Update to 4896 (M100) (#72)

.gn

@@ -63,4 +63,8 @@ default_args = {
   enable_libaom = false
 
   gtest_enable_absl_printers = true
+
+  # RingRTC change to support SDK >= 19.
+  # Differently from Chromium, WebRTC still support SDK 19.
+  default_min_sdk_version = 19
 }

.mailmap

@@ -0,0 +1 @@
+Tommi <tommi@webrtc.org> Tomas Gunnarsson <tommi@webrtc.org>

.style.yapf

@@ -0,0 +1,4 @@
+[style]
+based_on_style = pep8
+indent_width = 2
+column_limit = 80

.vpython3

@@ -0,0 +1,82 @@
+# This is a vpython "spec" file.
+#
+# It describes patterns for python wheel dependencies of the python scripts in
+# the chromium repo, particularly for dependencies that have compiled components
+# (since pure-python dependencies can be easily vendored into third_party).
+#
+# When vpython is invoked, it finds this file and builds a python VirtualEnv,
+# containing all of the dependencies described in this file, fetching them from
+# CIPD (the "Chrome Infrastructure Package Deployer" service). Unlike `pip`,
+# this never requires the end-user machine to have a working python extension
+# compilation environment. All of these packages are built using:
+#   https://chromium.googlesource.com/infra/infra/+/main/infra/tools/dockerbuild/
+#
+# All python scripts in the repo share this same spec, to avoid dependency
+# fragmentation.
+#
+# If you have depot_tools installed in your $PATH, you can invoke python scripts
+# in this repo by running them as you normally would run them, except
+# substituting `vpython` instead of `python` on the command line, e.g.:
+#   vpython path/to/script.py some --arguments
+#
+# Read more about `vpython` and how to modify this file here:
+#   https://chromium.googlesource.com/infra/infra/+/main/doc/users/vpython.md
+
+python_version: "3.8"
+
+# Used by:
+#   third_party/catapult
+wheel: <
+  name: "infra/python/wheels/psutil/${vpython_platform}"
+  version: "version:5.8.0.chromium.2"
+>
+
+# Used by tools_webrtc/perf/webrtc_dashboard_upload.py.
+wheel: <
+  name: "infra/python/wheels/httplib2-py3"
+  version: "version:0.19.1"
+>
+
+wheel: <
+  name: "infra/python/wheels/pyparsing-py2_py3"
+  version: "version:2.4.7"
+>
+
+
+# Used by:
+#   build/toolchain/win
+wheel: <
+  name: "infra/python/wheels/pywin32/${vpython_platform}"
+  version: "version:300"
+  match_tag: <
+    platform: "win32"
+  >
+  match_tag: <
+    platform: "win_amd64"
+  >
+>
+
+wheel: <
+  name: "infra/python/wheels/six-py2_py3"
+  version: "version:1.15.0"
+>
+wheel: <
+  name: "infra/python/wheels/pbr-py2_py3"
+  version: "version:3.0.0"
+>
+wheel: <
+  name: "infra/python/wheels/funcsigs-py2_py3"
+  version: "version:1.0.2"
+>
+wheel: <
+  name: "infra/python/wheels/mock-py2_py3"
+  version: "version:2.0.0"
+>
+wheel: <
+  name: "infra/python/wheels/protobuf-py2_py3"
+  version: "version:3.13.0"
+>
+wheel: <
+  name: "infra/python/wheels/requests-py2_py3"
+  version: "version:2.13.0"
+>

AUTHORS

@@ -58,6 +58,7 @@ Jesús Leganés-Combarro <piranna@gmail.com>
 Jiawei Ou <jiawei.ou@gmail.com>
 Jie Mao <maojie0924@gmail.com>
 Jiwon Kim <jwkim0000@gmail.com>
+Johnny Wong <hellojinqiang@gmail.com>
 Jose Antonio Olivera Ortega <josea.olivera@gmail.com>
 Keiichi Enomoto <enm10k@gmail.com>
 Kiran Thind <kiran.thind@gmail.com>
@@ -69,6 +70,7 @@ Mallikarjuna Rao V <vm.arjun@samsung.com>
 Manish Jethani <manish.jethani@gmail.com>
 Martin Storsjo <martin@martin.st>
 Matthias Liebig <matthias.gcode@gmail.com>
+Maksim Sisov <msisov@igalia.com>
 Maxim Pavlov <pavllovmax@gmail.com>
 Maxim Potapov <vopatop.skam@gmail.com>
 Michael Iedema <michael@kapsulate.com>
@@ -77,6 +79,8 @@ Miguel Paris <mparisdiaz@gmail.com>
 Mike Gilbert <floppymaster@gmail.com>
 Min Wang <mingewang@gmail.com>
 Mo Zanaty <mzanaty@cisco.com>
+Niek van der Maas <mail@niekvandermaas.nl>
+Olivier Crête <olivier.crete@ocrete.ca>
 Pali Rohar
 Paul Kapustin <pkapustin@gmail.com>
 Peng Yu <yupeng323@gmail.com>
@@ -88,16 +92,19 @@ Raman Budny <budnyjj@gmail.com>
 Ramprakash Jelari <ennajelari@gmail.com>
 Riku Voipio <riku.voipio@linaro.org>
 Robert Bares <robert@bares.me>
+Robert Mader <robert.mader@posteo.de>
 Robert Nagy <robert.nagy@gmail.com>
 Ryan Yoakum <ryoakum@skobalt.com>
 Sarah Thompson <sarah@telergy.com>
 Satender Saroha <ssaroha@yahoo.com>
 Saul Kravitz <Saul.Kravitz@celera.com>
 Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
+Shuhai Peng <shuhai.peng@intel.com>
 Silviu Caragea <silviu.cpp@gmail.com>
 Stefan Gula <steweg@gmail.com>
 Stephan Hartmann <stha09@googlemail.com>
 Steve Reid <sreid@sea-to-sky.net>
+Takaaki Suzuki <takaakisuzuki.14@gmail.com>
 Tarun Chawla <trnkumarchawla@gmail.com>
 Todd Wong <todd.wong.ndq@gmail.com>
 Tomas Popela <tomas.popela@gmail.com>
@@ -108,6 +115,7 @@ Victor Costan <costan@gmail.com>
 Vladimir Beloborodov <VladimirTechMan@gmail.com>
 Xiaohong Xu <freemine@yeah.net>
 Xiaolei Yu <dreifachstein@gmail.com>
+Yaowen Guo <albertguo88@gmail.com>
 Yura Yaroshevich <yura.yaroshevich@gmail.com>
 Yuriy Pavlyshak <yuriy@appear.in>
 Yusuke Suzuki <utatane.tea@gmail.com>
@@ -115,6 +123,7 @@ Yusuke Suzuki <utatane.tea@gmail.com>
 
 # BEGIN organizations section.
 8x8 Inc. <*@8x8.com>
+8x8 Inc. <*@jitsi.org>
 8x8 Inc. <*@sip-communicator.org>
 Agora IO <*@agora.io>
 ARM Holdings <*@arm.com>
@@ -130,6 +139,7 @@ Microsoft Corporation <*@microsoft.com>
 MIPS Technologies <*@mips.com>
 Mozilla Foundation <*@mozilla.com>
 Netgem S.A. <*@netgem.com>
+Nutanix Inc. <*@nutanix.com>
 NVIDIA Corporation <*@nvidia.com>
 Opera Software ASA <*@opera.com>
 Optical Tone Ltd <*@opticaltone.com>

BUILD.gn

@@ -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("//third_party/google_benchmark/buildconfig.gni")
@@ -39,7 +48,6 @@ if (!build_with_chromium) {
     if (rtc_include_tests) {
       deps += [
         ":rtc_unittests",
-        ":slow_tests",
         ":video_engine_tests",
         ":voip_unittests",
         ":webrtc_nonparallel_tests",
@@ -73,6 +81,13 @@ 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",
@@ -84,7 +99,7 @@ if (!build_with_chromium) {
       }
       if (rtc_enable_protobuf) {
         deps += [
-          "audio:low_bandwidth_audio_test",
+          "audio:low_bandwidth_audio_perf_test",
           "logging:rtc_event_log_rtp_dump",
           "tools_webrtc/perf:webrtc_dashboard_upload",
         ]
@@ -118,15 +133,15 @@ config("common_inherited_config") {
   cflags = []
   ldflags = []
 
-  if (rtc_enable_symbol_export || is_component_build) {
+  if (rtc_dlog_always_on) {
-    defines = [ "WEBRTC_ENABLE_SYMBOL_EXPORT" ]
+    defines += [ "DLOG_ALWAYS_ON" ]
-  }
-  if (rtc_enable_objc_symbol_export) {
-    defines = [ "WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT" ]
   }
 
-  if (build_with_mozilla) {
+  if (rtc_enable_symbol_export || is_component_build) {
-    defines += [ "WEBRTC_MOZILLA_BUILD" ]
+    defines += [ "WEBRTC_ENABLE_SYMBOL_EXPORT" ]
+  }
+  if (rtc_enable_objc_symbol_export) {
+    defines += [ "WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT" ]
   }
 
   if (!rtc_builtin_ssl_root_certificates) {
@@ -222,14 +237,6 @@ 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") {
@@ -269,6 +276,10 @@ config("common_config") {
     defines += [ "RTC_ENABLE_VP9" ]
   }
 
+  if (rtc_include_dav1d_in_internal_decoder_factory) {
+    defines += [ "RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY" ]
+  }
+
   if (rtc_enable_sctp) {
     defines += [ "WEBRTC_HAVE_SCTP" ]
   }
@@ -305,7 +316,10 @@ config("common_config") {
     defines += [ "WEBRTC_EXCLUDE_AUDIO_PROCESSING_MODULE" ]
   }
 
-  cflags = []
+  # TODO(webrtc:13219): Fix -Wshadow instances and enable.
+  if (is_clang) {
+    cflags += [ "-Wno-shadow" ]
+  }
 
   if (build_with_chromium) {
     defines += [
@@ -346,21 +360,10 @@ config("common_config") {
         "-Wundef",
       ]
 
-      # use_xcode_clang only refers to the iOS toolchain, host binaries use
+      if (!is_nacl) {
-      # chromium's clang always.
+        # Flags NaCl (Clang 3.7) do not recognize.
-      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.
         cflags += [ "-Wunused-lambda-capture" ]
       }
-
-      if (use_xcode_clang) {
-        # This may be removed if the clang version in xcode > 12.4 includes the
-        # fix https://reviews.llvm.org/D73007.
-        # https://bugs.llvm.org/show_bug.cgi?id=44556
-        cflags += [ "-Wno-range-loop-analysis" ]
-      }
     }
 
     if (is_win && !is_clang) {
@@ -575,7 +578,9 @@ if (rtc_include_tests && !build_with_chromium) {
       "rtc_base/experiments:experiments_unittests",
       "rtc_base/system:file_wrapper_unittests",
       "rtc_base/task_utils:pending_task_safety_flag_unittests",
+      "rtc_base/task_utils:repeating_task_unittests",
       "rtc_base/task_utils:to_queued_task_unittests",
+      "rtc_base/units:units_unittests",
       "sdk:sdk_tests",
       "test:rtp_test_utils",
       "test:test_main",
@@ -613,17 +618,6 @@ if (rtc_include_tests && !build_with_chromium) {
     }
   }
 
-  # 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",
@@ -650,6 +644,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "test:test_common",
       "test:test_main",
       "test:video_test_common",
+      "video:video_legacy_tests",
       "video:video_tests",
       "video/adaptation:video_adaptation_tests",
     ]
@@ -746,6 +741,9 @@ group("poison_audio_codecs") {
 group("poison_default_task_queue") {
 }
 
+group("poison_default_echo_detector") {
+}
+
 group("poison_rtc_json") {
 }
 

ENG_REVIEW_OWNERS

@@ -7,4 +7,3 @@
 
 danilchap@webrtc.org
 mbonadei@webrtc.org
-phoglund@webrtc.org

OWNERS

@@ -10,9 +10,13 @@ per-file .../BUILD.gn=mbonadei@webrtc.org
 per-file *.gni=mbonadei@webrtc.org
 per-file .../*.gni=mbonadei@webrtc.org
 per-file .vpython=mbonadei@webrtc.org
+per-file .vpython3=mbonadei@webrtc.org
 per-file AUTHORS=*
 per-file DEPS=*
 per-file pylintrc=mbonadei@webrtc.org
 per-file WATCHLISTS=*
 per-file native-api.md=mbonadei@webrtc.org
 per-file ....lua=titovartem@webrtc.org
+per-file .style.yapf=jleconte@webrtc.org
+per-file *.py=jansson@webrtc.org
+per-file *.py=jleconte@webrtc.org

