M139 -> M138

.gn

@@ -81,8 +81,7 @@ 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

PRESUBMIT.py

@@ -991,6 +991,8 @@ def CommonChecks(input_api, output_api):
             bot_allowlist=[
                 '[email protected]',
                 '[email protected]',
+                ('chrome-cherry-picker'
+                 '@chops-service-accounts.iam.gserviceaccount.com'),
             ]))
     results.extend(
         input_api.canned_checks.CheckChangeTodoHasOwner(

api/BUILD.gn

@@ -216,6 +216,7 @@ rtc_source_set("ice_transport_interface") {
   sources = [ "ice_transport_interface.h" ]
   deps = [
     ":async_dns_resolver",
+    ":local_network_access_permission",
     ":packet_socket_factory",
     ":ref_count",
     ":rtc_error",
@@ -323,6 +324,7 @@ rtc_library("libjingle_peerconnection_api") {
     ":frame_transformer_interface",
     ":ice_transport_interface",
     ":libjingle_logging_api",
+    ":local_network_access_permission",
     ":make_ref_counted",
     ":media_stream_interface",
     ":network_state_predictor_api",
@@ -384,9 +386,11 @@ rtc_library("libjingle_peerconnection_api") {
     "video_codecs:video_codecs_api",
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/base:nullability",
     "//third_party/abseil-cpp/absl/functional:any_invocable",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/strings:str_format",
     "//third_party/abseil-cpp/absl/strings:string_view",
 
     # Basically, don't add stuff here. You might break sensitive downstream
@@ -471,6 +475,17 @@ rtc_source_set("async_dns_resolver") {
   ]
 }
 
+rtc_source_set("local_network_access_permission") {
+  visibility = [ "*" ]
+  sources = [ "local_network_access_permission.h" ]
+  deps = [
+    "../rtc_base:checks",
+    "../rtc_base:socket_address",
+    "../rtc_base/system:rtc_export",
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
+  ]
+}
+
 rtc_source_set("ref_count") {
   visibility = [ "*" ]
   sources = [ "ref_count.h" ]
@@ -704,20 +719,6 @@ if (rtc_include_tests) {
       ]
     }
 
-    rtc_library("test_dependency_factory") {
-      visibility = [ "*" ]
-      testonly = true
-      sources = [
-        "test/test_dependency_factory.cc",
-        "test/test_dependency_factory.h",
-      ]
-      deps = [
-        ":video_quality_test_fixture_api",
-        "../rtc_base:checks",
-        "../rtc_base:platform_thread_types",
-      ]
-    }
-
     rtc_library("create_video_quality_test_fixture_api") {
       visibility = [ "*" ]
       testonly = true
@@ -726,9 +727,6 @@ if (rtc_include_tests) {
         "test/create_video_quality_test_fixture.h",
       ]
       deps = [
-        ":fec_controller_api",
-        ":network_state_predictor_api",
-        ":scoped_refptr",
         ":video_quality_test_fixture_api",
         "../video:video_quality_test",
       ]
@@ -1108,6 +1106,7 @@ if (rtc_include_tests) {
     testonly = true
     sources = [ "test/videocodec_test_fixture.h" ]
     deps = [
+      ":field_trials",
       ":videocodec_test_stats_api",
       "../modules/video_coding:codec_globals_headers",
       "../modules/video_coding:video_codec_interface",
@@ -1394,6 +1393,18 @@ if (rtc_include_tests) {
     ]
   }
 
