Merge pull request #492 from open-webrtc-toolkit/m88

Merge M88 branch to master.

Author: jianjunz

DEPS

@@ -77,4 +77,14 @@ declare_args() {
 
   # Some non-Chromium builds don't support building java targets.
   enable_java_templates = true
+
+  # If true, it assumes that //third_party/abseil-cpp is an available
+  # dependency for googletest.
+  gtest_enable_absl_printers = true
+
+  # WebRTC doesn't depend on //base from production code but only for testing
+  # purposes. In any case, it doesn't depend on //third_party/perfetto which
+  # is used for base tracing, so this feature is disabled.
+  enable_base_tracing = false
+  use_perfetto_client_library = false
 }

build_overrides/build.gni

@@ -20,10 +20,14 @@
 import re
 import distutils.dir_util
 
+HOME_PATH = os.path.abspath(os.path.join(os.path.dirname(__file__), ".."))
+
 sys.path.append(os.path.join(os.path.dirname(os.path.dirname(__file__)),'build','config','mac'))
+sys.path.append(os.path.join(HOME_PATH, 'third_party', 'webrtc','tools_webrtc','apple'))
 import sdk_info
+import copy_framework_header
+
 
-HOME_PATH = os.path.abspath(os.path.join(os.path.dirname(__file__), ".."))
 OUT_PATH = os.path.join(HOME_PATH, 'out')
 # The lib contains all target architectures and external libs(OpenSSL).
 OUT_LIB_NAME = 'libowt.a'
@@ -79,7 +83,11 @@ def ninjabuild(arch, scheme, targets):
 def copyheaders(headers_target_folder):
   if os.path.exists(headers_target_folder):
     shutil.rmtree(headers_target_folder)
-  shutil.copytree(HEADER_PATH, headers_target_folder)
+    os.mkdir(headers_target_folder)
+  for _, _, file_names in os.walk(HEADER_PATH):
+    for file_name in file_names:
+      copy_framework_header.process(os.path.join(
+          HEADER_PATH, file_name), os.path.join(headers_target_folder, file_name))
 
 def getexternalliblist(ssl_root):
   libs = []
@@ -147,7 +155,7 @@ def getsdkversion():
 
 # Run unit tests on simulator. Return True if all tests are passed.
 def runtest(ssl_root):
-  print 'Start running unit tests.'
+  print('Start running unit tests.')
   # Build app targets checks link issue.
   if not ninjabuild(TEST_ARCH, TEST_SCHEME, SDK_TARGETS + APP_TARGETS + TEST_TARGETS):
     return False
@@ -197,7 +205,7 @@ def main():
         return 1
     if not runtest(opts.ssl_root):
       return 1
-  print 'Done.'
+  print('Done.')
   return 0
 
 if __name__ == '__main__':

scripts/build.py

@@ -17,10 +17,10 @@
 TESTING_PATH = os.path.join(HOME_PATH, 'testing')
 THIRD_PARTY_PATH = os.path.join(HOME_PATH, 'third_party')
 LIBSRTP_PATH = os.path.join(THIRD_PARTY_PATH, 'libsrtp')
-LIBJPEG_TURBO_PATH = os.path.join(THIRD_PARTY_PATH, 'libjpeg_turbo')
 FFMPEG_PATH = os.path.join(THIRD_PARTY_PATH, 'ffmpeg')
 WEBRTC_OVERRIDES_PATH = os.path.join(THIRD_PARTY_PATH, 'webrtc_overrides')
 BUILD_PATH = os.path.join(HOME_PATH, 'build')
+CONFIG_PATH = os.path.join(BUILD_PATH, 'config')
 TOOL_PATH = os.path.join(HOME_PATH, 'tools')
 BASE_PATH = os.path.join(HOME_PATH, 'base')
 platform = os.name
@@ -38,8 +38,7 @@
     ('0007-Fix-examples-path-error.patch', BUILD_PATH),
     ('0008-Disable-loop-range-analysis-when-build-with-Xcode-cl.patch', BUILD_PATH),
     ('0009-Export-WebRTC-symbols-on-iOS.patch', BUILD_PATH),
-    ('0010-mac-make-find_sdk.py-work-when-the-sdk-goes-to-11.patch', BUILD_PATH),
-    ('0011-libjpeg_turbo-fix-for-CVE-2018-20330-and-19664.patch', LIBJPEG_TURBO_PATH),
+    ('0010-Restore-is_ash-for-backward-compatible.patch', BUILD_PATH),
     ('0013-Remove-unused-gni-for-av1-build.patch', THIRD_PARTY_PATH),
     ('0014-Fix-missing-ffmpeg-configure-item-for-msvc-build.patch', FFMPEG_PATH)
 ]
@@ -53,6 +52,10 @@ def _patch(ignoreFailures=False):
                                 cwd=applyPath)
             else:
                 sys.exit(1)
+    # create empty gclient_args.gni under bulid/config if not already
+    gclientArgPath = os.path.join(CONFIG_PATH, 'gclient_args.gni')
+    if not os.path.isfile(gclientArgPath):
+        open(gclientArgPath, "w").close()
 