PRESUBMIT.py

@@ -1,3 +1,5 @@
+#!/usr/bin/env vpython3
+
 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
@@ -13,6 +15,9 @@ import sys
 from collections import defaultdict
 from contextlib import contextmanager
 
+# Runs PRESUBMIT.py in py3 mode by git cl presubmit.
+USE_PYTHON3 = True
+
 # Files and directories that are *skipped* by cpplint in the presubmit script.
 CPPLINT_EXCEPTIONS = [
     'api/video_codecs/video_decoder.h',
@@ -31,6 +36,9 @@ CPPLINT_EXCEPTIONS = [
     'modules/video_capture',
     'p2p/base/pseudo_tcp.cc',
     'p2p/base/pseudo_tcp.h',
+    'PRESUBMIT.py',
+    'presubmit_test_mocks.py',
+    'presubmit_test.py',
     'rtc_base',
     'sdk/android/src/jni',
     'sdk/objc',
@@ -137,8 +145,8 @@ def VerifyNativeApiHeadersListIsValid(input_api, output_api):
   """Ensures the list of native API header directories is up to date."""
   non_existing_paths = []
   native_api_full_paths = [
-        input_api.os_path.join(input_api.PresubmitLocalPath(),
+      input_api.os_path.join(input_api.PresubmitLocalPath(), *path.split('/'))
-                               *path.split('/')) for path in API_DIRS
+      for path in API_DIRS
   ]
   for path in native_api_full_paths:
     if not os.path.isdir(path):
@@ -200,8 +208,8 @@ def CheckNoIOStreamInHeaders(input_api, output_api, source_file_filter):
   files = []
   pattern = input_api.re.compile(r'^#include\s*<iostream>',
                                  input_api.re.MULTILINE)
-    file_filter = lambda x: (input_api.FilterSourceFile(x) and
+  file_filter = lambda x: (input_api.FilterSourceFile(x) and source_file_filter(
-                             source_file_filter(x))
+      x))
   for f in input_api.AffectedSourceFiles(file_filter):
     if not f.LocalPath().endswith('.h'):
       continue
@@ -209,13 +217,12 @@ def CheckNoIOStreamInHeaders(input_api, output_api, source_file_filter):
     if pattern.search(contents):
       files.append(f)
 
-    if len(files):
+  if len(files) > 0:
     return [
         output_api.PresubmitError(
             'Do not #include <iostream> in header files, since it inserts '
             'static initialization into every file including the header. '
-                'Instead, #include <ostream>. See http://crbug.com/94794',
+            'Instead, #include <ostream>. See http://crbug.com/94794', files)
-                files)
     ]
   return []
 
@@ -224,8 +231,8 @@ def CheckNoPragmaOnce(input_api, output_api, source_file_filter):
   """Make sure that banned functions are not used."""
   files = []
   pattern = input_api.re.compile(r'^#pragma\s+once', input_api.re.MULTILINE)
-    file_filter = lambda x: (input_api.FilterSourceFile(x) and
+  file_filter = lambda x: (input_api.FilterSourceFile(x) and source_file_filter(
-                             source_file_filter(x))
+      x))
   for f in input_api.AffectedSourceFiles(file_filter):
     if not f.LocalPath().endswith('.h'):
       continue
@@ -238,11 +245,11 @@ def CheckNoPragmaOnce(input_api, output_api, source_file_filter):
         output_api.PresubmitError(
             'Do not use #pragma once in header files.\n'
             'See http://www.chromium.org/developers/coding-style'
-                '#TOC-File-headers',
+            '#TOC-File-headers', files)
-                files)
     ]
   return []
 
+
 def CheckNoFRIEND_TEST(# pylint: disable=invalid-name
         input_api,
         output_api,
@@ -308,8 +315,7 @@ def CheckApprovedFilesLintClean(input_api, output_api,
   files = []
   for f in input_api.AffectedSourceFiles(source_file_filter):
     # Note that moved/renamed files also count as added.
-        if f.Action() == 'A' or not IsLintDisabled(disabled_paths,
+    if f.Action() == 'A' or not IsLintDisabled(disabled_paths, f.LocalPath()):
-                                                   f.LocalPath()):
       files.append(f.AbsoluteLocalPath())
 
   for file_name in files:
@@ -337,8 +343,7 @@ def CheckNoSourcesAbove(input_api, gn_files, output_api):
     for source_block_match in source_pattern.finditer(contents):
       # Find all source list entries starting with ../ in the source block
       # (exclude overrides entries).
-            for file_list_match in file_pattern.finditer(
+      for file_list_match in file_pattern.finditer(source_block_match.group(1)):
-                    source_block_match.group(1)):
         source_file = file_list_match.group(1)
         if 'overrides/' not in source_file:
           violating_source_entries.append(source_file)
@@ -370,6 +375,7 @@ def CheckAbseilDependencies(input_api, gn_files, output_api):
                'should be moved to the "absl_deps" parameter.')
   errors = []
 
+  # pylint: disable=too-many-nested-blocks
   for gn_file in gn_files:
     gn_file_content = input_api.ReadFile(gn_file)
     for target_match in TARGET_RE.finditer(gn_file_content):
@@ -382,8 +388,7 @@ def CheckAbseilDependencies(input_api, gn_files, output_api):
           for dep in deps:
             if re.search(absl_re, dep):
               errors.append(
-                                output_api.PresubmitError(
+                  output_api.PresubmitError(error_msg %
-                                    error_msg %
                                             (target_name, gn_file.LocalPath())))
               break  # no need to warn more than once per target
   return errors
@@ -398,7 +403,7 @@ def CheckNoMixingSources(input_api, gn_files, output_api):
   def _MoreThanOneSourceUsed(*sources_lists):
     sources_used = 0
     for source_list in sources_lists:
-            if len(source_list):
+      if len(source_list) > 0:
         sources_used += 1
     return sources_used > 1
 
@@ -432,8 +437,7 @@ def CheckNoMixingSources(input_api, gn_files, output_api):
           c_files = []
           cc_files = []
           objc_files = []
-                for file_match in FILE_PATH_RE.finditer(
+        for file_match in FILE_PATH_RE.finditer(sources_match.group(1)):
-                        sources_match.group(1)):
           file_path = file_match.group('file_path')
           extension = file_match.group('extension')
           if extension == '.c':
@@ -444,12 +448,9 @@ def CheckNoMixingSources(input_api, gn_files, output_api):
             objc_files.append(file_path + extension)
       list_of_sources.append((c_files, cc_files, objc_files))
       for c_files_list, cc_files_list, objc_files_list in list_of_sources:
-                if _MoreThanOneSourceUsed(c_files_list, cc_files_list,
+        if _MoreThanOneSourceUsed(c_files_list, cc_files_list, objc_files_list):
-                                          objc_files_list):
+          all_sources = sorted(c_files_list + cc_files_list + objc_files_list)
-                    all_sources = sorted(c_files_list + cc_files_list +
+          errors[gn_file.LocalPath()].append((target_name, all_sources))
-                                         objc_files_list)
-                    errors[gn_file.LocalPath()].append(
-                        (target_name, all_sources))
   if errors:
     return [
         output_api.PresubmitError(
@@ -459,7 +460,7 @@ def CheckNoMixingSources(input_api, gn_files, output_api):
             'Mixed sources: \n'
             '%s\n'
             'Violating GN files:\n%s\n' %
-                (json.dumps(errors, indent=2), '\n'.join(errors.keys())))
+            (json.dumps(errors, indent=2), '\n'.join(list(errors.keys()))))
     ]
   return []
 
@@ -467,18 +468,16 @@ def CheckNoMixingSources(input_api, gn_files, output_api):
 def CheckNoPackageBoundaryViolations(input_api, gn_files, output_api):
   cwd = input_api.PresubmitLocalPath()
   with _AddToPath(
-            input_api.os_path.join(cwd, 'tools_webrtc',
+      input_api.os_path.join(cwd, 'tools_webrtc', 'presubmit_checks_lib')):
-                                   'presubmit_checks_lib')):
     from check_package_boundaries import CheckPackageBoundaries
-    build_files = [
+  build_files = [os.path.join(cwd, gn_file.LocalPath()) for gn_file in gn_files]
-        os.path.join(cwd, gn_file.LocalPath()) for gn_file in gn_files
-    ]
   errors = CheckPackageBoundaries(cwd, build_files)[:5]
   if errors:
     return [
         output_api.PresubmitError(
             'There are package boundary violations in the following GN '
-                'files:', long_text='\n\n'.join(str(err) for err in errors))
+            'files:',
+            long_text='\n\n'.join(str(err) for err in errors))
     ]
   return []
 
@@ -492,7 +491,7 @@ def CheckNoWarningSuppressionFlagsAreAdded(gn_files,
                                            input_api,
                                            output_api,
                                            error_formatter=_ReportFileAndLine):
-    """Ensure warning suppression flags are not added wihtout a reason."""
+  """Ensure warning suppression flags are not added without a reason."""
   msg = ('Usage of //build/config/clang:extra_warnings is discouraged '
          'in WebRTC.\n'
          'If you are not adding this code (e.g. you are just moving '
@@ -502,8 +501,8 @@ def CheckNoWarningSuppressionFlagsAreAdded(gn_files,
          '\n'
          'Affected files:\n')
   errors = []  # 2-element tuples with (file, line number)
-    clang_warn_re = input_api.re.compile(
+  clang_warn_re = input_api.re.compile(r'//build/config/clang:extra_warnings')
-        r'//build/config/clang:extra_warnings')
+  # pylint: disable-next=fixme
   no_presubmit_re = input_api.re.compile(
       r'# no-presubmit-check TODO\(bugs\.webrtc\.org/\d+\)')
   for f in gn_files:
@@ -541,8 +540,7 @@ def CheckNoStreamUsageIsAdded(input_api,
                               source_file_filter,
                               error_formatter=_ReportFileAndLine):
   """Make sure that no more dependencies on stringstream are added."""
-    error_msg = (
+  error_msg = ('Usage of <sstream>, <istream> and <ostream> in WebRTC is '
-        'Usage of <sstream>, <istream> and <ostream> in WebRTC is '
                'deprecated.\n'
                'This includes the following types:\n'
                'std::istringstream, std::ostringstream, std::wistringstream, '
@@ -562,19 +560,18 @@ def CheckNoStreamUsageIsAdded(input_api,
                'Affected files:\n')
   errors = []  # 2-element tuples with (file, line number)
   include_re = input_api.re.compile(r'#include <(i|o|s)stream>')
-    usage_re = input_api.re.compile(
+  usage_re = input_api.re.compile(r'std::(w|i|o|io|wi|wo|wio)(string)*stream')
-        r'std::(w|i|o|io|wi|wo|wio)(string)*stream')
   no_presubmit_re = input_api.re.compile(
       r'// no-presubmit-check TODO\(webrtc:8982\)')
-    file_filter = lambda x: (input_api.FilterSourceFile(x) and
+  file_filter = lambda x: (input_api.FilterSourceFile(x) and source_file_filter(
-                             source_file_filter(x))
+      x))
 
   def _IsException(file_path):
     is_test = any(
-            file_path.endswith(x) for x in
+        file_path.endswith(x)
-            ['_test.cc', '_tests.cc', '_unittest.cc', '_unittests.cc'])
+        for x in ['_test.cc', '_tests.cc', '_unittest.cc', '_unittests.cc'])
-        return (file_path.startswith('examples')
+    return (file_path.startswith('examples') or file_path.startswith('test')
-                or file_path.startswith('test') or is_test)
+            or is_test)
 
   for f in input_api.AffectedSourceFiles(file_filter):
     # Usage of stringstream is allowed under examples/ and in tests.
@@ -608,21 +605,20 @@ def CheckPublicDepsIsNotUsed(gn_files, input_api, output_api):
         if not surpressed:
           result.append(
               output_api.PresubmitError(
-                            error_msg %
+                  error_msg % (affected_file.LocalPath(), line_number)))
-                            (affected_file.LocalPath(), line_number)))
   return result
 
 
 def CheckCheckIncludesIsNotUsed(gn_files, input_api, output_api):
   result = []
-    error_msg = (
+  error_msg = ('check_includes overrides are not allowed since it can cause '
-        'check_includes overrides are not allowed since it can cause '
                'incorrect dependencies to form. It effectively means that your '
                'module can include any .h file without depending on its '
                'corresponding target. There are some exceptional cases when '
                'this is allowed: if so, get approval from a .gn owner in the '
                'root OWNERS file.\n'
                'Used in: %s (line %d).')
+  # pylint: disable-next=fixme
   no_presubmit_re = input_api.re.compile(
       r'# no-presubmit-check TODO\(bugs\.webrtc\.org/\d+\)')
   for affected_file in gn_files:
@@ -630,8 +626,8 @@ def CheckCheckIncludesIsNotUsed(gn_files, input_api, output_api):
       if ('check_includes' in affected_line
           and not no_presubmit_re.search(affected_line)):
         result.append(
-                    output_api.PresubmitError(
+            output_api.PresubmitError(error_msg %
-                        error_msg % (affected_file.LocalPath(), line_number)))
+                                      (affected_file.LocalPath(), line_number)))
   return result
 
