AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M120 release

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Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions
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231
webrtc/BUILD.gn
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@ -12,6 +12,15 @@
# you add a new build file, there must be some path of dependencies from this
# file to your new one or GN won't know about it.
# Use of visibility = clauses:
# The default visibility for all rtc_ targets is equivalent to "//*", or
# "all targets in webrtc can depend on this, nothing outside can".
#
# When overriding, the choices are:
# - visibility = [ "*" ] - public. Stuff outside webrtc can use this.
# - visibility = [ ":*" ] - directory private.
# As a general guideline, only targets in api/ should have public visibility.
import("//build/config/linux/pkg_config.gni")
import("//build/config/sanitizers/sanitizers.gni")
import("webrtc.gni")
@ -21,6 +30,7 @@ if (rtc_enable_protobuf) {
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
import("//third_party/jni_zero/jni_zero.gni")
}
if (!build_with_chromium) {
@ -38,7 +48,6 @@ if (!build_with_chromium) {
if (rtc_include_tests) {
deps += [
":rtc_unittests",
":slow_tests",
":video_engine_tests",
":voip_unittests",
":webrtc_nonparallel_tests",
@ -54,9 +63,14 @@ if (!build_with_chromium) {
"modules/remote_bitrate_estimator:rtp_to_text",
"modules/rtp_rtcp:test_packet_masks_metrics",
"modules/video_capture:video_capture_internal_impl",
"modules/video_coding:video_codec_perf_tests",
"net/dcsctp:dcsctp_unittests",
"pc:peerconnection_unittests",
"pc:rtc_pc_unittests",
"pc:slow_peer_connection_unittests",
"pc:svc_tests",
"rtc_tools:rtp_generator",
"rtc_tools:video_encoder",
"rtc_tools:video_replay",
"stats:rtc_stats_unittests",
"system_wrappers:system_wrappers_unittests",
@ -71,6 +85,13 @@ if (!build_with_chromium) {
# see bugs.webrtc.org/11027#c5.
deps += [ ":webrtc_lib_link_test" ]
}
if (is_ios) {
deps += [
"examples:apprtcmobile_tests",
"sdk:sdk_framework_unittests",
"sdk:sdk_unittests",
]
}
if (is_android) {
deps += [
"examples:android_examples_junit_tests",
@ -82,11 +103,16 @@ if (!build_with_chromium) {
}
if (rtc_enable_protobuf) {
deps += [
"audio:low_bandwidth_audio_test",
"logging:rtc_event_log_rtp_dump",
"tools_webrtc/perf:webrtc_dashboard_upload",
]
}
if ((is_linux || is_chromeos) && rtc_use_pipewire) {
deps += [ "modules/desktop_capture:shared_screencast_stream_test" ]
}
}
if (target_os == "android") {
deps += [ "tools_webrtc:binary_version_check" ]
}
}
}
@ -113,12 +139,23 @@ config("common_inherited_config") {
cflags = []
ldflags = []
if (rtc_enable_symbol_export || is_component_build) {
defines = [ "WEBRTC_ENABLE_SYMBOL_EXPORT" ]
if (rtc_jni_generator_legacy_symbols) {
defines += [ "RTC_JNI_GENERATOR_LEGACY_SYMBOLS" ]
}
if (build_with_mozilla) {
defines += [ "WEBRTC_MOZILLA_BUILD" ]
if (rtc_objc_prefix != "") {
defines += [ "RTC_OBJC_TYPE_PREFIX=${rtc_objc_prefix}" ]
}
if (rtc_dlog_always_on) {
defines += [ "DLOG_ALWAYS_ON" ]
}
if (rtc_enable_symbol_export || is_component_build) {
defines += [ "WEBRTC_ENABLE_SYMBOL_EXPORT" ]
}
if (rtc_enable_objc_symbol_export) {
defines += [ "WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT" ]
}
if (!rtc_builtin_ssl_root_certificates) {
@ -133,6 +170,10 @@ config("common_inherited_config") {
defines += [ "WEBRTC_ENABLE_AVX2" ]
}
if (rtc_enable_win_wgc) {
defines += [ "RTC_ENABLE_WIN_WGC" ]