 def main(argv):
   _patch()

scripts/prepare_dev.py

@@ -178,20 +178,13 @@ static_library("owt_sdk_base") {
       include_dirs += [ owt_quic_header_root ]
     }
   }
-  
+
   if (is_win || is_linux) {
     # Custom audio/video input and output.
     # When rebasing libwebrtc to a new version, custom audio/video input/output
     # usually need some changes to fit the latest API.
     defines += [ "OWT_CUSTOM_AVIO" ]
   }
-  if (is_ios) {
-    sources += [
-      "sdk/base/ios/networkmonitorios.cc",
-      "sdk/base/ios/networkmonitorios.h",
-    ]
-    public_deps += [ "//third_party/webrtc/sdk:framework_objc+link" ]
-  }
   if (is_win) {
     sources += [
       "sdk/base/desktopcapturer.cc",
@@ -341,9 +334,14 @@ if (is_ios) {
       ":owt_sdk_p2p",
       "//third_party/jsoncpp",
     ]
+    if (is_ios) {
+      deps += [ "//third_party/webrtc/sdk:framework_objc+link" ]
+    }
     include_dirs = [
       "//third_party",
-      "//third_party/webrtc/sdk/objc/Framework/Headers",
+      "//third_party/webrtc/sdk/objc/api/peerconnection",
+      "//third_party/webrtc/sdk/objc/api/logging",
+      "//third_party/webrtc/sdk/objc/api/video_codec",
       "sdk/include/cpp",
       "sdk/include/objc",
     ]
@@ -362,8 +360,6 @@ if (is_ios) {
       "sdk/base/objc/OWTDefaultVideoEncoderFactory.h",
       "sdk/base/objc/OWTDefaultVideoEncoderFactory.m",
       "sdk/base/objc/OWTErrors.m",
-
-      #"sdk/base/objc/OWTGlobalConfiguration.mm",
       "sdk/base/objc/OWTLocalStream+Private.h",
       "sdk/base/objc/OWTLocalStream.mm",
       "sdk/base/objc/OWTMediaFormat+Internal.h",
@@ -382,8 +378,6 @@ if (is_ios) {
       "sdk/base/objc/RemoteStreamObserverObjcImpl.mm",
       "sdk/base/objc/public/OWTErrors.h",
       "sdk/base/objc/public/OWTFrameGeneratorProtocol.h",
-
-      #"sdk/base/objc/public/OWTGlobalConfiguration.h",
       "sdk/base/objc/public/OWTLocalCameraStream.h",
       "sdk/base/objc/public/OWTLocalCameraStreamParameters.h",
       "sdk/base/objc/public/OWTLocalCustomizedStream.h",

talk/owt/BUILD.gn

@@ -1,27 +1,27 @@
-From 4928b9815f8063c4dd832d3a3a61355d60c5a0cc Mon Sep 17 00:00:00 2001
+From b17939a2d0c21ae5cfd6956ed98b47789d4dee8f Mon Sep 17 00:00:00 2001
 From: Qiu Jainlin <[email protected]>
-Date: Thu, 7 May 2020 09:30:40 +0800
+Date: Mon, 4 Jan 2021 15:49:43 +0800
 Subject: [PATCH] Use OpenSSL for usrsctp
 