+  rtc_source_set("mock_local_network_access_permission") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [ "test/mock_local_network_access_permission.h" ]
+    deps = [
+      ":local_network_access_permission",
+      "../rtc_base:socket_address",
+      "../test:test_support",
+      "//third_party/abseil-cpp/absl/functional:any_invocable",
+    ]
+  }
+
   rtc_source_set("mock_rtp") {
     visibility = [ "*" ]
     testonly = true
@@ -1574,40 +1585,28 @@ if (rtc_include_tests) {
     deps = [
       ":array_view",
       ":candidate",
-      ":create_time_controller",
       ":field_trials",
-      ":field_trials_view",
       ":function_view",
       ":libjingle_peerconnection_api",
-      ":peer_connection_quality_test_fixture_api",
       ":rtc_error",
       ":rtc_event_log_output_file",
       ":rtp_headers",
-      ":rtp_headers",
       ":rtp_packet_info",
       ":rtp_parameters",
       ":scoped_refptr",
       ":sequence_checker",
-      ":time_controller",
       "../p2p:p2p_constants",
       "../rtc_base:buffer",
       "../rtc_base:checks",
-      "../rtc_base:gunit_helpers",
       "../rtc_base:logging",
       "../rtc_base:macromagic",
       "../rtc_base:platform_thread",
       "../rtc_base:rtc_event",
       "../rtc_base:socket_address",
-      "../rtc_base:ssl",
       "../rtc_base:task_queue_for_test",
       "../rtc_base/containers:flat_set",
       "../rtc_base/synchronization:sequence_checker_internal",
-      "../rtc_base/task_utils:repeating_task",
-      "../system_wrappers",
-      "../system_wrappers:field_trial",
-      "../test:field_trial",
       "../test:fileutils",
-      "../test:rtc_expect_death",
       "../test:test_support",
       "audio_codecs/opus:unittests",
       "environment:environment_unittests",
@@ -1621,7 +1620,6 @@ if (rtc_include_tests) {
       "video:rtp_video_frame_assembler_unittests",
       "video:video_frame",
       "video:video_frame_metadata_unittest",
-      "//testing/gtest",
       "//third_party/abseil-cpp/absl/functional:any_invocable",
       "//third_party/abseil-cpp/absl/strings",
       "//third_party/abseil-cpp/absl/strings:string_view",
@@ -1696,12 +1694,6 @@ rtc_source_set("field_trials_view") {
   ]
 }
 
-rtc_source_set("webrtc_key_value_config") {
-  visibility = [ "*" ]
-  sources = [ "webrtc_key_value_config.h" ]
-  deps = [ ":field_trials_view" ]
-}
-
 rtc_library("field_trials") {
   visibility = [ "*" ]
   sources = [

api/DEPS

@@ -77,7 +77,7 @@ specific_include_rules = {
   "audio_device_defines\.h": [
     "+rtc_base/strings/string_builder.h",
   ],
-  
+
   "audio_format\.h": [
     "+rtc_base/strings/string_builder.h",
   ],
@@ -107,6 +107,15 @@ specific_include_rules = {
     "+modules/include/module_fec_types.h",
   ],
 
+  "jsep\.h": [
+    "+absl/strings/has_absl_stringify.h",
+    "+absl/strings/str_format.h",
+  ],
+
+  "local_network_access_permission\.h": [
+    "+rtc_base/socket_address.h",
+  ],
+
   "packet_socket_factory\.h": [
     "+rtc_base/async_packet_socket.h",
     "+rtc_base/socket_address.h",
@@ -198,10 +207,6 @@ specific_include_rules = {
     "+rtc_base/thread_annotations.h",
   ],
 
-  "test_dependency_factory\.h": [
-    "+rtc_base/thread_checker.h",
-  ],
-
   "time_controller\.h": [
     "+rtc_base/thread.h",
   ],