 
@@ -652,13 +648,10 @@ def CheckGnChanges(input_api, output_api):
     result.extend(CheckAbseilDependencies(input_api, gn_files, output_api))
     result.extend(
         CheckNoPackageBoundaryViolations(input_api, gn_files, output_api))
-        result.extend(CheckPublicDepsIsNotUsed(gn_files, input_api,
+    result.extend(CheckPublicDepsIsNotUsed(gn_files, input_api, output_api))
-                                               output_api))
+    result.extend(CheckCheckIncludesIsNotUsed(gn_files, input_api, output_api))
     result.extend(
-            CheckCheckIncludesIsNotUsed(gn_files, input_api, output_api))
+        CheckNoWarningSuppressionFlagsAreAdded(gn_files, input_api, output_api))
-        result.extend(
-            CheckNoWarningSuppressionFlagsAreAdded(gn_files, input_api,
-                                                   output_api))
   return result
 
 
@@ -668,8 +661,8 @@ def CheckGnGen(input_api, output_api):
   errors.
   """
   with _AddToPath(
-            input_api.os_path.join(input_api.PresubmitLocalPath(),
+      input_api.os_path.join(input_api.PresubmitLocalPath(), 'tools_webrtc',
-                                   'tools_webrtc', 'presubmit_checks_lib')):
+                             'presubmit_checks_lib')):
     from build_helpers import RunGnCheck
   errors = RunGnCheck(FindSrcDirPath(input_api.PresubmitLocalPath()))[:5]
   if errors:
@@ -694,8 +687,7 @@ def CheckUnwantedDependencies(input_api, output_api, source_file_filter):
   # roundabout construct to import checkdeps because this file is
   # eval-ed and thus doesn't have __file__.
   src_path = FindSrcDirPath(input_api.PresubmitLocalPath())
-    checkdeps_path = input_api.os_path.join(src_path, 'buildtools',
+  checkdeps_path = input_api.os_path.join(src_path, 'buildtools', 'checkdeps')
-                                            'checkdeps')
   if not os.path.exists(checkdeps_path):
     return [
         output_api.PresubmitError(
@@ -752,8 +744,7 @@ def CheckCommitMessageBugEntry(input_api, output_api):
   """Check that bug entries are well-formed in commit message."""
   bogus_bug_msg = (
       'Bogus Bug entry: %s. Please specify the issue tracker prefix and the '
-        'issue number, separated by a colon, e.g. webrtc:123 or chromium:12345.'
+      'issue number, separated by a colon, e.g. webrtc:123 or chromium:12345.')
-    )
   results = []
   for bug in input_api.change.BugsFromDescription():
     bug = bug.strip()
@@ -766,8 +757,7 @@ def CheckCommitMessageBugEntry(input_api, output_api):
           prefix_guess = 'chromium'
         else:
           prefix_guess = 'webrtc'
-                results.append(
+        results.append('Bug entry requires issue tracker prefix, e.g. %s:%s' %
-                    'Bug entry requires issue tracker prefix, e.g. %s:%s' %
                        (prefix_guess, bug))
       except ValueError:
         results.append(bogus_bug_msg % bug)
@@ -788,15 +778,13 @@ def CheckChangeHasBugField(input_api, output_api):
   """
   if input_api.change.BugsFromDescription():
     return []
-    else:
   return [
       output_api.PresubmitError(
           'The "Bug: [bug number]" footer is mandatory. Please create a '
           'bug and reference it using either of:\n'
           ' * https://bugs.webrtc.org - reference it using Bug: '
           'webrtc:XXXX\n'
-                ' * https://crbug.com - reference it using Bug: chromium:XXXXXX'
+          ' * https://crbug.com - reference it using Bug: chromium:XXXXXX')
-            )
   ]
 
 
@@ -804,8 +792,7 @@ def CheckJSONParseErrors(input_api, output_api, source_file_filter):
   """Check that JSON files do not contain syntax errors."""
 
   def FilterFile(affected_file):
-        return (input_api.os_path.splitext(
+    return (input_api.os_path.splitext(affected_file.LocalPath())[1] == '.json'
-            affected_file.LocalPath())[1] == '.json'
             and source_file_filter(affected_file))
 
   def GetJSONParseError(input_api, filename):
@@ -823,8 +810,7 @@ def CheckJSONParseErrors(input_api, output_api, source_file_filter):
                                     affected_file.AbsoluteLocalPath())
     if parse_error:
       results.append(
-                output_api.PresubmitError(
+          output_api.PresubmitError('%s could not be parsed: %s' %
-                    '%s could not be parsed: %s' %
                                     (affected_file.LocalPath(), parse_error)))
   return results
 
@@ -833,23 +819,32 @@ def RunPythonTests(input_api, output_api):
   def Join(*args):
     return input_api.os_path.join(input_api.PresubmitLocalPath(), *args)
 
+  excluded_files = [
+      # These tests should be run manually after webrtc_dashboard_upload target
+      # has been built.
+      'catapult_uploader_test.py',
+      'process_perf_results_test.py',
+  ]
+
   test_directories = [
       input_api.PresubmitLocalPath(),
       Join('rtc_tools', 'py_event_log_analyzer'),
       Join('audio', 'test', 'unittests'),
   ] + [
       root for root, _, files in os.walk(Join('tools_webrtc')) if any(
-            f.endswith('_test.py') for f in files)
+          f.endswith('_test.py') and f not in excluded_files for f in files)
   ]
 
   tests = []
+
   for directory in test_directories:
     tests.extend(
         input_api.canned_checks.GetUnitTestsInDirectory(
             input_api,
             output_api,
             directory,
-                files_to_check=[r'.+_test\.py$']))
+            files_to_check=[r'.+_test\.py$'],
+            run_on_python2=False))
   return input_api.RunTests(tests, parallel=True)
 
 
@@ -859,8 +854,8 @@ def CheckUsageOfGoogleProtobufNamespace(input_api, output_api,
   files = []
   pattern = input_api.re.compile(r'google::protobuf')
   proto_utils_path = os.path.join('rtc_base', 'protobuf_utils.h')
-    file_filter = lambda x: (input_api.FilterSourceFile(x) and
+  file_filter = lambda x: (input_api.FilterSourceFile(x) and source_file_filter(
-                             source_file_filter(x))
+      x))
   for f in input_api.AffectedSourceFiles(file_filter):
     if f.LocalPath() in [proto_utils_path, 'PRESUBMIT.py']:
       continue
@@ -872,8 +867,8 @@ def CheckUsageOfGoogleProtobufNamespace(input_api, output_api,
     return [
         output_api.PresubmitError(
             'Please avoid to use namespace `google::protobuf` directly.\n'
-                'Add a using directive in `%s` and include that header instead.'
+            'Add a using directive in `%s` and include that header instead.' %
-                % proto_utils_path, files)
+            proto_utils_path, files)
     ]
   return []
 
@@ -882,7 +877,7 @@ def _LicenseHeader(input_api):
   """Returns the license header regexp."""
   # Accept any year number from 2003 to the current year
   current_year = int(input_api.time.strftime('%Y'))
-    allowed_years = (str(s) for s in reversed(xrange(2003, current_year + 1)))
+  allowed_years = (str(s) for s in reversed(range(2003, current_year + 1)))
   years_re = '(' + '|'.join(allowed_years) + ')'
   license_header = (
       r'.*? Copyright( \(c\))? %(year)s The WebRTC [Pp]roject [Aa]uthors\. '
@@ -921,8 +916,11 @@ def CommonChecks(input_api, output_api):
   # all python files. This is a temporary solution.
   python_file_filter = lambda f: (f.LocalPath().endswith('.py') and
                                   source_file_filter(f))
-    python_changed_files = [f.LocalPath() for f in input_api.AffectedFiles(
+  python_changed_files = [
-        include_deletes=False, file_filter=python_file_filter)]
+      f.LocalPath()
+      for f in input_api.AffectedFiles(include_deletes=False,
+                                       file_filter=python_file_filter)
+  ]
 
   results.extend(
       input_api.canned_checks.RunPylint(
@@ -939,13 +937,14 @@ def CommonChecks(input_api, output_api):
               r'^testing[\\\/].*\.py$',
               r'^third_party[\\\/].*\.py$',
               r'^tools[\\\/].*\.py$',
-                # TODO(phoglund): should arguably be checked.
+              # TODO(bugs.webrtc.org/13605): should arguably be checked.
               r'^tools_webrtc[\\\/]mb[\\\/].*\.py$',
               r'^xcodebuild.*[\\\/].*\.py$',
           ),
-            pylintrc='pylintrc'))
+          pylintrc='pylintrc',
+          version='2.7'))
 
-    # TODO(nisse): talk/ is no more, so make below checks simpler?
+  # TODO(bugs.webrtc.org/13606): talk/ is no more, so make below checks simpler?
   # WebRTC can't use the presubmit_canned_checks.PanProjectChecks function
   # since we need to have different license checks
   # in talk/ and webrtc/directories.
@@ -1032,8 +1031,7 @@ def CommonChecks(input_api, output_api):
       CheckNewlineAtTheEndOfProtoFiles(
           input_api, output_api, source_file_filter=non_third_party_sources))
   results.extend(
-        CheckNoStreamUsageIsAdded(input_api, output_api,
+      CheckNoStreamUsageIsAdded(input_api, output_api, non_third_party_sources))
-                                  non_third_party_sources))
   results.extend(
       CheckNoTestCaseUsageIsAdded(input_api, output_api,
                                   non_third_party_sources))
@@ -1044,8 +1042,7 @@ def CommonChecks(input_api, output_api):
   results.extend(
       CheckAssertUsage(input_api, output_api, non_third_party_sources))
   results.extend(
-        CheckBannedAbslMakeUnique(input_api, output_api,
+      CheckBannedAbslMakeUnique(input_api, output_api, non_third_party_sources))
-                                  non_third_party_sources))
   results.extend(
       CheckObjcApiSymbols(input_api, output_api, non_third_party_sources))
   return results
@@ -1075,8 +1072,7 @@ def CheckApiDepsFileIsUpToDate(input_api, output_api):
     path_tokens = [t for t in f.LocalPath().split(os.sep) if t]
     if len(path_tokens) > 1:
       if (path_tokens[0] not in dirs_to_skip and os.path.isdir(
-                    os.path.join(input_api.PresubmitLocalPath(),
+          os.path.join(input_api.PresubmitLocalPath(), path_tokens[0]))):
-                                 path_tokens[0]))):
         dirs_to_check.add(path_tokens[0])
 
   missing_include_rules = set()
@@ -1119,7 +1115,7 @@ def CheckBannedAbslMakeUnique(input_api, output_api, source_file_filter):
         files.append(f)
         break
 
