blob: 2a29ceb22bcfc364381b4ae39793a3c728781fc9 [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")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.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",
"null_audio_poller.cc",
"null_audio_poller.h",
"scoped_voe_interface.h",
"time_interval.cc",
"time_interval.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",
"../api:optional",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../call:bitrate_allocator",
"../call:call_interfaces",
"../call:rtp_interfaces",
"../common_audio",
"../modules/audio_coding:cng",
"../modules/audio_device",
"../modules/audio_processing",
"../modules/bitrate_controller:bitrate_controller",
"../modules/congestion_controller:congestion_controller",
"../modules/pacing:pacing",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../system_wrappers",
"../voice_engine",
]
}
if (rtc_include_tests) {
rtc_source_set("audio_end_to_end_test") {
testonly = true
sources = [
"test/audio_end_to_end_test.cc",
"test/audio_end_to_end_test.h",
]
deps = [
":audio",
"../system_wrappers:system_wrappers",
"../test:fake_audio_device",
"../test:test_common",
"../test:test_support",
]
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("audio_tests") {
testonly = true
sources = [
"audio_receive_stream_unittest.cc",
"audio_send_stream_tests.cc",
"audio_send_stream_unittest.cc",
"audio_state_unittest.cc",
"time_interval_unittest.cc",
]
deps = [
":audio",
":audio_end_to_end_test",
"../api:mock_audio_mixer",
"../call:mock_rtp_interfaces",
"../call:rtp_interfaces",
"../call:rtp_receiver",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:audio_processing_statistics",
"../modules/congestion_controller:congestion_controller",
"../modules/congestion_controller:mock_congestion_controller",
"../modules/pacing:mock_paced_sender",
"../modules/pacing:pacing",
"../modules/rtp_rtcp:mock_rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_task_queue",
"../system_wrappers:system_wrappers",
"../test:audio_codec_mocks",
"../test:rtp_test_utils",
"../test:test_common",
"../test:test_support",
"../voice_engine",
"utility:utility_tests",
"//testing/gmock",
"//testing/gtest",
]
if (!rtc_use_memcheck) {
# This test is timing dependent, which rules out running on memcheck bots.
sources += [ "test/audio_stats_test.cc" ]
}
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" ]
}
}
if (rtc_enable_protobuf) {
rtc_test("low_bandwidth_audio_test") {
testonly = true
sources = [
"test/low_bandwidth_audio_test.cc",
]
deps = [
":audio_end_to_end_test",
"../common_audio",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../test:fake_audio_device",
"../test:test_common",
"../test:test_main",
"//testing/gmock",
"//testing/gtest",
]
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
}
data = [
"../resources/voice_engine/audio_dtx16.wav",
"../resources/voice_engine/audio_tiny16.wav",
"../resources/voice_engine/audio_tiny48.wav",
"test/low_bandwidth_audio_test.py",
]
if (is_linux) {
data += [
"../tools_webrtc/audio_quality/linux/PolqaOem64",
"../tools_webrtc/audio_quality/linux/pesq",
]
}
if (is_win) {
data += [
"../tools_webrtc/audio_quality/win/PolqaOem64.dll",
"../tools_webrtc/audio_quality/win/PolqaOem64.exe",
"../tools_webrtc/audio_quality/win/pesq.exe",
"../tools_webrtc/audio_quality/win/vcomp120.dll",
]
}
if (is_mac) {
data += [ "../tools_webrtc/audio_quality/mac/pesq" ]
}
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("audio_perf_tests") {
testonly = true
sources = [
"test/audio_bwe_integration_test.cc",
"test/audio_bwe_integration_test.h",
]
deps = [
"../common_audio",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../test:fake_audio_device",
"../test:field_trial",
"../test:single_threaded_task_queue",
"../test:test_common",
"../test:test_main",
"//testing/gmock",
"//testing/gtest",
]
data = [
"//resources/voice_engine/audio_dtx16.wav",
]
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" ]
}
}
}