api/array_view.h

@@ -22,7 +22,7 @@
 
 namespace webrtc {
 
-// tl;dr: webrtc::ArrayView is the same thing as gsl::span from the Guideline
+// tl;dr: ArrayView is the same thing as gsl::span from the Guideline
 //        Support Library.
 //
 // Many functions read from or write to arrays. The obvious way to do this is
@@ -37,7 +37,7 @@ namespace webrtc {
 //   }
 //
 // This is flexible, since it doesn't matter how the array is stored (C array,
-// std::vector, webrtc::Buffer, ...), but it's error-prone because the caller
+// std::vector, Buffer, ...), but it's error-prone because the caller
 // has to correctly specify the array length:
 //
 //   Contains17(arr, std::size(arr));     // C array
@@ -48,11 +48,11 @@ namespace webrtc {
 // It's also kind of messy to have two separate arguments for what is
 // conceptually a single thing.
 //
-// Enter webrtc::ArrayView<T>. It contains a T pointer (to an array it doesn't
+// Enter ArrayView<T>. It contains a T pointer (to an array it doesn't
 // own) and a count, and supports the basic things you'd expect, such as
 // indexing and iteration. It allows us to write our function like this:
 //
-//   bool Contains17(webrtc::ArrayView<const int> arr) {
+//   bool Contains17(ArrayView<const int> arr) {
 //     for (auto e : arr) {
 //       if (e == 17)
 //         return true;
@@ -65,7 +65,7 @@ namespace webrtc {
 //
 //   Contains17(arr);                             // C array
 //   Contains17(arr);                             // std::vector
-//   Contains17(webrtc::ArrayView<int>(arr, size));  // pointer + size
+//   Contains17(ArrayView<int>(arr, size));  // pointer + size
 //   Contains17(nullptr);                         // nullptr -> empty ArrayView
 //   ...
 //
@@ -239,8 +239,8 @@ class ArrayView final : public array_view_internal::ArrayViewBase<T, Size> {
   // ArrayView<T, N> to ArrayView<T> or ArrayView<const T>,
   // std::vector<T> to ArrayView<T> or ArrayView<const T>,
   // const std::vector<T> to ArrayView<const T>,
-  // webrtc::Buffer to ArrayView<uint8_t> or ArrayView<const uint8_t>, and
-  // const webrtc::Buffer to ArrayView<const uint8_t>.
+  // Buffer to ArrayView<uint8_t> or ArrayView<const uint8_t>, and
+  // const Buffer to ArrayView<const uint8_t>.
   template <
       typename U,
       typename std::enable_if<Size == array_view_internal::kArrayViewVarSize &&

api/async_dns_resolver.h

@@ -79,22 +79,22 @@ class AsyncDnsResolverFactoryInterface {
   // Creates an AsyncDnsResolver and starts resolving the name. The callback
   // will be called when resolution is finished.
   // The callback will be called on the sequence that the caller runs on.
-  virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> CreateAndResolve(
+  virtual std::unique_ptr<AsyncDnsResolverInterface> CreateAndResolve(
       const SocketAddress& addr,
       absl::AnyInvocable<void()> callback) = 0;
   // Creates an AsyncDnsResolver and starts resolving the name to an address
   // matching the specified family. The callback will be called when resolution
   // is finished. The callback will be called on the sequence that the caller
   // runs on.
-  virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> CreateAndResolve(
+  virtual std::unique_ptr<AsyncDnsResolverInterface> CreateAndResolve(
       const SocketAddress& addr,
       int family,
       absl::AnyInvocable<void()> callback) = 0;
   // Creates an AsyncDnsResolver and does not start it.
   // For backwards compatibility, will be deprecated and removed.
   // One has to do a separate Start() call on the
   // resolver to start name resolution.
-  virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> Create() = 0;
+  virtual std::unique_ptr<AsyncDnsResolverInterface> Create() = 0;
 };
 
 }  // namespace webrtc

api/audio/audio_device.h

@@ -21,7 +21,7 @@ namespace webrtc {
 
 class AudioDeviceModuleForTest;
 
