|  | # 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("audio") { | 
|  | sources = [ | 
|  | "audio_receive_stream.cc", | 
|  | "audio_receive_stream.h", | 
|  | "audio_send_stream.cc", | 
|  | "audio_send_stream.h", | 
|  | "audio_state.cc", | 
|  | "audio_state.h", | 
|  | "audio_transport_proxy.cc", | 
|  | "audio_transport_proxy.h", | 
|  | "conversion.h", | 
|  | "scoped_voe_interface.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 = [ | 
|  | "..:webrtc_common", | 
|  | "../api:audio_mixer_api", | 
|  | "../api:call_api", | 
|  | "../base:rtc_base_approved", | 
|  | "../base:rtc_task_queue", | 
|  | "../call:call_interfaces", | 
|  | "../common_audio", | 
|  | "../modules/audio_device", | 
|  | "../modules/audio_processing", | 
|  | "../modules/congestion_controller:congestion_controller", | 
|  | "../modules/pacing:pacing", | 
|  | "../modules/remote_bitrate_estimator:remote_bitrate_estimator", | 
|  | "../modules/rtp_rtcp:rtp_rtcp", | 
|  | "../system_wrappers", | 
|  | "../voice_engine", | 
|  | ] | 
|  | } | 
|  | if (rtc_include_tests) { | 
|  | rtc_source_set("audio_tests") { | 
|  | testonly = true | 
|  |  | 
|  | # TODO(kjellander): Remove (bugs.webrtc.org/6828) | 
|  | # This needs remote_bitrate_estimator to be moved to webrtc/api first. | 
|  | check_includes = false | 
|  |  | 
|  | sources = [ | 
|  | "audio_receive_stream_unittest.cc", | 
|  | "audio_send_stream_unittest.cc", | 
|  | "audio_state_unittest.cc", | 
|  | ] | 
|  | deps = [ | 
|  | ":audio", | 
|  | "../api:mock_audio_mixer", | 
|  | "../base:rtc_base_approved", | 
|  | "../base:rtc_task_queue", | 
|  | "../modules/audio_device:mock_audio_device", | 
|  | "../modules/audio_mixer:audio_mixer_impl", | 
|  | "../modules/congestion_controller:congestion_controller", | 
|  | "../modules/pacing:pacing", | 
|  | "../test:test_common", | 
|  | "../test:test_support", | 
|  | "utility:utility_tests", | 
|  | "//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" ] | 
|  | } | 
|  | } | 
|  | } |