 ---
  usrsctp/BUILD.gn | 10 +++++++++-
  1 file changed, 9 insertions(+), 1 deletion(-)
 
 diff --git a/usrsctp/BUILD.gn b/usrsctp/BUILD.gn
-index e8a5b1f9d41..684603c4ce3 100644
+index 0c5947f1785..7aff0d50687 100644
 --- a/usrsctp/BUILD.gn
 +++ b/usrsctp/BUILD.gn
 @@ -3,6 +3,7 @@
  # found in the LICENSE file.
 
  import("//build/toolchain/toolchain.gni")
 +import("//build_overrides/ssl/ssl.gni")
+ import("//testing/libfuzzer/fuzzer_test.gni")
 
- config("usrsctp_config") {
-   include_dirs = [
-@@ -145,5 +146,12 @@ static_library("usrsctp") {
-   if (is_fuchsia) {
-     defines += [ "__Userspace_os_Fuchsia" ]
+ config("usrsctp_public_config") {
+@@ -161,7 +162,14 @@ static_library("usrsctp") {
+   if (use_fuzzing_engine) {
+     defines += [ "INVARIANTS" ]
    }
 -  deps = [ "//third_party/boringssl" ]
 +
@@ -33,6 +33,8 @@ index e8a5b1f9d41..684603c4ce3 100644
 +    ]
 +  }
  }
+ 
+ fuzzer_test("usrsctp_fuzzer_listen") {
 -- 
-2.24.1.windows.2
+2.28.0.windows.1
 

talk/owt/patches/0001-Use-OpenSSL-for-usrsctp.patch

@@ -1,26 +1,26 @@
-From a97d58537750064c1d4929a2c141e52fe0206ff9 Mon Sep 17 00:00:00 2001
+From 690f2315a218e771f3de98297bca456ca48b5f89 Mon Sep 17 00:00:00 2001
 From: Qiu Jainlin <[email protected]>
-Date: Thu, 7 May 2020 09:57:47 +0800
+Date: Mon, 4 Jan 2021 12:44:41 +0800
 Subject: [PATCH] Remove webrtc_overrides
 
 ---
- webrtc_overrides/BUILD.gn                     | 168 -------------
+ webrtc_overrides/BUILD.gn                     | 179 --------------
  webrtc_overrides/DEPS                         |   7 -
  webrtc_overrides/OWNERS                       |   6 -
  webrtc_overrides/README.chromium              |  20 --
  webrtc_overrides/field_trial.cc               |  15 --
- webrtc_overrides/init_webrtc.cc               |  45 ----
+ webrtc_overrides/init_webrtc.cc               |  44 ----
  webrtc_overrides/init_webrtc.h                |  15 --
  webrtc_overrides/metrics.cc                   |  57 -----
  .../rtc_base/diagnostic_logging.h             | 126 ----------
  webrtc_overrides/rtc_base/event.cc            |  40 ---
  webrtc_overrides/rtc_base/event.h             |  48 ----
  webrtc_overrides/rtc_base/gtest_prod_util.h   |  10 -
- webrtc_overrides/rtc_base/logging.cc          | 230 ------------------
+ webrtc_overrides/rtc_base/logging.cc          | 231 ------------------
  webrtc_overrides/rtc_base/logging.h           | 123 ----------
- webrtc_overrides/task_queue_factory.cc        | 144 -----------
+ webrtc_overrides/task_queue_factory.cc        | 143 -----------
  webrtc_overrides/task_queue_factory.h         |  24 --
- 16 files changed, 1078 deletions(-)
+ 16 files changed, 1088 deletions(-)
  delete mode 100644 webrtc_overrides/BUILD.gn
  delete mode 100644 webrtc_overrides/DEPS
  delete mode 100644 webrtc_overrides/OWNERS
@@ -40,10 +40,10 @@ Subject: [PATCH] Remove webrtc_overrides
 