}
# Some tests need to declare their own trace event handlers. If this define is
# not set, the first time TRACE_EVENT_* is called it will store the return
# value for the current handler in an static variable, so that subsequent
@ -210,14 +251,6 @@ config("common_inherited_config") {
}
}
# TODO(bugs.webrtc.org/9693): Remove the possibility to suppress this warning
# as soon as WebRTC compiles without it.
config("no_exit_time_destructors") {
if (is_clang) {
cflags = [ "-Wno-exit-time-destructors" ]
}
}
# TODO(bugs.webrtc.org/9693): Remove the possibility to suppress this warning
# as soon as WebRTC compiles without it.
config("no_global_constructors") {
@ -249,6 +282,18 @@ config("common_config") {
defines += [ "WEBRTC_ENABLE_PROTOBUF=0" ]
}
if (rtc_strict_field_trials == "") {
defines += [ "WEBRTC_STRICT_FIELD_TRIALS=0" ]
} else if (rtc_strict_field_trials == "dcheck") {
defines += [ "WEBRTC_STRICT_FIELD_TRIALS=1" ]
} else if (rtc_strict_field_trials == "warn") {
defines += [ "WEBRTC_STRICT_FIELD_TRIALS=2" ]
} else {
assert(false,
"Unsupported value for rtc_strict_field_trials: " +
"$rtc_strict_field_trials")
}
if (rtc_include_internal_audio_device) {
defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
}
@ -257,8 +302,16 @@ config("common_config") {
defines += [ "RTC_ENABLE_VP9" ]
}
if (rtc_use_h265) {
defines += [ "RTC_ENABLE_H265" ]
}
if (rtc_include_dav1d_in_internal_decoder_factory) {
defines += [ "RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY" ]
}
if (rtc_enable_sctp) {
defines += [ "HAVE_SCTP" ]
defines += [ "WEBRTC_HAVE_SCTP" ]
}
if (rtc_enable_external_auth) {
@ -273,6 +326,10 @@ config("common_config") {
defines += [ "WEBRTC_ABSL_MUTEX" ]
}
if (rtc_enable_libevent) {
defines += [ "WEBRTC_ENABLE_LIBEVENT" ]
}
if (rtc_disable_logging) {
defines += [ "RTC_DISABLE_LOGGING" ]
}
@ -293,7 +350,16 @@ config("common_config") {
defines += [ "WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE" ]
}
cflags = []
if (is_clang) {
cflags += [
# TODO(webrtc:13219): Fix -Wshadow instances and enable.
"-Wno-shadow",
# See https://reviews.llvm.org/D56731 for details about this
# warning.
"-Wctad-maybe-unsupported",
]
}
if (build_with_chromium) {
defines += [
@ -329,20 +395,17 @@ config("common_config") {
}
if (is_clang) {
cflags += [
"-Wc++11-narrowing",
"-Wimplicit-fallthrough",
"-Wthread-safety",
"-Winconsistent-missing-override",
"-Wundef",
]
cflags += [ "-Wc++11-narrowing" ]
# use_xcode_clang only refers to the iOS toolchain, host binaries use
# chromium's clang always.
if (!is_nacl &&
(!use_xcode_clang || current_toolchain == host_toolchain)) {
# Flags NaCl (Clang 3.7) and Xcode 7.3 (Clang clang-703.0.31) do not
# recognize.
if (!is_fuchsia) {
# Compiling with the Fuchsia SDK results in Wundef errors
# TODO(bugs.fuchsia.dev/100722): Remove from (!is_fuchsia) branch when
# Fuchsia build errors are fixed.
cflags += [ "-Wundef" ]