-    if len(files):
+  if files:
     return [
         output_api.PresubmitError(
             'Please use std::make_unique instead of absl::make_unique.\n'
@@ -1135,8 +1131,8 @@ def CheckObjcApiSymbols(input_api, output_api, source_file_filter):
                            source_file_filter(f))
 
   files = []
-    file_filter = lambda x: (input_api.FilterSourceFile(x) and
+  file_filter = lambda x: (input_api.FilterSourceFile(x) and source_file_filter(
-                             source_file_filter(x))
+      x))
   for f in input_api.AffectedSourceFiles(file_filter):
     if not f.LocalPath().endswith('.h') or not 'sdk/objc' in f.LocalPath():
       continue
@@ -1148,11 +1144,11 @@ def CheckObjcApiSymbols(input_api, output_api, source_file_filter):
       if 'RTC_OBJC_TYPE' not in export_block:
         files.append(f.LocalPath())
 
-    if len(files):
+  if len(files) > 0:
     return [
         output_api.PresubmitError(
-                'RTC_OBJC_EXPORT types must be wrapped into an RTC_OBJC_TYPE() '
+            'RTC_OBJC_EXPORT types must be wrapped into an RTC_OBJC_TYPE() ' +
-                + 'macro.\n\n' + 'For example:\n' +
+            'macro.\n\n' + 'For example:\n' +
             'RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE(RtcFoo)\n\n' +
             'RTC_OBJC_EXPORT @interface RTC_OBJC_TYPE(RtcFoo)\n\n' +
             'Please fix the following files:', files)
@@ -1173,7 +1169,7 @@ def CheckAssertUsage(input_api, output_api, source_file_filter):
         files.append(f.LocalPath())
         break
 
-    if len(files):
+  if len(files) > 0:
     return [
         output_api.PresubmitError(
             'Usage of assert() has been detected in the following files, '
@@ -1199,7 +1195,7 @@ def CheckAbslMemoryInclude(input_api, output_api, source_file_filter):
         files.append(f)
         break
 
-    if len(files):
+  if len(files) > 0:
     return [
         output_api.PresubmitError(
             'Please include "absl/memory/memory.h" header for '
@@ -1213,8 +1209,8 @@ def CheckChangeOnUpload(input_api, output_api):
   results = []
   results.extend(CommonChecks(input_api, output_api))
   results.extend(CheckGnGen(input_api, output_api))
-    results.extend(
+  results.extend(input_api.canned_checks.CheckGNFormatted(
-        input_api.canned_checks.CheckGNFormatted(input_api, output_api))
+      input_api, output_api))
   return results
 
 
@@ -1226,8 +1222,7 @@ def CheckChangeOnCommit(input_api, output_api):
   results.extend(
       input_api.canned_checks.CheckChangeWasUploaded(input_api, output_api))
   results.extend(
-        input_api.canned_checks.CheckChangeHasDescription(
+      input_api.canned_checks.CheckChangeHasDescription(input_api, output_api))
-            input_api, output_api))
   results.extend(CheckChangeHasBugField(input_api, output_api))
   results.extend(CheckCommitMessageBugEntry(input_api, output_api))
   results.extend(
@@ -1244,12 +1239,11 @@ def CheckOrphanHeaders(input_api, output_api, source_file_filter):
   # eval-ed and thus doesn't have __file__.
   error_msg = """{} should be listed in {}."""
   results = []
-    exempt_paths = [
+  exempt_paths = [re.escape(os.path.join('tools_webrtc', 'ios', 'SDK'))]
-        os.path.join('tools_webrtc', 'ios', 'SDK'),
+
-    ]
   with _AddToPath(
-            input_api.os_path.join(input_api.PresubmitLocalPath(),
+      input_api.os_path.join(input_api.PresubmitLocalPath(), 'tools_webrtc',
-                                   'tools_webrtc', 'presubmit_checks_lib')):
+                             'presubmit_checks_lib')):
     from check_orphan_headers import GetBuildGnPathFromFilePath
     from check_orphan_headers import IsHeaderInBuildGn
 
@@ -1259,14 +1253,13 @@ def CheckOrphanHeaders(input_api, output_api, source_file_filter):
     if f.LocalPath().endswith('.h'):
       file_path = os.path.abspath(f.LocalPath())
       root_dir = os.getcwd()
-            gn_file_path = GetBuildGnPathFromFilePath(file_path,
+      gn_file_path = GetBuildGnPathFromFilePath(file_path, os.path.exists,
-                                                      os.path.exists, root_dir)
+                                                root_dir)
       in_build_gn = IsHeaderInBuildGn(file_path, gn_file_path)
       if not in_build_gn:
         results.append(
             output_api.PresubmitError(
-                        error_msg.format(f.LocalPath(),
+                error_msg.format(f.LocalPath(), os.path.relpath(gn_file_path))))
-                                         os.path.relpath(gn_file_path))))
   return results
 
 
@@ -1282,8 +1275,7 @@ def CheckNewlineAtTheEndOfProtoFiles(input_api, output_api,
     with open(file_path) as f:
       lines = f.readlines()
       if len(lines) > 0 and not lines[-1].endswith('\n'):
-                results.append(
+        results.append(output_api.PresubmitError(error_msg.format(file_path)))
-                    output_api.PresubmitError(error_msg.format(file_path)))
   return results
 
 
@@ -1297,7 +1289,7 @@ def _ExtractAddRulesFromParsedDeps(parsed_deps):
       rule[1:] for rule in parsed_deps.get('include_rules', [])
       if rule.startswith('+') or rule.startswith('!')
   ])
-    for _, rules in parsed_deps.get('specific_include_rules', {}).iteritems():
+  for _, rules in parsed_deps.get('specific_include_rules', {}).items():
     add_rules.update([
         rule[1:] for rule in rules
         if rule.startswith('+') or rule.startswith('!')
@@ -1309,7 +1301,7 @@ def _ParseDeps(contents):
   """Simple helper for parsing DEPS files."""
 
   # Stubs for handling special syntax in the root DEPS file.
-    class VarImpl(object):
+  class VarImpl:
     def __init__(self, local_scope):
       self._local_scope = local_scope
 
@@ -1317,14 +1309,15 @@ def _ParseDeps(contents):
       """Implements the Var syntax."""
       try:
         return self._local_scope['vars'][var_name]
-            except KeyError:
+      except KeyError as var_not_defined:
-                raise Exception('Var is not defined: %s' % var_name)
+        raise Exception('Var is not defined: %s' %
+                        var_name) from var_not_defined
 
   local_scope = {}
   global_scope = {
       'Var': VarImpl(local_scope).Lookup,
   }
-    exec contents in global_scope, local_scope
+  exec(contents, global_scope, local_scope)
   return local_scope
 
 
@@ -1371,8 +1364,7 @@ def CheckAddedDepsHaveTargetApprovals(input_api, output_api):
     filename = input_api.os_path.basename(f.LocalPath())
     if filename == 'DEPS':
       virtual_depended_on_files.update(
-                _CalculateAddedDeps(input_api.os_path,
+          _CalculateAddedDeps(input_api.os_path, '\n'.join(f.OldContents()),
-                                    '\n'.join(f.OldContents()),
                               '\n'.join(f.NewContents())))
 
   if not virtual_depended_on_files:
@@ -1383,15 +1375,13 @@ def CheckAddedDepsHaveTargetApprovals(input_api, output_api):
       return [
           output_api.PresubmitNotifyResult(
               '--tbr was specified, skipping OWNERS check for DEPS '
-                    'additions'
+              'additions')
-                )
       ]
     if input_api.dry_run:
       return [
           output_api.PresubmitNotifyResult(
               'This is a dry run, skipping OWNERS check for DEPS '
-                    'additions'
+              'additions')
-                )
       ]
     if not input_api.change.issue:
       return [
@@ -1405,9 +1395,7 @@ def CheckAddedDepsHaveTargetApprovals(input_api, output_api):
 
   owner_email, reviewers = (
       input_api.canned_checks.GetCodereviewOwnerAndReviewers(
-            input_api,
+          input_api, None, approval_needed=input_api.is_committing))
-            None,
-            approval_needed=input_api.is_committing))
 
   owner_email = owner_email or input_api.change.author_email
 
@@ -1415,7 +1403,8 @@ def CheckAddedDepsHaveTargetApprovals(input_api, output_api):
       virtual_depended_on_files, reviewers.union([owner_email]), [])
   missing_files = [
       f for f in virtual_depended_on_files
-        if approval_status[f] != input_api.owners_client.APPROVED]
+      if approval_status[f] != input_api.owners_client.APPROVED
+  ]
 
   # We strip the /DEPS part that was added by
   # _FilesToCheckForIncomingDeps to fake a path to a file in a
@@ -1424,7 +1413,6 @@ def CheckAddedDepsHaveTargetApprovals(input_api, output_api):
     start_deps = path.rfind('/DEPS')
     if start_deps != -1:
       return path[:start_deps]
-        else:
     return path
 
   unapproved_dependencies = [
@@ -1433,8 +1421,7 @@ def CheckAddedDepsHaveTargetApprovals(input_api, output_api):
 
   if unapproved_dependencies:
     output_list = [
-            output(
+        output('You need LGTM from owners of depends-on paths in DEPS that '
-                'You need LGTM from owners of depends-on paths in DEPS that '
                ' were modified in this CL:\n    %s' %
                '\n    '.join(sorted(unapproved_dependencies)))
     ]

api/BUILD.gn

@@ -98,6 +98,12 @@ rtc_library("rtp_packet_info") {
   absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
+rtc_source_set("video_track_source_constraints") {
+  visibility = [ "*" ]
+  sources = [ "video_track_source_constraints.h" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
 rtc_library("media_stream_interface") {
   visibility = [ "*" ]
   sources = [
@@ -110,9 +116,12 @@ rtc_library("media_stream_interface") {
     ":audio_options_api",
     ":rtp_parameters",
     ":scoped_refptr",
+    ":sequence_checker",
+    ":video_track_source_constraints",
     "../modules/audio_processing:audio_processing_statistics",
     "../rtc_base:checks",
     "../rtc_base:refcount",
+    "../rtc_base/system:no_unique_address",
     "../rtc_base/system:rtc_export",
     "video:recordable_encoded_frame",
     "video:video_frame",
@@ -185,6 +194,7 @@ rtc_library("libjingle_peerconnection_api") {
     "crypto:frame_decryptor_interface",
     "crypto:frame_encryptor_interface",
     "crypto:options",
+    "metronome",
     "neteq:neteq_api",
     "rtc_event_log",
     "task_queue",
@@ -257,6 +267,7 @@ rtc_source_set("packet_socket_factory") {
   ]
   deps = [
     ":async_dns_resolver",
+    ":wrapping_async_dns_resolver",
     "../rtc_base:async_resolver_interface",
     "../rtc_base:rtc_base",
     "../rtc_base:socket_address",
@@ -273,6 +284,28 @@ rtc_source_set("async_dns_resolver") {
   ]
 }
 
+rtc_source_set("wrapping_async_dns_resolver") {
+  visibility = [
+    ":*",
+    "../p2p:rtc_p2p",
+  ]
+  sources = [
+    "wrapping_async_dns_resolver.cc",
+    "wrapping_async_dns_resolver.h",
+  ]
+  deps = [
+    ":async_dns_resolver",
+    ":sequence_checker",
+    "../rtc_base:async_resolver_interface",
+    "../rtc_base:checks",
+    "../rtc_base:macromagic",
+    "../rtc_base:socket_address",
+    "../rtc_base:threading",
+    "../rtc_base/third_party/sigslot",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+}
+
 rtc_source_set("scoped_refptr") {
   visibility = [ "*" ]
   sources = [ "scoped_refptr.h" ]
@@ -382,6 +415,16 @@ rtc_source_set("stats_observer_interface") {
   absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
 }
 
