mirror of
https://github.com/klzgrad/naiveproxy.git
synced 2024-12-01 01:36:09 +03:00
505 lines
14 KiB
Plaintext
505 lines
14 KiB
Plaintext
# Copyright (c) 2014 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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config("audio_device_warnings_config") {
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if (is_win && is_clang) {
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cflags = [
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# Disable warnings failing when compiling with Clang on Windows.
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# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
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"-Wno-delete-non-virtual-dtor",
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"-Wno-microsoft-goto",
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]
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}
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}
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rtc_source_set("audio_device") {
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visibility = [ "*" ]
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public_deps = [
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":audio_device_api",
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# Deprecated.
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# TODO(webrtc:7452): Remove this public dep. audio_device_impl should
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# be depended on directly if needed.
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":audio_device_impl",
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]
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}
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if (rtc_include_internal_audio_device && is_ios) {
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rtc_source_set("audio_device_ios_objc") {
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visibility = [
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":audio_device_impl",
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":audio_device_ios_objc_unittests",
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]
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sources = [
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"ios/audio_device_ios.h",
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"ios/audio_device_ios.mm",
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"ios/audio_device_not_implemented_ios.mm",
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"ios/audio_session_observer.h",
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"ios/objc/RTCAudioSession.h",
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"ios/objc/RTCAudioSessionConfiguration.h",
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"ios/objc/RTCAudioSessionDelegateAdapter.h",
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"ios/objc/RTCAudioSessionDelegateAdapter.mm",
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"ios/voice_processing_audio_unit.h",
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"ios/voice_processing_audio_unit.mm",
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]
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libs = [
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"AudioToolbox.framework",
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"AVFoundation.framework",
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"Foundation.framework",
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"UIKit.framework",
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]
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deps = [
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":audio_device_api",
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":audio_device_buffer",
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":audio_device_generic",
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"../../api:array_view",
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"../../rtc_base:checks",
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"../../rtc_base:gtest_prod",
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"../../rtc_base:rtc_base",
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"../../rtc_base/system:fallthrough",
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"../../sdk:audio_objc",
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"../../sdk:common_objc",
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"../../system_wrappers:metrics_api",
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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_device_api") {
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visibility = [ "*" ]
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sources = [
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"include/audio_device.h",
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"include/audio_device_defines.h",
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]
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deps = [
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"../../rtc_base:checks",
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"../../rtc_base:deprecation",
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"../../rtc_base:rtc_base_approved",
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"../../rtc_base:stringutils",
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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_device_buffer") {
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sources = [
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"audio_device_buffer.cc",
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"audio_device_buffer.h",
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"audio_device_config.h",
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"fine_audio_buffer.cc",
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"fine_audio_buffer.h",
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]
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deps = [
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":audio_device_api",
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"../../api:array_view",
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"../../common_audio:common_audio_c",
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"../../rtc_base:checks",
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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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"../../system_wrappers:metrics_api",
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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_device_generic") {
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sources = [
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"audio_device_generic.cc",
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"audio_device_generic.h",
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]
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deps = [
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":audio_device_api",
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":audio_device_buffer",
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"../../rtc_base:rtc_base_approved",
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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_device_name") {
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sources = [
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"audio_device_name.cc",
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"audio_device_name.h",
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]
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}
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rtc_source_set("windows_core_audio_utility") {
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if (is_win && !build_with_chromium) {
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sources = [
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"win/core_audio_utility_win.cc",
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"win/core_audio_utility_win.h",
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]
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deps = [
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":audio_device_api",
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":audio_device_name",
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"../../api/units:time_delta",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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]
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}
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}
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# An ADM with a dedicated factory method which does not depend on the
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# audio_device_impl target. The goal is to use this new structure and
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# gradually phase out the old design.
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# TODO(henrika): currently only supported on Windows.
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rtc_source_set("audio_device_module_from_input_and_output") {
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visibility = [ ":*" ]
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if (is_win && !build_with_chromium) {
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sources = [
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"include/audio_device_factory.cc",
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"include/audio_device_factory.h",
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]
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sources += [
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"win/audio_device_module_win.cc",
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"win/audio_device_module_win.h",
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"win/core_audio_base_win.cc",
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"win/core_audio_base_win.h",
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"win/core_audio_input_win.cc",
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"win/core_audio_input_win.h",
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"win/core_audio_output_win.cc",
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"win/core_audio_output_win.h",
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]
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deps = [
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":audio_device_api",
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":audio_device_buffer",
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":windows_core_audio_utility",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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}
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# Contains default implementations of webrtc::AudioDeviceModule for Windows,
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# Linux, Mac, iOS and Android.
