rtc_source_set("call_perf_tests") {
testonly = true
sources = [
"call_perf_tests.cc",
"rampup_tests.cc",
"rampup_tests.h",
]
deps = [
":call_interfaces",
"..:webrtc_common",
"../base:rtc_base_approved",
"../logging:rtc_event_log_api",
"../modules/audio_coding",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/rtp_rtcp",
"../system_wrappers",
"../system_wrappers:metrics_default",
"../test:direct_transport",
"../test:fake_audio_device",
"../test:test_support",
"../test:video_test_common",
"../video",
"../voice_engine",
"//testing/gtest",
"//webrtc/test:field_trial",
"//webrtc/test:test_common",
]
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" ]
}
}
}