}
if (!is_nacl) {
# Flags NaCl (Clang 3.7) do not recognize.
cflags += [ "-Wunused-lambda-capture" ]
}
}
@ -404,7 +467,7 @@ config("common_config") {
]
}
if (use_fuzzing_engine && optimize_for_fuzzing) {
if (use_fuzzing_engine) {
# Used in Chromium's overrides to disable logging
defines += [ "WEBRTC_UNSAFE_FUZZER_MODE" ]
}
@ -419,10 +482,6 @@ config("common_config") {
config("common_objc") {
frameworks = [ "Foundation.framework" ]
if (rtc_use_metal_rendering) {
defines = [ "RTC_SUPPORTS_METAL" ]
}
}
if (!build_with_chromium) {
@ -448,6 +507,17 @@ if (!build_with_chromium) {
"api/rtc_event_log:rtc_event_log_factory",
"api/task_queue",
"api/task_queue:default_task_queue_factory",
"api/test/metrics",
"api/video_codecs:video_decoder_factory_template",
"api/video_codecs:video_decoder_factory_template_dav1d_adapter",
"api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
"api/video_codecs:video_decoder_factory_template_libvpx_vp9_adapter",
"api/video_codecs:video_decoder_factory_template_open_h264_adapter",
"api/video_codecs:video_encoder_factory_template",
"api/video_codecs:video_encoder_factory_template_libaom_av1_adapter",
"api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
"api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
"api/video_codecs:video_encoder_factory_template_open_h264_adapter",
"audio",
"call",
"common_audio",
@ -458,10 +528,7 @@ if (!build_with_chromium) {
"modules/video_capture:video_capture_internal_impl",
"p2p:rtc_p2p",
"pc:libjingle_peerconnection",
"pc:peerconnection",
"pc:rtc_pc",
"pc:rtc_pc_base",
"rtc_base",
"sdk",
"video",
]
@ -473,13 +540,6 @@ if (!build_with_chromium) {
]
}
if (rtc_include_builtin_video_codecs) {
deps += [
"api/video_codecs:builtin_video_decoder_factory",
"api/video_codecs:builtin_video_encoder_factory",
]
}
if (build_with_mozilla) {
deps += [
"api/video:video_frame",
@ -504,6 +564,10 @@ if (!build_with_chromium) {
rtc_executable("webrtc_lib_link_test") {
testonly = true
# This target is used for checking to link, so do not check dependencies
# on gn check.
check_includes = false # no-presubmit-check TODO(bugs.webrtc.org/12785)
sources = [ "webrtc_lib_link_test.cc" ]
deps = [
# NOTE: Don't add deps here. If this test fails to link, it means you
@ -523,7 +587,7 @@ if (use_libfuzzer || use_afl) {
}
}
if (rtc_include_tests) {
if (rtc_include_tests && !build_with_chromium) {
rtc_test("rtc_unittests") {
testonly = true
@ -533,13 +597,17 @@ if (rtc_include_tests) {
"api/audio/test:audio_api_unittests",
"api/audio_codecs/test:audio_codecs_api_unittests",
"api/numerics:numerics_unittests",
"api/task_queue:pending_task_safety_flag_unittests",
"api/test/metrics:metrics_unittests",
"api/transport:stun_unittest",
"api/video/test:rtc_api_video_unittests",
"api/video_codecs/test:video_codecs_api_unittests",
"api/voip:compile_all_headers",
"call:fake_network_pipe_unittests",
"p2p:libstunprober_unittests",
"p2p:rtc_p2p_unittests",
"rtc_base:robo_caller_unittests",
"rtc_base:async_dns_resolver_unittests",
"rtc_base:callback_list_unittests",
"rtc_base:rtc_base_approved_unittests",
"rtc_base:rtc_base_unittests",
"rtc_base:rtc_json_unittests",
@ -547,12 +615,13 @@ if (rtc_include_tests) {
"rtc_base:rtc_operations_chain_unittests",
"rtc_base:rtc_task_queue_unittests",
"rtc_base:sigslot_unittest",
"rtc_base:task_queue_stdlib_unittest",