-class AudioDeviceModule : public webrtc::RefCountInterface {
+class AudioDeviceModule : public RefCountInterface {
  public:
   enum AudioLayer {
     kPlatformDefaultAudio = 0,

api/audio/audio_frame.cc

@@ -148,7 +148,7 @@ InterleavedView<const int16_t> AudioFrame::data_view() const {
   // If you get a nullptr from `data_view()`, it's likely because the
   // samples_per_channel_ and/or num_channels_ members haven't been properly
   // set. Since `data_view()` returns an InterleavedView<> (which internally
-  // uses webrtc::ArrayView<>), we inherit the behavior in InterleavedView when
+  // uses ArrayView<>), we inherit the behavior in InterleavedView when
   // the view size is 0 that ArrayView<>::data() returns nullptr. So, even when
   // an AudioFrame is muted and we want to return `zeroed_data()`, if
   // samples_per_channel_ or  num_channels_ is 0, the view will point to

api/audio/audio_frame.h

@@ -225,7 +225,7 @@ class AudioFrame {
   // Absolute capture timestamp when this audio frame was originally captured.
   // This is only valid for audio frames captured on this machine. The absolute
   // capture timestamp of a received frame is found in `packet_infos_`.
-  // This timestamp MUST be based on the same clock as webrtc::TimeMillis().
+  // This timestamp MUST be based on the same clock as TimeMillis().
   std::optional<int64_t> absolute_capture_timestamp_ms_;
 };
 

api/audio/audio_processing.h

@@ -11,18 +11,11 @@
 #ifndef API_AUDIO_AUDIO_PROCESSING_H_
 #define API_AUDIO_AUDIO_PROCESSING_H_
 
-// MSVC++ requires this to be set before any other includes to get M_PI.
-#ifndef _USE_MATH_DEFINES
-#define _USE_MATH_DEFINES
-#endif
-
-#include <math.h>
-#include <stddef.h>  // size_t
-#include <stdio.h>   // FILE
-#include <string.h>
-
 #include <array>
+#include <cstddef>
 #include <cstdint>
+#include <cstdio>
+#include <cstring>
 #include <memory>
 #include <optional>
 #include <string>
@@ -622,7 +615,7 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
   // with this chunk of audio.
   virtual void set_stream_key_pressed(bool key_pressed) = 0;
 
-  // Creates and attaches an webrtc::AecDump for recording debugging
+  // Creates and attaches an AecDump for recording debugging
   // information.
   // The `worker_queue` may not be null and must outlive the created
   // AecDump instance. |max_log_size_bytes == -1| means the log size
@@ -641,7 +634,7 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
                                           worker_queue) = 0;
 
   // TODO(webrtc:5298) Deprecated variant.
-  // Attaches provided webrtc::AecDump for recording debugging
+  // Attaches provided AecDump for recording debugging
   // information. Log file and maximum file size logic is supposed to
   // be handled by implementing instance of AecDump. Calling this
   // method when another AecDump is attached resets the active AecDump

api/audio/audio_view.h

@@ -33,7 +33,7 @@ namespace webrtc {
 //   buffer. Channels can be enumerated and accessing the individual channel
 //   data is done via MonoView<>.
 //
-// The views are comparable to and built on webrtc::ArrayView<> but add
+// The views are comparable to and built on ArrayView<> but add
 // audio specific properties for the dimensions of the buffer and the above
 // specialized [de]interleaved support.
 //

api/audio/echo_detector_creator.h

@@ -17,7 +17,7 @@
 namespace webrtc {
 
 // Returns an instance of the WebRTC implementation of a residual echo detector.
-// It can be provided to the webrtc::BuiltinAudioProcessingBuilder to obtain the
+// It can be provided to the BuiltinAudioProcessingBuilder to obtain the
 // usual residual echo metrics.
 scoped_refptr<EchoDetector> CreateEchoDetector();
 

api/audio/test/BUILD.gn

@@ -26,7 +26,6 @@ if (rtc_include_tests) {
       "../..:array_view",
       "../../../modules/audio_processing:aec3_config_json",
       "../../../rtc_base:checks",
-      "../../../rtc_base:macromagic",
       "../../../test:test_support",
     ]
   }