+rtc_source_set("peer_network_dependencies") {
+  visibility = [ "*" ]
+  sources = [ "test/peer_network_dependencies.h" ]
+  deps = [
+    ":packet_socket_factory",
+    "../rtc_base",
+    "../rtc_base:threading",
+  ]
+}
+
 rtc_source_set("peer_connection_quality_test_fixture_api") {
   visibility = [ "*" ]
   testonly = true
@@ -397,14 +440,17 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
     ":media_stream_interface",
     ":network_state_predictor_api",
     ":packet_socket_factory",
+    ":peer_network_dependencies",
     ":rtp_parameters",
     ":simulated_network_api",
     ":stats_observer_interface",
     ":track_id_stream_info_map",
     ":video_quality_analyzer_api",
     "../media:rtc_media_base",
+    "../modules/audio_processing:api",
     "../rtc_base:rtc_base",
     "../rtc_base:threading",
+    "audio:audio_mixer_api",
     "rtc_event_log",
     "task_queue",
     "transport:network_control",
@@ -625,6 +671,8 @@ rtc_source_set("network_emulation_manager_api") {
   ]
   deps = [
     ":array_view",
+    ":packet_socket_factory",
+    ":peer_network_dependencies",
     ":simulated_network_api",
     ":time_controller",
     "../call:simulated_network",
@@ -1177,3 +1225,10 @@ if (rtc_include_tests) {
     ]
   }
 }
+
+rtc_source_set("webrtc_key_value_config") {
+  visibility = [ "*" ]
+  sources = [ "webrtc_key_value_config.h" ]
+  deps = [ "../rtc_base/system:rtc_export" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}

api/DEPS

@@ -112,14 +112,6 @@ specific_include_rules = {
     "+rtc_base/ref_count.h",
   ],
 
-  "jsep_ice_candidate\.h": [
-    "+rtc_base/constructor_magic.h",
-  ],
-
-  "jsep_session_description\.h": [
-    "+rtc_base/constructor_magic.h",
-  ],
-
   "media_stream_interface\.h": [
     "+modules/audio_processing/include/audio_processing_statistics.h",
     "+rtc_base/ref_count.h",
@@ -156,7 +148,6 @@ specific_include_rules = {
   ],
 
   "ref_counted_base\.h": [
-    "+rtc_base/constructor_magic.h",
     "+rtc_base/ref_count.h",
     "+rtc_base/ref_counter.h",
   ],
@@ -168,6 +159,9 @@ specific_include_rules = {
     # For private member and constructor.
     "+rtc_base/system/file_wrapper.h",
   ],
+  "rtp_packet_infos\.h": [
+    "+rtc_base/ref_counted_object.h",
+  ],
   "rtp_receiver_interface\.h": [
     "+rtc_base/ref_count.h",
   ],
@@ -192,7 +186,6 @@ specific_include_rules = {
   ],
 
   "stats_types\.h": [
-    "+rtc_base/constructor_magic.h",
     "+rtc_base/ref_count.h",
     "+rtc_base/thread_checker.h",
   ],
@@ -201,17 +194,12 @@ specific_include_rules = {
     "+rtc_base/ref_count.h",
   ],
 
-  "audio_frame\.h": [
-    "+rtc_base/constructor_magic.h",
-  ],
-
   "audio_mixer\.h": [
     "+rtc_base/ref_count.h",
   ],
 
   "audio_decoder\.h": [
     "+rtc_base/buffer.h",
-    "+rtc_base/constructor_magic.h",
   ],
 
   "audio_decoder_factory\.h": [
@@ -267,6 +255,13 @@ specific_include_rules = {
     "+rtc_base/ref_counted_object.h",
   ],
 
+  "fake_metronome\.h": [
+    "+rtc_base/synchronization/mutex.h",
+    "+rtc_base/task_queue.h",
+    "+rtc_base/task_utils/repeating_task.h",
+    "+rtc_base/thread_annotations.h",
+  ],
+
   "mock.*\.h": [
     "+test/gmock.h",
   ],
@@ -279,6 +274,10 @@ specific_include_rules = {
     "+rtc_base/ref_counted_object.h",
   ],
 
+  "notifier\.h": [
+    "+rtc_base/system/no_unique_address.h",
+  ],
+
   "simulated_network\.h": [
     "+rtc_base/random.h",
     "+rtc_base/thread_annotations.h",
@@ -305,6 +304,18 @@ specific_include_rules = {
     "+rtc_base/thread_annotations.h",
   ],
 
+  "wrapping_async_dns_resolver\.h": [
+    "+rtc_base/async_resolver.h",
+    "+rtc_base/async_resolver_interface.h",
+    "+rtc_base/socket_address.h",
+    "+rtc_base/third_party/sigslot/sigslot.h",
+    "+rtc_base/thread_annotations.h",
+  ],
+
+  "video_encoder_factory_template.*\.h": [
+    "+modules/video_coding",
+  ],
+
   # .cc files in api/ should not be restricted in what they can #include,
   # so we re-add all the top-level directories here. (That's because .h
   # files leak their #includes to whoever's #including them, but .cc files

api/async_dns_resolver.h

@@ -51,6 +51,11 @@ class AsyncDnsResolverResult {
   virtual int GetError() const = 0;
 };
 
+// The API for a single name query.
+// The constructor, destructor and all functions must be called from
+// the same sequence, and the callback will also be called on that sequence.
+// The class guarantees that the callback will not be called if the
+// resolver's destructor has been called.
 class RTC_EXPORT AsyncDnsResolverInterface {
  public:
   virtual ~AsyncDnsResolverInterface() = default;
@@ -70,7 +75,7 @@ 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 thread that the caller runs on.
+  // The callback will be called on the sequence that the caller runs on.
   virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface> CreateAndResolve(
       const rtc::SocketAddress& addr,
       std::function<void()> callback) = 0;

api/audio/BUILD.gn

@@ -95,6 +95,7 @@ rtc_source_set("echo_control") {
 
 rtc_source_set("echo_detector_creator") {
   visibility = [ "*" ]
+  allow_poison = [ "default_echo_detector" ]
   sources = [
     "echo_detector_creator.cc",
     "echo_detector_creator.h",
@@ -102,7 +103,7 @@ rtc_source_set("echo_detector_creator") {
   deps = [
     "../../api:scoped_refptr",
     "../../modules/audio_processing:api",
-    "../../modules/audio_processing:audio_processing",
+    "../../modules/audio_processing:residual_echo_detector",
     "../../rtc_base:refcount",
   ]
 }

api/audio/audio_frame.cc

@@ -11,8 +11,6 @@
 #include "api/audio/audio_frame.h"
 
 #include <string.h>
-#include <algorithm>
-#include <utility>
 
 #include "rtc_base/checks.h"
 #include "rtc_base/time_utils.h"
@@ -24,28 +22,6 @@ AudioFrame::AudioFrame() {
   static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
 }
 
-void swap(AudioFrame& a, AudioFrame& b) {
-  using std::swap;
-  swap(a.timestamp_, b.timestamp_);
-  swap(a.elapsed_time_ms_, b.elapsed_time_ms_);
-  swap(a.ntp_time_ms_, b.ntp_time_ms_);
-  swap(a.samples_per_channel_, b.samples_per_channel_);
-  swap(a.sample_rate_hz_, b.sample_rate_hz_);
-  swap(a.num_channels_, b.num_channels_);
-  swap(a.channel_layout_, b.channel_layout_);
-  swap(a.speech_type_, b.speech_type_);
-  swap(a.vad_activity_, b.vad_activity_);
-  swap(a.profile_timestamp_ms_, b.profile_timestamp_ms_);
-  swap(a.packet_infos_, b.packet_infos_);
-  const size_t length_a = a.samples_per_channel_ * a.num_channels_;
-  const size_t length_b = b.samples_per_channel_ * b.num_channels_;
-  RTC_DCHECK_LE(length_a, AudioFrame::kMaxDataSizeSamples);
-  RTC_DCHECK_LE(length_b, AudioFrame::kMaxDataSizeSamples);
-  std::swap_ranges(a.data_, a.data_ + std::max(length_a, length_b), b.data_);
-  swap(a.muted_, b.muted_);
-  swap(a.absolute_capture_timestamp_ms_, b.absolute_capture_timestamp_ms_);
-}
-
 void AudioFrame::Reset() {
   ResetWithoutMuting();
   muted_ = true;

api/audio/audio_frame.h

@@ -14,11 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include <utility>
-
 #include "api/audio/channel_layout.h"
 #include "api/rtp_packet_infos.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -60,7 +57,8 @@ class AudioFrame {
 
   AudioFrame();
 
-  friend void swap(AudioFrame& a, AudioFrame& b);
+  AudioFrame(const AudioFrame&) = delete;
+  AudioFrame& operator=(const AudioFrame&) = delete;
 
   // Resets all members to their default state.
   void Reset();
@@ -168,8 +166,6 @@ class AudioFrame {
   // capture timestamp of a received frame is found in `packet_infos_`.
   // This timestamp MUST be based on the same clock as rtc::TimeMillis().
   absl::optional<int64_t> absolute_capture_timestamp_ms_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
 };
 
 }  // namespace webrtc

api/audio/channel_layout.cc

@@ -275,7 +275,7 @@ const char* ChannelLayoutToString(ChannelLayout layout) {
     case CHANNEL_LAYOUT_BITSTREAM:
       return "BITSTREAM";
   }
-  RTC_NOTREACHED() << "Invalid channel layout provided: " << layout;
+  RTC_DCHECK_NOTREACHED() << "Invalid channel layout provided: " << layout;
   return "";
 }
 

api/audio/echo_canceller3_config.h

@@ -112,7 +112,7 @@ struct RTC_EXPORT EchoCanceller3Config {
     bool echo_can_saturate = true;
     bool bounded_erl = false;
     bool erle_onset_compensation_in_dominant_nearend = false;
-    bool use_conservative_tail_frequency_response = false;
+    bool use_conservative_tail_frequency_response = true;
   } ep_strength;
 
   struct EchoAudibility {
@@ -209,6 +209,7 @@ struct RTC_EXPORT EchoCanceller3Config {
       int hold_duration = 50;
       int trigger_threshold = 12;
       bool use_during_initial_phase = true;
+      bool use_unbounded_echo_spectrum = true;
     } dominant_nearend_detection;
 
     struct SubbandNearendDetection {

api/audio/echo_canceller3_config_json.cc

@@ -374,6 +374,9 @@ void Aec3ConfigFromJsonString(absl::string_view json_string,
       ReadParam(
           subsection, "use_during_initial_phase",
           &cfg.suppressor.dominant_nearend_detection.use_during_initial_phase);
+      ReadParam(subsection, "use_unbounded_echo_spectrum",
+                &cfg.suppressor.dominant_nearend_detection
+                     .use_unbounded_echo_spectrum);
     }
 
     if (rtc::GetValueFromJsonObject(section, "subband_nearend_detection",
@@ -684,20 +687,20 @@ std::string Aec3ConfigToJsonString(const EchoCanceller3Config& config) {
       << config.suppressor.last_lf_smoothing_band << ",";
   ost << "\"last_lf_band\": " << config.suppressor.last_lf_band << ",";
   ost << "\"first_hf_band\": " << config.suppressor.first_hf_band << ",";
+  {
+    const auto& dnd = config.suppressor.dominant_nearend_detection;
     ost << "\"dominant_nearend_detection\": {";
-  ost << "\"enr_threshold\": "
+    ost << "\"enr_threshold\": " << dnd.enr_threshold << ",";
-      << config.suppressor.dominant_nearend_detection.enr_threshold << ",";
+    ost << "\"enr_exit_threshold\": " << dnd.enr_exit_threshold << ",";
-  ost << "\"enr_exit_threshold\": "
+    ost << "\"snr_threshold\": " << dnd.snr_threshold << ",";
-      << config.suppressor.dominant_nearend_detection.enr_exit_threshold << ",";
+    ost << "\"hold_duration\": " << dnd.hold_duration << ",";
-  ost << "\"snr_threshold\": "
+    ost << "\"trigger_threshold\": " << dnd.trigger_threshold << ",";
-      << config.suppressor.dominant_nearend_detection.snr_threshold << ",";
+    ost << "\"use_during_initial_phase\": " << dnd.use_during_initial_phase
-  ost << "\"hold_duration\": "
+        << ",";
-      << config.suppressor.dominant_nearend_detection.hold_duration << ",";
+    ost << "\"use_unbounded_echo_spectrum\": "
-  ost << "\"trigger_threshold\": "
+        << dnd.use_unbounded_echo_spectrum;
-      << config.suppressor.dominant_nearend_detection.trigger_threshold << ",";
-  ost << "\"use_during_initial_phase\": "
-      << config.suppressor.dominant_nearend_detection.use_during_initial_phase;
     ost << "},";
+  }
   ost << "\"subband_nearend_detection\": {";
   ost << "\"nearend_average_blocks\": "
       << config.suppressor.subband_nearend_detection.nearend_average_blocks

api/audio/test/audio_frame_unittest.cc

@@ -133,54 +133,4 @@ TEST(AudioFrameTest, CopyFrom) {
   EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
 }
 
