# 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_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", ] } 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_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" ] } } }