Fredrik Hernqvist efbe753617 Add RTCAudioPlayoutStats to GetStats().
This is done by allowing implementations of AudioDeviceModule to
implement the GetStats() method. The default implementation returns
nullopt, in which case RTCAudioPlayoutStats will not be visible in the
stats.

Bug: webrtc:14653
Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39115}
2023-01-16 13:19:45 +00:00

496 lines
14 KiB
Plaintext

# Copyright (c) 2014 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")
}
config("audio_device_warnings_config") {
if (is_win && is_clang) {
cflags = [
# Disable warnings failing when compiling with Clang on Windows.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
"-Wno-microsoft-goto",
]
}
}
rtc_source_set("audio_device_default") {
visibility = [ "*" ]
sources = [ "include/audio_device_default.h" ]
deps = [ ":audio_device_api" ]
}
rtc_source_set("audio_device") {
visibility = [ "*" ]
public_deps = [
":audio_device_api",
# Deprecated.
# TODO(webrtc:7452): Remove this public dep. audio_device_impl should
# be depended on directly if needed.
":audio_device_impl",
]
}
rtc_source_set("audio_device_api") {
visibility = [ "*" ]
sources = [
"include/audio_device.h",
"include/audio_device_defines.h",
]
deps = [
"../../api:scoped_refptr",
"../../api/task_queue",
"../../rtc_base:checks",
"../../rtc_base:refcount",
"../../rtc_base:stringutils",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("audio_device_buffer") {
sources = [
"audio_device_buffer.cc",
"audio_device_buffer.h",
"audio_device_config.h",
"fine_audio_buffer.cc",
"fine_audio_buffer.h",
]
deps = [
":audio_device_api",
"../../api:array_view",
"../../api:sequence_checker",
"../../api/task_queue",
"../../common_audio:common_audio_c",
"../../rtc_base:buffer",
"../../rtc_base:checks",
"../../rtc_base:event_tracer",
"../../rtc_base:logging",
"../../rtc_base:macromagic",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:safe_conversions",
"../../rtc_base:timestamp_aligner",
"../../rtc_base:timeutils",
"../../rtc_base/synchronization:mutex",
"../../system_wrappers",
"../../system_wrappers:metrics",
]
}
rtc_library("audio_device_generic") {
sources = [
"audio_device_generic.cc",
"audio_device_generic.h",
]
deps = [
":audio_device_api",
":audio_device_buffer",
"../../rtc_base:logging",
]
}
rtc_library("audio_device_name") {
sources = [
"audio_device_name.cc",
"audio_device_name.h",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
rtc_source_set("windows_core_audio_utility") {
if (is_win && !build_with_chromium) {
sources = [
"win/core_audio_utility_win.cc",
"win/core_audio_utility_win.h",
]
deps = [
":audio_device_api",
":audio_device_name",
"../../api/units:time_delta",
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../rtc_base:macromagic",
"../../rtc_base:platform_thread_types",
"../../rtc_base:stringutils",
"../../rtc_base/win:windows_version",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
libs = [ "oleaut32.lib" ]
}
}
# An ADM with a dedicated factory method which does not depend on the
# audio_device_impl target. The goal is to use this new structure and
# gradually phase out the old design.
# TODO(henrika): currently only supported on Windows.
rtc_source_set("audio_device_module_from_input_and_output") {
visibility = [ "*" ]
if (is_win && !build_with_chromium) {
sources = [
"include/audio_device_factory.cc",
"include/audio_device_factory.h",
]
sources += [
"win/audio_device_module_win.cc",
"win/audio_device_module_win.h",
"win/core_audio_base_win.cc",
"win/core_audio_base_win.h",
"win/core_audio_input_win.cc",
"win/core_audio_input_win.h",
"win/core_audio_output_win.cc",
"win/core_audio_output_win.h",
]
deps = [
":audio_device_api",
":audio_device_buffer",
":windows_core_audio_utility",
"../../api:make_ref_counted",
"../../api:scoped_refptr",
"../../api:sequence_checker",
"../../api/task_queue",
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../rtc_base:macromagic",
"../../rtc_base:platform_thread",
"../../rtc_base:safe_conversions",
"../../rtc_base:stringutils",
"../../rtc_base:timeutils",
"../../rtc_base/win:scoped_com_initializer",
"../../rtc_base/win:windows_version",
]
absl_deps = [
"//third_party/abseil-cpp/absl/strings:strings",
"//third_party/abseil-cpp/absl/types:optional",
]