-TEST(AudioFrameTest, SwapFrames) {
-  AudioFrame frame1, frame2;
-  int16_t samples1[kNumChannelsMono * kSamplesPerChannel];
-  for (size_t i = 0; i < kNumChannelsMono * kSamplesPerChannel; ++i) {
-    samples1[i] = i;
-  }
-  frame1.UpdateFrame(kTimestamp, samples1, kSamplesPerChannel, kSampleRateHz,
-                     AudioFrame::kPLC, AudioFrame::kVadActive,
-                     kNumChannelsMono);
-  frame1.set_absolute_capture_timestamp_ms(12345678);
-  const auto frame1_channel_layout = frame1.channel_layout();
-
-  int16_t samples2[(kNumChannelsMono + 1) * (kSamplesPerChannel + 1)];
-  for (size_t i = 0; i < (kNumChannelsMono + 1) * (kSamplesPerChannel + 1);
-       ++i) {
-    samples2[i] = 1000 + i;
-  }
-  frame2.UpdateFrame(kTimestamp + 1, samples2, kSamplesPerChannel + 1,
-                     kSampleRateHz + 1, AudioFrame::kNormalSpeech,
-                     AudioFrame::kVadPassive, kNumChannelsMono + 1);
-  const auto frame2_channel_layout = frame2.channel_layout();
-
-  swap(frame1, frame2);
-
-  EXPECT_EQ(kTimestamp + 1, frame1.timestamp_);
-  ASSERT_EQ(kSamplesPerChannel + 1, frame1.samples_per_channel_);
-  EXPECT_EQ(kSampleRateHz + 1, frame1.sample_rate_hz_);
-  EXPECT_EQ(AudioFrame::kNormalSpeech, frame1.speech_type_);
-  EXPECT_EQ(AudioFrame::kVadPassive, frame1.vad_activity_);
-  ASSERT_EQ(kNumChannelsMono + 1, frame1.num_channels_);
-  for (size_t i = 0; i < (kNumChannelsMono + 1) * (kSamplesPerChannel + 1);
-       ++i) {
-    EXPECT_EQ(samples2[i], frame1.data()[i]);
-  }
-  EXPECT_FALSE(frame1.absolute_capture_timestamp_ms());
-  EXPECT_EQ(frame2_channel_layout, frame1.channel_layout());
-
-  EXPECT_EQ(kTimestamp, frame2.timestamp_);
-  ASSERT_EQ(kSamplesPerChannel, frame2.samples_per_channel_);
-  EXPECT_EQ(kSampleRateHz, frame2.sample_rate_hz_);
-  EXPECT_EQ(AudioFrame::kPLC, frame2.speech_type_);
-  EXPECT_EQ(AudioFrame::kVadActive, frame2.vad_activity_);
-  ASSERT_EQ(kNumChannelsMono, frame2.num_channels_);
-  for (size_t i = 0; i < kNumChannelsMono * kSamplesPerChannel; ++i) {
-    EXPECT_EQ(samples1[i], frame2.data()[i]);
-  }
-  EXPECT_EQ(12345678, frame2.absolute_capture_timestamp_ms());
-  EXPECT_EQ(frame1_channel_layout, frame2.channel_layout());
-}
-
 }  // namespace webrtc

api/audio_codecs/BUILD.gn

@@ -32,6 +32,7 @@ rtc_library("audio_codecs_api") {
     "..:array_view",
     "..:bitrate_allocation",
     "..:scoped_refptr",
+    "../../api:webrtc_key_value_config",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:sanitizer",

api/audio_codecs/L16/BUILD.gn

@@ -21,6 +21,7 @@ rtc_library("audio_encoder_L16") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:pcm16b",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
@@ -41,6 +42,7 @@ rtc_library("audio_decoder_L16") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:pcm16b",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",

api/audio_codecs/L16/audio_decoder_L16.cc

@@ -24,9 +24,10 @@ absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
   Config config;
   config.sample_rate_hz = format.clockrate_hz;
   config.num_channels = rtc::checked_cast<int>(format.num_channels);
-  return absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()
+  if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
-             ? absl::optional<Config>(config)
+    return config;
-             : absl::nullopt;
+  }
+  return absl::nullopt;
 }
 
 void AudioDecoderL16::AppendSupportedDecoders(
@@ -37,10 +38,13 @@ void AudioDecoderL16::AppendSupportedDecoders(
 
 std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
     const Config& config,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
-  return config.IsOk() ? std::make_unique<AudioDecoderPcm16B>(
+    const WebRtcKeyValueConfig* field_trials) {
-                             config.sample_rate_hz, config.num_channels)
+  if (!config.IsOk()) {
-                       : nullptr;
+    return nullptr;
+  }
+  return std::make_unique<AudioDecoderPcm16B>(config.sample_rate_hz,
+                                              config.num_channels);
 }
 
 }  // namespace webrtc

api/audio_codecs/L16/audio_decoder_L16.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -29,7 +30,8 @@ struct RTC_EXPORT AudioDecoderL16 {
     bool IsOk() const {
       return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
               sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
-             num_channels >= 1;
+             (num_channels >= 1 &&
+              num_channels <= AudioDecoder::kMaxNumberOfChannels);
     }
     int sample_rate_hz = 8000;
     int num_channels = 1;
@@ -38,7 +40,8 @@ struct RTC_EXPORT AudioDecoderL16 {
   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       const Config& config,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/L16/audio_encoder_L16.cc

@@ -24,6 +24,7 @@ namespace webrtc {
 absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
     const SdpAudioFormat& format) {
   if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
+    RTC_DCHECK_NOTREACHED();
     return absl::nullopt;
   }
   Config config;
@@ -36,9 +37,10 @@ absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
       config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
     }
   }
-  return absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()
+  if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
-             ? absl::optional<Config>(config)
+    return config;
-             : absl::nullopt;
+  }
+  return absl::nullopt;
 }
 
 void AudioEncoderL16::AppendSupportedEncoders(
@@ -58,13 +60,17 @@ AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
 std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
     const AudioEncoderL16::Config& config,
     int payload_type,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
-  RTC_DCHECK(config.IsOk());
+    const WebRtcKeyValueConfig* field_trials) {
   AudioEncoderPcm16B::Config c;
   c.sample_rate_hz = config.sample_rate_hz;
   c.num_channels = config.num_channels;
   c.frame_size_ms = config.frame_size_ms;
   c.payload_type = payload_type;
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
   return std::make_unique<AudioEncoderPcm16B>(c);
 }
 

api/audio_codecs/L16/audio_encoder_L16.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -29,7 +30,9 @@ struct RTC_EXPORT AudioEncoderL16 {
     bool IsOk() const {
       return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
               sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
-             num_channels >= 1 && frame_size_ms > 0 && frame_size_ms <= 120 &&
+             num_channels >= 1 &&
+             num_channels <= AudioEncoder::kMaxNumberOfChannels &&
+             frame_size_ms > 0 && frame_size_ms <= 120 &&
              frame_size_ms % 10 == 0;
     }
     int sample_rate_hz = 8000;
@@ -42,7 +45,8 @@ struct RTC_EXPORT AudioEncoderL16 {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       const Config& config,
       int payload_type,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/audio_decoder.cc

@@ -161,9 +161,10 @@ AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
     case 2:
       return kComfortNoise;
     default:
-      RTC_NOTREACHED();
+      RTC_DCHECK_NOTREACHED();
       return kSpeech;
   }
 }
 
+constexpr int AudioDecoder::kMaxNumberOfChannels;
 }  // namespace webrtc

api/audio_codecs/audio_decoder.h

@@ -20,7 +20,6 @@
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -37,6 +36,9 @@ class AudioDecoder {
   AudioDecoder() = default;
   virtual ~AudioDecoder() = default;
 
+  AudioDecoder(const AudioDecoder&) = delete;
+  AudioDecoder& operator=(const AudioDecoder&) = delete;
+
   class EncodedAudioFrame {
    public:
     struct DecodeResult {
@@ -170,6 +172,9 @@ class AudioDecoder {
   // during the lifetime of the decoder.
   virtual size_t Channels() const = 0;
 
+  // The maximum number of audio channels supported by WebRTC decoders.
+  static constexpr int kMaxNumberOfChannels = 24;
+
  protected:
   static SpeechType ConvertSpeechType(int16_t type);
 
@@ -184,9 +189,6 @@ class AudioDecoder {
                                       int sample_rate_hz,
                                       int16_t* decoded,
                                       SpeechType* speech_type);
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioDecoder);
 };
 
 }  // namespace webrtc

api/audio_codecs/audio_decoder_factory_template.h

@@ -16,6 +16,7 @@
 
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/scoped_refptr.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/ref_counted_object.h"
 
 namespace webrtc {
@@ -32,7 +33,8 @@ struct Helper<> {
   static bool IsSupportedDecoder(const SdpAudioFormat& format) { return false; }
   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       const SdpAudioFormat& format,
-      absl::optional<AudioCodecPairId> codec_pair_id) {
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const WebRtcKeyValueConfig* field_trials) {
     return nullptr;
   }
 };
@@ -55,16 +57,22 @@ struct Helper<T, Ts...> {
   }
   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       const SdpAudioFormat& format,
-      absl::optional<AudioCodecPairId> codec_pair_id) {
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const WebRtcKeyValueConfig* field_trials) {
     auto opt_config = T::SdpToConfig(format);
     return opt_config ? T::MakeAudioDecoder(*opt_config, codec_pair_id)
-                      : Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id);
+                      : Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id,
+                                                        field_trials);
   }
 };
 
 template <typename... Ts>
 class AudioDecoderFactoryT : public AudioDecoderFactory {
  public:
+  explicit AudioDecoderFactoryT(const WebRtcKeyValueConfig* field_trials) {
+    field_trials_ = field_trials;
+  }
+
   std::vector<AudioCodecSpec> GetSupportedDecoders() override {
     std::vector<AudioCodecSpec> specs;
     Helper<Ts...>::AppendSupportedDecoders(&specs);
@@ -78,8 +86,11 @@ class AudioDecoderFactoryT : public AudioDecoderFactory {
   std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       const SdpAudioFormat& format,
       absl::optional<AudioCodecPairId> codec_pair_id) override {
-    return Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id);
+    return Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id,
+                                           field_trials_);
   }
+
+  const WebRtcKeyValueConfig* field_trials_;
 };
 
 }  // namespace audio_decoder_factory_template_impl
@@ -115,7 +126,8 @@ class AudioDecoderFactoryT : public AudioDecoderFactory {
 // TODO(kwiberg): Point at CreateBuiltinAudioDecoderFactory() for an example of
 // how it is used.
 template <typename... Ts>
-rtc::scoped_refptr<AudioDecoderFactory> CreateAudioDecoderFactory() {
+rtc::scoped_refptr<AudioDecoderFactory> CreateAudioDecoderFactory(
+    const WebRtcKeyValueConfig* field_trials = nullptr) {
   // There's no technical reason we couldn't allow zero template parameters,
   // but such a factory couldn't create any decoders, and callers can do this
   // by mistake by simply forgetting the <> altogether. So we forbid it in
@@ -124,7 +136,8 @@ rtc::scoped_refptr<AudioDecoderFactory> CreateAudioDecoderFactory() {
                 "Caller must give at least one template parameter");
 
   return rtc::make_ref_counted<
-      audio_decoder_factory_template_impl::AudioDecoderFactoryT<Ts...>>();
+      audio_decoder_factory_template_impl::AudioDecoderFactoryT<Ts...>>(
+      field_trials);
 }
 
 }  // namespace webrtc

api/audio_codecs/audio_encoder.cc

@@ -83,7 +83,7 @@ void AudioEncoder::OnReceivedUplinkPacketLossFraction(
 
 void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
     float uplink_recoverable_packet_loss_fraction) {
-  RTC_NOTREACHED();
+  RTC_DCHECK_NOTREACHED();
 }
 
 void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
@@ -110,4 +110,5 @@ ANAStats AudioEncoder::GetANAStats() const {
   return ANAStats();
 }
 
+constexpr int AudioEncoder::kMaxNumberOfChannels;
 }  // namespace webrtc

api/audio_codecs/audio_encoder.h

@@ -276,6 +276,9 @@ class AudioEncoder {
   virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
       const = 0;
 