 diff --git a/webrtc_overrides/BUILD.gn b/webrtc_overrides/BUILD.gn
 deleted file mode 100644
-index 4f44c48d26b..00000000000
+index f62ec73be77..00000000000
 --- a/webrtc_overrides/BUILD.gn
 +++ /dev/null
-@@ -1,168 +0,0 @@
+@@ -1,179 +0,0 @@
 -# Copyright 2017 The Chromium Authors. All rights reserved.
 -# Use of this source code is governed by a BSD-style license that can be
 -# found in the LICENSE file.
@@ -58,15 +58,18 @@ index 4f44c48d26b..00000000000
 -    ":metrics",
 -    ":task_queue_factory",
 -    "//third_party/webrtc/api:callfactory_api",
+-    "//third_party/webrtc/api:frame_transformer_interface",
 -    "//third_party/webrtc/api:ice_transport_factory",
 -    "//third_party/webrtc/api:libjingle_logging_api",
 -    "//third_party/webrtc/api:libjingle_peerconnection_api",
 -    "//third_party/webrtc/api:media_stream_interface",
 -    "//third_party/webrtc/api:packet_socket_factory",
 -    "//third_party/webrtc/api:rtc_error",
 -    "//third_party/webrtc/api:rtc_stats_api",
+-    "//third_party/webrtc/api:rtp_packet_info",
 -    "//third_party/webrtc/api:rtp_parameters",
 -    "//third_party/webrtc/api:scoped_refptr",
+-    "//third_party/webrtc/api/adaptation:resource_adaptation_api",
 -    "//third_party/webrtc/api/audio:aec3_config",
 -    "//third_party/webrtc/api/audio:aec3_config_json",
 -    "//third_party/webrtc/api/audio:aec3_factory",
@@ -84,16 +87,19 @@ index 4f44c48d26b..00000000000
 -    "//third_party/webrtc/api/audio_codecs/opus:audio_encoder_multiopus",
 -    "//third_party/webrtc/api/audio_codecs/opus:audio_encoder_opus",
 -    "//third_party/webrtc/api/rtc_event_log:rtc_event_log_factory",
+-    "//third_party/webrtc/api/transport:enums",
+-    "//third_party/webrtc/api/video:recordable_encoded_frame",
 -    "//third_party/webrtc/api/video:video_bitrate_allocation",
 -    "//third_party/webrtc/api/video:video_frame",
--    "//third_party/webrtc/api/video:video_frame_i420",
+-    "//third_party/webrtc/api/video:video_frame_metadata",
 -    "//third_party/webrtc/api/video:video_rtp_headers",
 -    "//third_party/webrtc/api/video_codecs:builtin_video_decoder_factory",
 -    "//third_party/webrtc/api/video_codecs:rtc_software_fallback_wrappers",
 -    "//third_party/webrtc/api/video_codecs:video_codecs_api",
 -    "//third_party/webrtc/common_video",
 -    "//third_party/webrtc/common_video:common_video",
 -    "//third_party/webrtc/media:rtc_audio_video",
+-    "//third_party/webrtc/media:rtc_h264_profile_id",
 -    "//third_party/webrtc/media:rtc_internal_video_codecs",
 -    "//third_party/webrtc/media:rtc_media",
 -    "//third_party/webrtc/media:rtc_media_base",
@@ -135,9 +141,14 @@ index 4f44c48d26b..00000000000
 -  ]
 -  if (is_chromecast) {
 -    public_deps += [
+-      "//third_party/webrtc/api:network_state_predictor_api",
 -      "//third_party/webrtc/api/audio:audio_frame_api",
 -      "//third_party/webrtc/api/task_queue",
+-      "//third_party/webrtc/api/transport:goog_cc",
+-      "//third_party/webrtc/api/transport:network_control",