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rtc_source_set("audio_device_impl") {
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visibility = [ "*" ]
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deps = [
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":audio_device_api",
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":audio_device_buffer",
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":audio_device_generic",
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"../..:webrtc_common",
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"../../api:array_view",
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"../../common_audio",
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"../../common_audio:common_audio_c",
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"../../rtc_base:checks",
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"../../rtc_base:deprecation",
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"../../rtc_base:rtc_base_approved",
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"../../rtc_base:rtc_task_queue",
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"../../rtc_base/system:file_wrapper",
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"../../system_wrappers",
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"../../system_wrappers:metrics_api",
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"../utility",
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]
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if (rtc_include_internal_audio_device && is_ios) {
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deps += [ ":audio_device_ios_objc" ]
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}
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sources = [
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"dummy/audio_device_dummy.cc",
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"dummy/audio_device_dummy.h",
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"dummy/file_audio_device.cc",
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"dummy/file_audio_device.h",
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"include/fake_audio_device.h",
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"include/test_audio_device.cc",
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"include/test_audio_device.h",
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]
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if (build_with_mozilla) {
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sources += [
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"opensl/single_rw_fifo.cc",
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"opensl/single_rw_fifo.h",
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]
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}
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defines = []
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cflags = []
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if (rtc_audio_device_plays_sinus_tone) {
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defines += [ "AUDIO_DEVICE_PLAYS_SINUS_TONE" ]
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}
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if (rtc_enable_android_aaudio) {
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defines += [ "AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO" ]
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}
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if (rtc_include_internal_audio_device) {
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# TODO(bugs.webrtc.org/8850): remove this when the circular dependency will be fixed.
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check_includes = false
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sources += [
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"audio_device_data_observer.cc",
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"audio_device_impl.cc",
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"audio_device_impl.h",
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"include/audio_device_data_observer.h",
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]
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if (is_android) {
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sources += [
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"android/audio_common.h",
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"android/audio_device_template.h",
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"android/audio_manager.cc",
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"android/audio_manager.h",
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"android/audio_record_jni.cc",
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"android/audio_record_jni.h",
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"android/audio_track_jni.cc",
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"android/audio_track_jni.h",
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"android/build_info.cc",
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"android/build_info.h",
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"android/opensles_common.cc",
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"android/opensles_common.h",
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"android/opensles_player.cc",
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"android/opensles_player.h",
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"android/opensles_recorder.cc",
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"android/opensles_recorder.h",
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]
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libs = [
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"log",
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"OpenSLES",
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]
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if (rtc_enable_android_aaudio) {
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sources += [
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"android/aaudio_player.cc",
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"android/aaudio_player.h",
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"android/aaudio_recorder.cc",
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"android/aaudio_recorder.h",
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"android/aaudio_wrapper.cc",
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"android/aaudio_wrapper.h",
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]
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libs += [ "aaudio" ]
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}
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if (build_with_mozilla) {
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include_dirs += [
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"/config/external/nspr",
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"/nsprpub/lib/ds",
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"/nsprpub/pr/include",
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]
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}
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}
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if (rtc_use_dummy_audio_file_devices) {
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defines += [ "WEBRTC_DUMMY_FILE_DEVICES" ]
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} else {
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if (is_linux) {
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sources += [
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"linux/alsasymboltable_linux.cc",
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"linux/alsasymboltable_linux.h",
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"linux/audio_device_alsa_linux.cc",
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"linux/audio_device_alsa_linux.h",
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"linux/audio_mixer_manager_alsa_linux.cc",
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"linux/audio_mixer_manager_alsa_linux.h",
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"linux/latebindingsymboltable_linux.cc",
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"linux/latebindingsymboltable_linux.h",
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]
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defines += [ "LINUX_ALSA" ]
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libs = [ "dl" ]
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if (rtc_use_x11) {
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libs += [ "X11" ]
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defines += [ "WEBRTC_USE_X11" ]
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}
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if (rtc_include_pulse_audio) {
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sources += [
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"linux/audio_device_pulse_linux.cc",
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"linux/audio_device_pulse_linux.h",
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"linux/audio_mixer_manager_pulse_linux.cc",
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"linux/audio_mixer_manager_pulse_linux.h",
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"linux/pulseaudiosymboltable_linux.cc",
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"linux/pulseaudiosymboltable_linux.h",
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]
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defines += [ "LINUX_PULSE" ]
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}
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}
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if (is_mac) {
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sources += [
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"mac/audio_device_mac.cc",
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"mac/audio_device_mac.h",
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"mac/audio_mixer_manager_mac.cc",
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"mac/audio_mixer_manager_mac.h",
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]
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deps += [ "../third_party/portaudio:mac_portaudio" ]
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libs = [
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# Needed for CoreGraphics:
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"ApplicationServices.framework",
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"AudioToolbox.framework",
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"CoreAudio.framework",
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# Needed for CGEventSourceKeyState in audio_device_mac.cc:
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"CoreGraphics.framework",
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]
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}
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if (is_win) {
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sources += [
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"win/audio_device_core_win.cc",
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"win/audio_device_core_win.h",
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]
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libs = [
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# Required for the built-in WASAPI AEC.