+  // The maximum number of audio channels supported by WebRTC encoders.
+  static constexpr int kMaxNumberOfChannels = 24;
+
   //  RingRTC Change to configure OPUS
   virtual bool Configure(const Config& config) {
     RTC_LOG(LS_WARNING) << "Default AudioEncoder::Configure(...) does nothing!";

api/audio_codecs/audio_encoder_factory_template.h

@@ -16,6 +16,7 @@
 
 #include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/scoped_refptr.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/ref_counted_object.h"
 
 namespace webrtc {
@@ -36,7 +37,8 @@ struct Helper<> {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       int payload_type,
       const SdpAudioFormat& format,
-      absl::optional<AudioCodecPairId> codec_pair_id) {
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const WebRtcKeyValueConfig* field_trials) {
     return nullptr;
   }
 };
@@ -63,13 +65,14 @@ struct Helper<T, Ts...> {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       int payload_type,
       const SdpAudioFormat& format,
-      absl::optional<AudioCodecPairId> codec_pair_id) {
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const WebRtcKeyValueConfig* field_trials) {
     auto opt_config = T::SdpToConfig(format);
     if (opt_config) {
       return T::MakeAudioEncoder(*opt_config, payload_type, codec_pair_id);
     } else {
       return Helper<Ts...>::MakeAudioEncoder(payload_type, format,
-                                             codec_pair_id);
+                                             codec_pair_id, field_trials);
     }
   }
 };
@@ -77,6 +80,10 @@ struct Helper<T, Ts...> {
 template <typename... Ts>
 class AudioEncoderFactoryT : public AudioEncoderFactory {
  public:
+  explicit AudioEncoderFactoryT(const WebRtcKeyValueConfig* field_trials) {
+    field_trials_ = field_trials;
+  }
+
   std::vector<AudioCodecSpec> GetSupportedEncoders() override {
     std::vector<AudioCodecSpec> specs;
     Helper<Ts...>::AppendSupportedEncoders(&specs);
@@ -92,8 +99,11 @@ class AudioEncoderFactoryT : public AudioEncoderFactory {
       int payload_type,
       const SdpAudioFormat& format,
       absl::optional<AudioCodecPairId> codec_pair_id) override {
-    return Helper<Ts...>::MakeAudioEncoder(payload_type, format, codec_pair_id);
+    return Helper<Ts...>::MakeAudioEncoder(payload_type, format, codec_pair_id,
+                                           field_trials_);
   }
+
+  const WebRtcKeyValueConfig* field_trials_;
 };
 
 }  // namespace audio_encoder_factory_template_impl
@@ -134,7 +144,8 @@ class AudioEncoderFactoryT : public AudioEncoderFactory {
 // TODO(kwiberg): Point at CreateBuiltinAudioEncoderFactory() for an example of
 // how it is used.
 template <typename... Ts>
-rtc::scoped_refptr<AudioEncoderFactory> CreateAudioEncoderFactory() {
+rtc::scoped_refptr<AudioEncoderFactory> CreateAudioEncoderFactory(
+    const WebRtcKeyValueConfig* field_trials = nullptr) {
   // There's no technical reason we couldn't allow zero template parameters,
   // but such a factory couldn't create any encoders, and callers can do this
   // by mistake by simply forgetting the <> altogether. So we forbid it in
@@ -143,7 +154,8 @@ rtc::scoped_refptr<AudioEncoderFactory> CreateAudioEncoderFactory() {
                 "Caller must give at least one template parameter");
 
   return rtc::make_ref_counted<
-      audio_encoder_factory_template_impl::AudioEncoderFactoryT<Ts...>>();
+      audio_encoder_factory_template_impl::AudioEncoderFactoryT<Ts...>>(
+      field_trials);
 }
 
 }  // namespace webrtc

api/audio_codecs/builtin_audio_encoder_factory.cc

@@ -47,8 +47,10 @@ struct NotAdvertised {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       const Config& config,
       int payload_type,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
-    return T::MakeAudioEncoder(config, payload_type, codec_pair_id);
+      const WebRtcKeyValueConfig* field_trials = nullptr) {
+    return T::MakeAudioEncoder(config, payload_type, codec_pair_id,
+                               field_trials);
   }
 };
 

api/audio_codecs/g711/BUILD.gn

@@ -21,6 +21,7 @@ rtc_library("audio_encoder_g711") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:g711",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
@@ -41,6 +42,7 @@ rtc_library("audio_decoder_g711") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:g711",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",

api/audio_codecs/g711/audio_decoder_g711.cc

@@ -28,7 +28,10 @@ absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
     Config config;
     config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
     config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
-    RTC_DCHECK(config.IsOk());
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
     return config;
   } else {
     return absl::nullopt;
@@ -45,14 +48,19 @@ void AudioDecoderG711::AppendSupportedDecoders(
 
 std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder(
     const Config& config,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
-  RTC_DCHECK(config.IsOk());
+    const WebRtcKeyValueConfig* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
   switch (config.type) {
     case Config::Type::kPcmU:
       return std::make_unique<AudioDecoderPcmU>(config.num_channels);
     case Config::Type::kPcmA:
       return std::make_unique<AudioDecoderPcmA>(config.num_channels);
     default:
+      RTC_DCHECK_NOTREACHED();
       return nullptr;
   }
 }

api/audio_codecs/g711/audio_decoder_g711.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -28,7 +29,9 @@ struct RTC_EXPORT AudioDecoderG711 {
   struct Config {
     enum class Type { kPcmU, kPcmA };
     bool IsOk() const {
-      return (type == Type::kPcmU || type == Type::kPcmA) && num_channels >= 1;
+      return (type == Type::kPcmU || type == Type::kPcmA) &&
+             num_channels >= 1 &&
+             num_channels <= AudioDecoder::kMaxNumberOfChannels;
     }
     Type type;
     int num_channels;
@@ -37,7 +40,8 @@ struct RTC_EXPORT AudioDecoderG711 {
   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       const Config& config,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/g711/audio_encoder_g711.cc

@@ -38,7 +38,10 @@ absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
         config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
       }
     }
-    RTC_DCHECK(config.IsOk());
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
     return config;
   } else {
     return absl::nullopt;
@@ -62,8 +65,12 @@ AudioCodecInfo AudioEncoderG711::QueryAudioEncoder(const Config& config) {
 std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
     const Config& config,
     int payload_type,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
-  RTC_DCHECK(config.IsOk());
+    const WebRtcKeyValueConfig* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
   switch (config.type) {
     case Config::Type::kPcmU: {
       AudioEncoderPcmU::Config impl_config;
@@ -80,6 +87,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
       return std::make_unique<AudioEncoderPcmA>(impl_config);
     }
     default: {
+      RTC_DCHECK_NOTREACHED();
       return nullptr;
     }
   }

api/audio_codecs/g711/audio_encoder_g711.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -29,7 +30,9 @@ struct RTC_EXPORT AudioEncoderG711 {
     enum class Type { kPcmU, kPcmA };
     bool IsOk() const {
       return (type == Type::kPcmU || type == Type::kPcmA) &&
-             frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1;
+             frame_size_ms > 0 && frame_size_ms % 10 == 0 &&
+             num_channels >= 1 &&
+             num_channels <= AudioEncoder::kMaxNumberOfChannels;
     }
     Type type = Type::kPcmU;
     int num_channels = 1;
@@ -42,7 +45,8 @@ struct RTC_EXPORT AudioEncoderG711 {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       const Config& config,
       int payload_type,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/g722/BUILD.gn

@@ -27,6 +27,7 @@ rtc_library("audio_encoder_g722") {
   deps = [
     ":audio_encoder_g722_config",
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:g722",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
@@ -47,6 +48,7 @@ rtc_library("audio_decoder_g722") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:g722",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",

api/audio_codecs/g722/audio_decoder_g722.cc

@@ -21,12 +21,12 @@ namespace webrtc {
 
 absl::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
     const SdpAudioFormat& format) {
-  return absl::EqualsIgnoreCase(format.name, "G722") &&
+  if (absl::EqualsIgnoreCase(format.name, "G722") &&
       format.clockrate_hz == 8000 &&
-                 (format.num_channels == 1 || format.num_channels == 2)
+      (format.num_channels == 1 || format.num_channels == 2)) {
-             ? absl::optional<Config>(
+    return Config{rtc::dchecked_cast<int>(format.num_channels)};
-                   Config{rtc::dchecked_cast<int>(format.num_channels)})
+  }
-             : absl::nullopt;
+  return absl::nullopt;
 }
 
 void AudioDecoderG722::AppendSupportedDecoders(
@@ -37,13 +37,19 @@ void AudioDecoderG722::AppendSupportedDecoders(
 
 std::unique_ptr<AudioDecoder> AudioDecoderG722::MakeAudioDecoder(
     Config config,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const WebRtcKeyValueConfig* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
   switch (config.num_channels) {
     case 1:
       return std::make_unique<AudioDecoderG722Impl>();
     case 2:
       return std::make_unique<AudioDecoderG722StereoImpl>();
     default:
+      RTC_DCHECK_NOTREACHED();
       return nullptr;
   }
 }

api/audio_codecs/g722/audio_decoder_g722.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -33,7 +34,8 @@ struct RTC_EXPORT AudioDecoderG722 {
   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       Config config,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/g722/audio_encoder_g722.cc

@@ -38,8 +38,11 @@ absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
       config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
     }
   }
-  return config.IsOk() ? absl::optional<AudioEncoderG722Config>(config)
+  if (!config.IsOk()) {
-                       : absl::nullopt;
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
 }
 
 void AudioEncoderG722::AppendSupportedEncoders(
@@ -60,8 +63,12 @@ AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
 std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
     const AudioEncoderG722Config& config,
     int payload_type,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
-  RTC_DCHECK(config.IsOk());
+    const WebRtcKeyValueConfig* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
   return std::make_unique<AudioEncoderG722Impl>(config, payload_type);
 }
 

api/audio_codecs/g722/audio_encoder_g722.h

@@ -19,6 +19,7 @@
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/g722/audio_encoder_g722_config.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -34,7 +35,8 @@ struct RTC_EXPORT AudioEncoderG722 {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       const AudioEncoderG722Config& config,
       int payload_type,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/g722/audio_encoder_g722_config.h

@@ -15,7 +15,8 @@ namespace webrtc {
 
 struct AudioEncoderG722Config {
   bool IsOk() const {
-    return frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1;
+    return frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1 &&
+           num_channels <= AudioEncoder::kMaxNumberOfChannels;
   }
   int frame_size_ms = 20;
   int num_channels = 1;

api/audio_codecs/ilbc/BUILD.gn

@@ -27,6 +27,7 @@ rtc_library("audio_encoder_ilbc") {
   deps = [
     ":audio_encoder_ilbc_config",
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:ilbc",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
@@ -46,6 +47,7 @@ rtc_library("audio_decoder_ilbc") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:ilbc",
     "../../../rtc_base:rtc_base_approved",
   ]

api/audio_codecs/ilbc/audio_decoder_ilbc.cc

@@ -20,10 +20,11 @@ namespace webrtc {
 
 absl::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
     const SdpAudioFormat& format) {
-  return absl::EqualsIgnoreCase(format.name, "ILBC") &&
+  if (absl::EqualsIgnoreCase(format.name, "ILBC") &&
-                 format.clockrate_hz == 8000 && format.num_channels == 1
+      format.clockrate_hz == 8000 && format.num_channels == 1) {
-             ? absl::optional<Config>(Config())
+    return Config();
-             : absl::nullopt;
+  }
+  return absl::nullopt;
 }
 
 void AudioDecoderIlbc::AppendSupportedDecoders(
@@ -34,7 +35,8 @@ void AudioDecoderIlbc::AppendSupportedDecoders(
 
 std::unique_ptr<AudioDecoder> AudioDecoderIlbc::MakeAudioDecoder(
     Config config,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const WebRtcKeyValueConfig* field_trials) {
   return std::make_unique<AudioDecoderIlbcImpl>();
 }
 

api/audio_codecs/ilbc/audio_decoder_ilbc.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/webrtc_key_value_config.h"
 
 namespace webrtc {
 
@@ -29,7 +30,8 @@ struct AudioDecoderIlbc {
   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       Config config,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/ilbc/audio_encoder_ilbc.cc

