Merge branch '7339-testing'

Author: jim-signal

.gn

@@ -83,8 +83,7 @@ default_args = {
   # let's temporarily disable it.
   enable_jni_multiplexing = false
 
-  # TODO(b/42223878): use_fuzztest_wrapper adds a dependency to //base so
-  # let's temporarly disable it.
+  # use_fuzztest_wrapper adds a dependency to //base so we have to disable it.
   use_fuzztest_wrapper = false
 
   # Enable Rust in WebRTC
@@ -94,4 +93,7 @@ default_args = {
 
   # See context on https://issues.chromium.org/issues/415076048.
   enable_js_protobuf = false
+
+  # Use Siso instead of Ninja.
+  use_siso = true
 }

BUILD.gn

@@ -70,7 +70,6 @@ if (!build_with_chromium) {
         "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",
@@ -83,7 +82,9 @@ if (!build_with_chromium) {
       if (use_libfuzzer) {
         deps += [ "test/fuzzers" ]
       }
-      if (!is_asan) {
+
+      # TODO(bugs.webrtc.org/430260876): Remove once rust links with libwebrtc.
+      if (!is_asan && !rtc_rusty_base64) {
         # Do not build :webrtc_lib_link_test because lld complains on some OS
         # (e.g. when target_os = "mac") when is_asan=true. For more details,
         # see bugs.webrtc.org/11027#c5.
@@ -251,10 +252,6 @@ config("common_inherited_config") {
   if (is_ubsan) {
     cflags += [ "-fsanitize=float-cast-overflow" ]
   }
-
-  if (rtc_allow_deprecated_namespaces) {
-    defines += [ "WEBRTC_ALLOW_DEPRECATED_NAMESPACES" ]
-  }
 }
 
 # TODO(bugs.webrtc.org/9693): Remove the possibility to suppress this warning
@@ -532,24 +529,24 @@ if (!build_with_chromium) {
     ]
 
     sources = []
+
     complete_static_lib = true
     suppressed_configs += [ "//build/config/compiler:thin_archive" ]
     defines = []
 
     # RingRTC change to exclude av1 and h264 factories
     deps = [
+      "api:create_modular_peer_connection_factory",
       "api:create_peerconnection_factory",
       "api:enable_media",
       "api:libjingle_peerconnection_api",
       "api:rtc_error",
       "api:transport_api",
-      "api/audio:create_audio_device_module",
       "api/audio_codecs:opus_audio_decoder_factory",
       "api/crypto",
       "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",
@@ -595,9 +592,15 @@ if (!build_with_chromium) {
     if (rtc_enable_protobuf) {
       deps += [ "logging:rtc_event_log_proto" ]
     }
+
+    if (rtc_include_internal_audio_device) {
+      deps += [ "api/audio:create_audio_device_module" ]
+    }
   }
 
-  if (rtc_include_tests && !is_asan) {
+  # TODO(bugs.webrtc.org/430260876): Compile webrtc lib with rust once toolchain
+  # is working.
+  if (rtc_include_tests && !is_asan && !rtc_rusty_base64) {
     rtc_executable("webrtc_lib_link_test") {
       testonly = true
 
@@ -648,6 +651,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "api/numerics:numerics_unittests",
       "api/task_queue:pending_task_safety_flag_unittests",
       "api/test/metrics:metrics_unittests",
+      "api/test/network_emulation:network_queue_unittests",
       "api/transport:stun_unittest",
       "api/transport/rtp:corruption_detection_message_unittest",
       "api/video/test:rtc_api_video_unittests",
@@ -658,6 +662,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "call:fake_network_pipe_unittests",
       "rtc_base:async_dns_resolver_unittests",
       "rtc_base:async_packet_socket_unittest",
+      "rtc_base:async_tcp_socket_unittest",
       "rtc_base:async_udp_socket_unittest",
       "rtc_base:callback_list_unittests",
       "rtc_base:rtc_base_approved_unittests",