"rtc_base:untyped_function_unittest",
"rtc_base:weak_ptr_unittests",
"rtc_base/experiments:experiments_unittests",
"rtc_base/synchronization:sequence_checker_unittests",
"rtc_base/task_utils:pending_task_safety_flag_unittests",
"rtc_base/task_utils:to_queued_task_unittests",
"rtc_base/system:file_wrapper_unittests",
"rtc_base/task_utils:repeating_task_unittests",
"rtc_base/units:units_unittests",
"sdk:sdk_tests",
"test:rtp_test_utils",
"test:test_main",
@ -574,31 +643,18 @@ if (rtc_include_tests) {
]
shard_timeout = 900
}
}
if (is_ios || is_mac) {
deps += [ "sdk:rtc_unittests_objc" ]
if (rtc_enable_google_benchmarks) {
rtc_test("benchmarks") {
testonly = true
deps = [
"rtc_base/synchronization:mutex_benchmark",
"test:benchmark_main",
]
}
}
rtc_test("benchmarks") {
testonly = true
deps = [
"rtc_base/synchronization:mutex_benchmark",
"test:benchmark_main",
]
}
# This runs tests that must run in real time and therefore can take some
# time to execute. They are in a separate executable to avoid making the
# regular unittest suite too slow to run frequently.
rtc_test("slow_tests") {
testonly = true
deps = [
"rtc_base/task_utils:repeating_task_unittests",
"test:test_main",
]
}
# TODO(pbos): Rename test suite, this is no longer "just" for video targets.
video_engine_tests_resources = [
"resources/foreman_cif_short.yuv",
@ -630,7 +686,12 @@ if (rtc_include_tests) {
]
data = video_engine_tests_resources
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
use_default_launcher = false
deps += [
"//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
"//testing/android/native_test:native_test_java",
"//testing/android/native_test:native_test_support",
]
shard_timeout = 900
}
if (is_ios) {
@ -663,7 +724,6 @@ if (rtc_include_tests) {
rtc_test("webrtc_perf_tests") {
testonly = true
deps = [
"audio:audio_perf_tests",
"call:call_perf_tests",
"modules/audio_coding:audio_coding_perf_tests",
"modules/audio_processing:audio_processing_perf_tests",
@ -675,7 +735,12 @@ if (rtc_include_tests) {
data = webrtc_perf_tests_resources
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
use_default_launcher = false
deps += [
"//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
"//testing/android/native_test:native_test_java",
"//testing/android/native_test:native_test_support",
]
shard_timeout = 4500
}
if (is_ios) {
@ -695,6 +760,7 @@ if (rtc_include_tests) {
rtc_test("voip_unittests") {
testonly = true
deps = [
"api/voip:compile_all_headers",
"api/voip:voip_engine_factory_unittests",
"audio/voip/test:audio_channel_unittests",
"audio/voip/test:audio_egress_unittests",
@ -705,6 +771,23 @@ if (rtc_include_tests) {
}
}
# Build target for standalone dcsctp
rtc_static_library("dcsctp") {
# Only the root target should depend on this.
visibility = [ "//:default" ]
sources = []
complete_static_lib = true
suppressed_configs += [ "//build/config/compiler:thin_archive" ]
defines = []
deps = [
"net/dcsctp/public:factory",
"net/dcsctp/public:socket",
"net/dcsctp/public:types",
"net/dcsctp/socket:dcsctp_socket",
"net/dcsctp/timer:task_queue_timeout",
]
}
# ---- Poisons ----
#
# Here is one empty dummy target for each poison type (needed because
@ -720,7 +803,7 @@ group("poison_audio_codecs") {
group("poison_default_task_queue") {
}
group("poison_rtc_json") {
group("poison_default_echo_detector") {
}
group("poison_software_video_codecs") {