}
}
# Contains default implementations of webrtc::AudioDeviceModule for Windows,
# Linux, Mac, iOS and Android.
rtc_library("audio_device_impl") {
visibility = [ "*" ]
deps = [
":audio_device_api",
":audio_device_buffer",
":audio_device_default",
":audio_device_generic",
"../../api:array_view",
"../../api:make_ref_counted",
"../../api:refcountedbase",
"../../api:scoped_refptr",
"../../api:sequence_checker",
"../../api/task_queue",
"../../common_audio",
"../../common_audio:common_audio_c",
"../../rtc_base:buffer",
"../../rtc_base:checks",
"../../rtc_base:logging",
"../../rtc_base:macromagic",
"../../rtc_base:platform_thread",
"../../rtc_base:random",
"../../rtc_base:rtc_event",
"../../rtc_base:rtc_task_queue",
"../../rtc_base:safe_conversions",
"../../rtc_base:stringutils",
"../../rtc_base:timeutils",
"../../rtc_base/synchronization:mutex",
"../../rtc_base/system:arch",
"../../rtc_base/system:file_wrapper",
"../../rtc_base/task_utils:repeating_task",
"../../system_wrappers",
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
"../utility",
]
absl_deps = [
"//third_party/abseil-cpp/absl/base:core_headers",
"//third_party/abseil-cpp/absl/strings:strings",
]
if (rtc_include_internal_audio_device && is_ios) {
deps += [ "../../sdk:audio_device" ]
}
sources = [
"dummy/audio_device_dummy.cc",
"dummy/audio_device_dummy.h",
"dummy/file_audio_device.cc",
"dummy/file_audio_device.h",
"include/fake_audio_device.h",
"include/test_audio_device.cc",
"include/test_audio_device.h",
]
if (build_with_mozilla) {
sources += [
"opensl/single_rw_fifo.cc",
"opensl/single_rw_fifo.h",
]
}
defines = []
cflags = []
if (rtc_audio_device_plays_sinus_tone) {
defines += [ "AUDIO_DEVICE_PLAYS_SINUS_TONE" ]
}
if (rtc_enable_android_aaudio) {
defines += [ "WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO" ]
}
if (rtc_include_internal_audio_device) {
sources += [
"audio_device_data_observer.cc",
"audio_device_impl.cc",
"audio_device_impl.h",
"include/audio_device_data_observer.h",
]
if (is_android) {
sources += [
"android/audio_common.h",
"android/audio_device_template.h",
"android/audio_manager.cc",
"android/audio_manager.h",
"android/audio_record_jni.cc",
"android/audio_record_jni.h",
"android/audio_track_jni.cc",
"android/audio_track_jni.h",
"android/build_info.cc",
"android/build_info.h",
"android/opensles_common.cc",
"android/opensles_common.h",
"android/opensles_player.cc",
"android/opensles_player.h",
"android/opensles_recorder.cc",
"android/opensles_recorder.h",
]
libs = [
"log",
"OpenSLES",
]
if (rtc_enable_android_aaudio) {
sources += [
"android/aaudio_player.cc",
"android/aaudio_player.h",
"android/aaudio_recorder.cc",
"android/aaudio_recorder.h",
"android/aaudio_wrapper.cc",
"android/aaudio_wrapper.h",
]
libs += [ "aaudio" ]
}
if (build_with_mozilla) {
include_dirs += [
"/config/external/nspr",
"/nsprpub/lib/ds",
"/nsprpub/pr/include",
]
}
}
if (rtc_use_dummy_audio_file_devices) {
defines += [ "WEBRTC_DUMMY_FILE_DEVICES" ]
} else {
if (is_linux || is_chromeos) {
sources += [
"linux/alsasymboltable_linux.cc",
"linux/alsasymboltable_linux.h",
"linux/audio_device_alsa_linux.cc",
"linux/audio_device_alsa_linux.h",
"linux/audio_mixer_manager_alsa_linux.cc",
"linux/audio_mixer_manager_alsa_linux.h",
"linux/latebindingsymboltable_linux.cc",
"linux/latebindingsymboltable_linux.h",
]
defines += [ "WEBRTC_ENABLE_LINUX_ALSA" ]
libs = [ "dl" ]
if (rtc_use_x11) {
libs += [ "X11" ]
defines += [ "WEBRTC_USE_X11" ]
}
if (rtc_include_pulse_audio) {
defines += [ "WEBRTC_ENABLE_LINUX_PULSE" ]
}
sources += [
"linux/audio_device_pulse_linux.cc",
"linux/audio_device_pulse_linux.h",
"linux/audio_mixer_manager_pulse_linux.cc",
"linux/audio_mixer_manager_pulse_linux.h",
"linux/pulseaudiosymboltable_linux.cc",
"linux/pulseaudiosymboltable_linux.h",
]
}
if (is_mac) {
sources += [
"mac/audio_device_mac.cc",
"mac/audio_device_mac.h",
"mac/audio_mixer_manager_mac.cc",
"mac/audio_mixer_manager_mac.h",
]
deps += [
":audio_device_impl_frameworks",
"../third_party/portaudio:mac_portaudio",
]
}
if (is_win) {
sources += [
"win/audio_device_core_win.cc",
"win/audio_device_core_win.h",
]
