blob: d29ca8a82ef613d5d660fd2b79e1cc9c9f26bcc7 [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("../build/webrtc.gni")
rtc_source_set("audio") {
sources = [
"audio_receive_stream.cc",
"audio_receive_stream.h",
"audio_send_stream.cc",
"audio_send_stream.h",
"audio_state.cc",
"audio_state.h",
"conversion.h",
"scoped_voe_interface.h",
]
if (is_clang) {
# Suppress warnings from Chrome's Clang plugins.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
"..:webrtc_common",
"../api:call_api",
"../system_wrappers",
"../voice_engine",
]
}
if (rtc_include_tests) {
rtc_source_set("audio_tests") {
testonly = true
sources = [
"audio_receive_stream_unittest.cc",
"audio_send_stream_unittest.cc",
"audio_state_unittest.cc",
]
deps = [
":audio",
"//testing/gmock",
"//testing/gtest",
]
if (is_clang) {
# Suppress warnings from the Chromium Clang plugin.
# See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}