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"dmoguids.lib",
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"wmcodecdspuuid.lib",
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"amstrmid.lib",
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"msdmo.lib",
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]
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}
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configs += [ ":audio_device_warnings_config" ]
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}
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} else {
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defines = [ "WEBRTC_DUMMY_AUDIO_BUILD" ]
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}
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if (!build_with_chromium) {
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sources += [
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# Do not link these into Chrome since they contain static data.
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"dummy/file_audio_device_factory.cc",
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"dummy/file_audio_device_factory.h",
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]
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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("mock_audio_device") {
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testonly = true
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sources = [
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"include/mock_audio_device.h",
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"include/mock_audio_transport.h",
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"mock_audio_device_buffer.h",
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]
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deps = [
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":audio_device",
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":audio_device_buffer",
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":audio_device_impl",
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"../../test:test_support",
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]
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}
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if (rtc_include_tests) {
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# TODO(kthelgason): Reenable these tests on simulator.
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# See bugs.webrtc.org/7812
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if (is_ios && !use_ios_simulator) {
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rtc_source_set("audio_device_ios_objc_unittests") {
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testonly = true
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visibility = [ ":*" ]
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sources = [
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"ios/audio_device_unittest_ios.mm",
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]
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deps = [
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":audio_device",
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":audio_device_buffer",
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":audio_device_impl",
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":audio_device_ios_objc",
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":mock_audio_device",
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"../../rtc_base:rtc_base_approved",
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"../../sdk:audio_objc",
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"../../system_wrappers",
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"../../test:fileutils",
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"../../test:test_support",
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"//third_party/ocmock",
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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_device_unittests") {
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testonly = true
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sources = [
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"fine_audio_buffer_unittest.cc",
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"include/test_audio_device_unittest.cc",
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]
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deps = [
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":audio_device",
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":audio_device_buffer",
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":audio_device_impl",
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":mock_audio_device",
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"../../api:array_view",
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"../../common_audio",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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"../../system_wrappers",
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"../../test:fileutils",
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"../../test:test_support",
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"../utility:utility",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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if (is_linux || is_mac || is_win) {
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sources += [ "audio_device_unittest.cc" ]
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}
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if (is_win) {
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sources += [ "win/core_audio_utility_win_unittest.cc" ]
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deps += [
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":audio_device_module_from_input_and_output",
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":windows_core_audio_utility",
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]
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}
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if (is_android) {
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# Need to disable error due to the line in
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# base/android/jni_android.h triggering it:
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# const BASE_EXPORT jobject GetApplicationContext()
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# error: type qualifiers ignored on function return type
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cflags = [ "-Wno-ignored-qualifiers" ]
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sources += [
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"android/audio_device_unittest.cc",
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"android/audio_manager_unittest.cc",
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"android/ensure_initialized.cc",
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"android/ensure_initialized.h",
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]
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deps += [
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"../../base",
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"../../sdk/android:libjingle_peerconnection_java",
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]
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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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if (!rtc_include_internal_audio_device) {
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defines = [ "WEBRTC_DUMMY_AUDIO_BUILD" ]
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}
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}
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}
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|
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if (!build_with_chromium && is_android) {
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rtc_android_library("audio_device_java") {
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java_files = [
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"android/java/src/org/webrtc/voiceengine/BuildInfo.java",
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"android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java",
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"android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java",
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"android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java",
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"android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java",
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"android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java",
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]
|
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deps = [
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|
"../../rtc_base:base_java",
|
|
]
|
|
}
|
|
}
|