mirror of
https://github.com/klzgrad/naiveproxy.git
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221 lines
6.5 KiB
Plaintext
221 lines
6.5 KiB
Plaintext
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../webrtc.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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rtc_static_library("audio") {
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sources = [
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"audio_receive_stream.cc",
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"audio_receive_stream.h",
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"audio_send_stream.cc",
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"audio_send_stream.h",
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"audio_state.cc",
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"audio_state.h",
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"audio_transport_proxy.cc",
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"audio_transport_proxy.h",
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"conversion.h",
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"null_audio_poller.cc",
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"null_audio_poller.h",
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"scoped_voe_interface.h",
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"time_interval.cc",
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"time_interval.h",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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deps = [
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"..:webrtc_common",
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"../api:audio_mixer_api",
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"../api:call_api",
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"../api:optional",
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"../api/audio_codecs:audio_codecs_api",
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"../api/audio_codecs:builtin_audio_encoder_factory",
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"../call:bitrate_allocator",
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"../call:call_interfaces",
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"../call:rtp_interfaces",
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"../common_audio",
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"../modules/audio_coding:cng",
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"../modules/audio_device",
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"../modules/audio_processing",
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"../modules/bitrate_controller:bitrate_controller",
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"../modules/congestion_controller:congestion_controller",
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"../modules/pacing:pacing",
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"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
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"../modules/rtp_rtcp:rtp_rtcp",
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"../rtc_base:rtc_base",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_task_queue",
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"../system_wrappers",
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"../voice_engine",
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]
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}
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if (rtc_include_tests) {
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rtc_source_set("audio_end_to_end_test") {
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testonly = true
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sources = [
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"test/audio_end_to_end_test.cc",
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"test/audio_end_to_end_test.h",
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]
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deps = [
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":audio",
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"../system_wrappers:system_wrappers",
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"../test:fake_audio_device",
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"../test:test_common",
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"../test:test_support",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_source_set("audio_tests") {
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testonly = true
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sources = [
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"audio_receive_stream_unittest.cc",
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"audio_send_stream_tests.cc",
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"audio_send_stream_unittest.cc",
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"audio_state_unittest.cc",
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"time_interval_unittest.cc",
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]
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deps = [
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":audio",
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":audio_end_to_end_test",
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"../api:mock_audio_mixer",
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"../call:mock_rtp_interfaces",
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"../call:rtp_interfaces",
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"../call:rtp_receiver",
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"../modules/audio_device:mock_audio_device",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/audio_processing:audio_processing_statistics",
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"../modules/congestion_controller:congestion_controller",
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"../modules/congestion_controller:mock_congestion_controller",
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"../modules/pacing:mock_paced_sender",
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"../modules/pacing:pacing",
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"../modules/rtp_rtcp:mock_rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_tests_utils",
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"../rtc_base:rtc_task_queue",
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"../system_wrappers:system_wrappers",
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"../test:audio_codec_mocks",
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"../test:rtp_test_utils",
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"../test:test_common",
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"../test:test_support",
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"../voice_engine",
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"utility:utility_tests",
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"//testing/gmock",
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"//testing/gtest",
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]
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if (!rtc_use_memcheck) {
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# This test is timing dependent, which rules out running on memcheck bots.
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sources += [ "test/audio_stats_test.cc" ]
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}
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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if (rtc_enable_protobuf) {
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rtc_test("low_bandwidth_audio_test") {
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testonly = true
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sources = [
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"test/low_bandwidth_audio_test.cc",
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]
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deps = [
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":audio_end_to_end_test",
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"../common_audio",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers",
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"../test:fake_audio_device",
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"../test:test_common",
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"../test:test_main",
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"//testing/gmock",
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"//testing/gtest",
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]
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if (is_android) {
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deps += [ "//testing/android/native_test:native_test_native_code" ]
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}
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data = [
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"../resources/voice_engine/audio_dtx16.wav",
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"../resources/voice_engine/audio_tiny16.wav",
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"../resources/voice_engine/audio_tiny48.wav",
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"test/low_bandwidth_audio_test.py",
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]
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if (is_linux) {
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data += [
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"../tools_webrtc/audio_quality/linux/PolqaOem64",
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"../tools_webrtc/audio_quality/linux/pesq",
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]
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}
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if (is_win) {
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data += [
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"../tools_webrtc/audio_quality/win/PolqaOem64.dll",
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"../tools_webrtc/audio_quality/win/PolqaOem64.exe",
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"../tools_webrtc/audio_quality/win/pesq.exe",
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"../tools_webrtc/audio_quality/win/vcomp120.dll",
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]
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}
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if (is_mac) {
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data += [ "../tools_webrtc/audio_quality/mac/pesq" ]
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}
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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}
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rtc_source_set("audio_perf_tests") {
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testonly = true
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sources = [
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"test/audio_bwe_integration_test.cc",
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"test/audio_bwe_integration_test.h",
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]
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deps = [
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"../common_audio",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers",
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"../test:fake_audio_device",
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"../test:field_trial",
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"../test:single_threaded_task_queue",
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"../test:test_common",
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"../test:test_main",
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"//testing/gmock",
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"//testing/gtest",
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]
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data = [
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"//resources/voice_engine/audio_dtx16.wav",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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}
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