libs = [
# Required for the built-in WASAPI AEC.
"dmoguids.lib",
"wmcodecdspuuid.lib",
"amstrmid.lib",
"msdmo.lib",
"oleaut32.lib",
]
deps += [
"../../rtc_base:win32",
"../../rtc_base/win:scoped_com_initializer",
]
}
configs += [ ":audio_device_warnings_config" ]
}
} else {
defines = [ "WEBRTC_DUMMY_AUDIO_BUILD" ]
}
if (!build_with_chromium) {
sources += [
# Do not link these into Chrome since they contain static data.
"dummy/file_audio_device_factory.cc",
"dummy/file_audio_device_factory.h",
]
}
}
if (is_mac) {
rtc_source_set("audio_device_impl_frameworks") {
visibility = [ ":*" ]
frameworks = [
# Needed for CoreGraphics:
"ApplicationServices.framework",
"AudioToolbox.framework",
"CoreAudio.framework",
# Needed for CGEventSourceKeyState in audio_device_mac.cc:
"CoreGraphics.framework",
]
}
}
rtc_source_set("mock_audio_device") {
visibility = [ "*" ]
testonly = true
sources = [
"include/mock_audio_device.h",
"include/mock_audio_transport.h",
"mock_audio_device_buffer.h",
]
deps = [
":audio_device",
":audio_device_buffer",
":audio_device_impl",
"../../api:make_ref_counted",
"../../test:test_support",
]
}
if (rtc_include_tests && !build_with_chromium) {
rtc_library("audio_device_unittests") {
testonly = true
sources = [
"fine_audio_buffer_unittest.cc",
"include/test_audio_device_unittest.cc",
]
deps = [
":audio_device",
":audio_device_buffer",
":audio_device_impl",
":mock_audio_device",
"../../api:array_view",
"../../api:scoped_refptr",
"../../api:sequence_checker",
"../../api/task_queue",
"../../api/task_queue:default_task_queue_factory",
"../../common_audio",
"../../rtc_base:buffer",
"../../rtc_base:checks",
"../../rtc_base:ignore_wundef",
"../../rtc_base:logging",
"../../rtc_base:macromagic",
"../../rtc_base:race_checker",
"../../rtc_base:rtc_event",
"../../rtc_base:safe_conversions",
"../../rtc_base:timeutils",
"../../rtc_base/synchronization:mutex",
"../../system_wrappers",
"../../test:fileutils",
"../../test:test_support",
]
absl_deps = [
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
if (is_linux || is_chromeos || is_mac || is_win) {
sources += [ "audio_device_unittest.cc" ]
}
if (is_win) {
sources += [ "win/core_audio_utility_win_unittest.cc" ]
deps += [
":audio_device_module_from_input_and_output",
":windows_core_audio_utility",
"../../rtc_base/win:scoped_com_initializer",
"../../rtc_base/win:windows_version",
]
}
if (is_android) {
sources += [
"android/audio_device_unittest.cc",
"android/audio_manager_unittest.cc",
"android/ensure_initialized.cc",
"android/ensure_initialized.h",
]
deps += [
"../../sdk/android:internal_jni",
"../../sdk/android:libjingle_peerconnection_java",
"../../sdk/android:native_api_jni",
"../../sdk/android:native_test_jni_onload",
"../utility",
]
}
if (!rtc_include_internal_audio_device) {
defines = [ "WEBRTC_DUMMY_AUDIO_BUILD" ]
}
}
}
if (!build_with_chromium && is_android) {
rtc_android_library("audio_device_java") {
sources = [
"android/java/src/org/webrtc/voiceengine/BuildInfo.java",
"android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java",
"android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java",
"android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java",
"android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java",
"android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java",
]
deps = [
"../../rtc_base:base_java",
"//third_party/androidx:androidx_annotation_annotation_java",
]
}
}