+-      "//third_party/webrtc/api/units:time_delta",
 -      "//third_party/webrtc/api/video:encoded_image",
+-      "//third_party/webrtc/call:call_interfaces",
 -      "//third_party/webrtc/media:rtc_h264_profile_id",
 -      "//third_party/webrtc/media:rtc_media_engine_defaults",
 -      "//third_party/webrtc/modules/audio_device:audio_device_default",
@@ -286,10 +297,10 @@ index 878e6cc576b..00000000000
 -}  // namespace webrtc
 diff --git a/webrtc_overrides/init_webrtc.cc b/webrtc_overrides/init_webrtc.cc
 deleted file mode 100644
-index db086b6f758..00000000000
+index effc5796caa..00000000000
 --- a/webrtc_overrides/init_webrtc.cc
 +++ /dev/null
-@@ -1,45 +0,0 @@
+@@ -1,44 +0,0 @@
 -// Copyright 2013 The Chromium Authors. All rights reserved.
 -// Use of this source code is governed by a BSD-style license that can be
 -// found in the LICENSE file.
@@ -299,7 +310,6 @@ index db086b6f758..00000000000
 -#include "base/command_line.h"
 -#include "base/files/file_path.h"
 -#include "base/files/file_util.h"
--#include "base/logging.h"
 -#include "base/native_library.h"
 -#include "base/path_service.h"
 -#include "base/trace_event/trace_event.h"
@@ -669,10 +679,10 @@ index d57937aa63b..00000000000
 -#endif  // THIRD_PARTY_WEBRTC_OVERRIDES_RTC_BASE_GTEST_PROD_UTIL_H_
 diff --git a/webrtc_overrides/rtc_base/logging.cc b/webrtc_overrides/rtc_base/logging.cc
 deleted file mode 100644
-index d10fd254db1..00000000000
+index 87ae618c496..00000000000
 --- a/webrtc_overrides/rtc_base/logging.cc
 +++ /dev/null
-@@ -1,230 +0,0 @@
+@@ -1,231 +0,0 @@
 -// Copyright 2015 The Chromium Authors. All rights reserved.
 -// Use of this source code is governed by a BSD-style license that can be
 -// found in the LICENSE file.
@@ -691,6 +701,7 @@ index d10fd254db1..00000000000
 -
 -#include "base/atomicops.h"
 -#include "base/logging.h"
+-#include "base/notreached.h"
 -#include "base/strings/string_util.h"
 -#include "base/threading/platform_thread.h"
 -#include "third_party/webrtc/rtc_base/string_utils.h"
@@ -1034,10 +1045,10 @@ index 5513bc6ef58..00000000000
 -#endif  // THIRD_PARTY_WEBRTC_OVERRIDES_WEBRTC_RTC_BASE_LOGGING_H_
 diff --git a/webrtc_overrides/task_queue_factory.cc b/webrtc_overrides/task_queue_factory.cc
 deleted file mode 100644
-index e81ef238494..00000000000
+index 2898c95ef09..00000000000
 --- a/webrtc_overrides/task_queue_factory.cc
 +++ /dev/null
-@@ -1,144 +0,0 @@
+@@ -1,143 +0,0 @@
 -// Copyright 2019 The Chromium Authors. All rights reserved.
 -// Use of this source code is governed by a BSD-style license that can be
 -// found in the LICENSE file.
@@ -1146,7 +1157,6 @@ index e81ef238494..00000000000
 -#else
 -      return {base::TaskPriority::HIGHEST};
 -#endif
--      break;
 -    case webrtc::TaskQueueFactory::Priority::LOW:
 -      return {base::MayBlock(), base::TaskPriority::BEST_EFFORT};
 -    case webrtc::TaskQueueFactory::Priority::NORMAL:
@@ -1213,5 +1223,5 @@ index 957cf75e4e9..00000000000
 -
 -#endif  // THIRD_PARTY_WEBRTC_OVERRIDES_TASK_QUEUE_FACTORY_H_
 -- 
-2.24.1.windows.2
+2.28.0.windows.1