@@ -53,8 +53,11 @@ absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
       config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 20, 60);
     }
   }
-  return config.IsOk() ? absl::optional<AudioEncoderIlbcConfig>(config)
+  if (!config.IsOk()) {
-                       : absl::nullopt;
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
 }
 
 void AudioEncoderIlbc::AppendSupportedEncoders(
@@ -74,8 +77,12 @@ AudioCodecInfo AudioEncoderIlbc::QueryAudioEncoder(
 std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
     const AudioEncoderIlbcConfig& config,
     int payload_type,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
-  RTC_DCHECK(config.IsOk());
+    const WebRtcKeyValueConfig* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
   return std::make_unique<AudioEncoderIlbcImpl>(config, payload_type);
 }
 

api/audio_codecs/ilbc/audio_encoder_ilbc.h

@@ -19,6 +19,7 @@
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
+#include "api/webrtc_key_value_config.h"
 
 namespace webrtc {
 
@@ -33,7 +34,8 @@ struct AudioEncoderIlbc {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       const AudioEncoderIlbcConfig& config,
       int payload_type,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/isac/BUILD.gn

@@ -65,6 +65,7 @@ rtc_library("audio_encoder_isac_fix") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:isac_fix",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
@@ -84,6 +85,7 @@ rtc_library("audio_decoder_isac_fix") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:isac_fix",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
@@ -103,6 +105,7 @@ rtc_library("audio_encoder_isac_float") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:isac",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
@@ -122,6 +125,7 @@ rtc_library("audio_decoder_isac_float") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:isac",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",

api/audio_codecs/isac/audio_decoder_isac_fix.cc

@@ -19,10 +19,11 @@ namespace webrtc {
 
 absl::optional<AudioDecoderIsacFix::Config> AudioDecoderIsacFix::SdpToConfig(
     const SdpAudioFormat& format) {
-  return absl::EqualsIgnoreCase(format.name, "ISAC") &&
+  if (absl::EqualsIgnoreCase(format.name, "ISAC") &&
-                 format.clockrate_hz == 16000 && format.num_channels == 1
+      format.clockrate_hz == 16000 && format.num_channels == 1) {
-             ? absl::optional<Config>(Config())
+    return Config();
-             : absl::nullopt;
+  }
+  return absl::nullopt;
 }
 
 void AudioDecoderIsacFix::AppendSupportedDecoders(
@@ -33,7 +34,8 @@ void AudioDecoderIsacFix::AppendSupportedDecoders(
 
 std::unique_ptr<AudioDecoder> AudioDecoderIsacFix::MakeAudioDecoder(
     Config config,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const WebRtcKeyValueConfig* field_trials) {
   AudioDecoderIsacFixImpl::Config c;
   c.sample_rate_hz = 16000;
   return std::make_unique<AudioDecoderIsacFixImpl>(c);

api/audio_codecs/isac/audio_decoder_isac_fix.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -30,7 +31,8 @@ struct RTC_EXPORT AudioDecoderIsacFix {
   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       Config config,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/isac/audio_decoder_isac_float.cc

@@ -24,6 +24,10 @@ AudioDecoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
       format.num_channels == 1) {
     Config config;
     config.sample_rate_hz = format.clockrate_hz;
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
     return config;
   } else {
     return absl::nullopt;
@@ -39,10 +43,14 @@ void AudioDecoderIsacFloat::AppendSupportedDecoders(
 
 std::unique_ptr<AudioDecoder> AudioDecoderIsacFloat::MakeAudioDecoder(
     Config config,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
-  RTC_DCHECK(config.IsOk());
+    const WebRtcKeyValueConfig* field_trials) {
   AudioDecoderIsacFloatImpl::Config c;
   c.sample_rate_hz = config.sample_rate_hz;
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
   return std::make_unique<AudioDecoderIsacFloatImpl>(c);
 }
 

api/audio_codecs/isac/audio_decoder_isac_float.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -35,7 +36,8 @@ struct RTC_EXPORT AudioDecoderIsacFloat {
   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       Config config,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/isac/audio_encoder_isac_fix.cc

@@ -30,6 +30,10 @@ absl::optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
         config.frame_size_ms = 60;
       }
     }
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
     return config;
   } else {
     return absl::nullopt;
@@ -53,12 +57,16 @@ AudioCodecInfo AudioEncoderIsacFix::QueryAudioEncoder(
 std::unique_ptr<AudioEncoder> AudioEncoderIsacFix::MakeAudioEncoder(
     AudioEncoderIsacFix::Config config,
     int payload_type,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
-  RTC_DCHECK(config.IsOk());
+    const WebRtcKeyValueConfig* field_trials) {
   AudioEncoderIsacFixImpl::Config c;
   c.frame_size_ms = config.frame_size_ms;
   c.bit_rate = config.bit_rate;
   c.payload_type = payload_type;
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
   return std::make_unique<AudioEncoderIsacFixImpl>(c);
 }
 

api/audio_codecs/isac/audio_encoder_isac_fix.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -44,7 +45,8 @@ struct RTC_EXPORT AudioEncoderIsacFix {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       Config config,
       int payload_type,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/isac/audio_encoder_isac_float.cc

@@ -37,6 +37,10 @@ AudioEncoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
         }
       }
     }
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
     return config;
   } else {
     return absl::nullopt;
@@ -65,13 +69,17 @@ AudioCodecInfo AudioEncoderIsacFloat::QueryAudioEncoder(
 std::unique_ptr<AudioEncoder> AudioEncoderIsacFloat::MakeAudioEncoder(
     const AudioEncoderIsacFloat::Config& config,
     int payload_type,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
-  RTC_DCHECK(config.IsOk());
+    const WebRtcKeyValueConfig* field_trials) {
   AudioEncoderIsacFloatImpl::Config c;
   c.payload_type = payload_type;
   c.sample_rate_hz = config.sample_rate_hz;
   c.frame_size_ms = config.frame_size_ms;
   c.bit_rate = config.bit_rate;
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
   return std::make_unique<AudioEncoderIsacFloatImpl>(c);
 }
 

api/audio_codecs/isac/audio_encoder_isac_float.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -58,7 +59,8 @@ struct RTC_EXPORT AudioEncoderIsacFloat {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       const Config& config,
       int payload_type,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/opus/BUILD.gn

@@ -46,6 +46,7 @@ rtc_library("audio_encoder_opus") {
   deps = [
     ":audio_encoder_opus_config",
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:webrtc_opus",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
@@ -65,6 +66,7 @@ rtc_library("audio_decoder_opus") {
   ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:webrtc_opus",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
@@ -82,6 +84,7 @@ rtc_library("audio_encoder_multiopus") {
   sources = [ "audio_encoder_multi_channel_opus.cc" ]
   deps = [
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:webrtc_multiopus",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
@@ -100,6 +103,7 @@ rtc_library("audio_decoder_multiopus") {
   deps = [
     ":audio_decoder_opus_config",
     "..:audio_codecs_api",
+    "../../../api:webrtc_key_value_config",
     "../../../modules/audio_coding:webrtc_multiopus",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",

api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc

@@ -64,7 +64,8 @@ void AudioDecoderMultiChannelOpus::AppendSupportedDecoders(
 
 std::unique_ptr<AudioDecoder> AudioDecoderMultiChannelOpus::MakeAudioDecoder(
     AudioDecoderMultiChannelOpusConfig config,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const WebRtcKeyValueConfig* field_trials) {
   return AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(config);
 }
 }  // namespace webrtc

api/audio_codecs/opus/audio_decoder_multi_channel_opus.h

@@ -19,6 +19,7 @@
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -32,7 +33,8 @@ struct RTC_EXPORT AudioDecoderMultiChannelOpus {
   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       AudioDecoderMultiChannelOpusConfig config,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h

@@ -30,7 +30,8 @@ struct AudioDecoderMultiChannelOpusConfig {
   std::vector<unsigned char> channel_mapping;
 
   bool IsOk() const {
-    if (num_channels < 0 || num_streams < 0 || coupled_streams < 0) {
+    if (num_channels < 1 || num_channels > AudioDecoder::kMaxNumberOfChannels ||
+        num_streams < 0 || coupled_streams < 0) {
       return false;
     }
     if (num_streams < coupled_streams) {

api/audio_codecs/opus/audio_decoder_opus.cc

@@ -51,7 +51,10 @@ absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
       num_channels) {
     Config config;
     config.num_channels = *num_channels;
-    RTC_DCHECK(config.IsOk());
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
     return config;
   } else {
     return absl::nullopt;
@@ -70,8 +73,12 @@ void AudioDecoderOpus::AppendSupportedDecoders(
 
 std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
     Config config,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
-  RTC_DCHECK(config.IsOk());
+    const WebRtcKeyValueConfig* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
   return std::make_unique<AudioDecoderOpusImpl>(config.num_channels,
                                                 config.sample_rate_hz);
 }

api/audio_codecs/opus/audio_decoder_opus.h

@@ -18,6 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -34,7 +35,8 @@ struct RTC_EXPORT AudioDecoderOpus {
   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
       Config config,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc

@@ -66,7 +66,8 @@ AudioCodecInfo AudioEncoderMultiChannelOpus::QueryAudioEncoder(
 std::unique_ptr<AudioEncoder> AudioEncoderMultiChannelOpus::MakeAudioEncoder(
     const AudioEncoderMultiChannelOpusConfig& config,
     int payload_type,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const WebRtcKeyValueConfig* field_trials) {
   return AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder(config,
                                                             payload_type);
 }

api/audio_codecs/opus/audio_encoder_multi_channel_opus.h

@@ -19,6 +19,7 @@
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -33,7 +34,8 @@ struct RTC_EXPORT AudioEncoderMultiChannelOpus {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       const Config& config,
       int payload_type,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/opus/audio_encoder_opus.cc

@@ -32,7 +32,12 @@ AudioCodecInfo AudioEncoderOpus::QueryAudioEncoder(
 std::unique_ptr<AudioEncoder> AudioEncoderOpus::MakeAudioEncoder(
     const AudioEncoderOpusConfig& config,
     int payload_type,
-    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const WebRtcKeyValueConfig* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
   return AudioEncoderOpusImpl::MakeAudioEncoder(config, payload_type);
 }
 

api/audio_codecs/opus/audio_encoder_opus.h

@@ -19,6 +19,7 @@
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "api/webrtc_key_value_config.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -34,7 +35,8 @@ struct RTC_EXPORT AudioEncoderOpus {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       const AudioEncoderOpusConfig& config,
       int payload_type,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const WebRtcKeyValueConfig* field_trials = nullptr);
 };
 
 }  // namespace webrtc

api/audio_codecs/opus_audio_encoder_factory.cc

@@ -37,8 +37,10 @@ struct NotAdvertised {
   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
       const Config& config,
       int payload_type,
-      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
-    return T::MakeAudioEncoder(config, payload_type, codec_pair_id);
+      const WebRtcKeyValueConfig* field_trials = nullptr) {
+    return T::MakeAudioEncoder(config, payload_type, codec_pair_id,
+                               field_trials);
   }
 };
 

api/audio_codecs/test/BUILD.gn

@@ -23,6 +23,7 @@ if (rtc_include_tests) {
       "..:audio_codecs_api",
       "../../../rtc_base:rtc_base_approved",
       "../../../test:audio_codec_mocks",
+      "../../../test:scoped_key_value_config",
       "../../../test:test_support",
       "../L16:audio_decoder_L16",
       "../L16:audio_encoder_L16",

api/audio_codecs/test/audio_decoder_factory_template_unittest.cc

@@ -22,6 +22,7 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/mock_audio_decoder.h"
+#include "test/scoped_key_value_config.h"
 
 namespace webrtc {
 
@@ -77,9 +78,11 @@ struct AudioDecoderFakeApi {
 }  // namespace
 
 TEST(AudioDecoderFactoryTemplateTest, NoDecoderTypes) {
+  test::ScopedKeyValueConfig field_trials;
   rtc::scoped_refptr<AudioDecoderFactory> factory(
       rtc::make_ref_counted<
-          audio_decoder_factory_template_impl::AudioDecoderFactoryT<>>());
+          audio_decoder_factory_template_impl::AudioDecoderFactoryT<>>(
+          &field_trials));
   EXPECT_THAT(factory->GetSupportedDecoders(), ::testing::IsEmpty());
   EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
   EXPECT_EQ(nullptr,

api/audio_codecs/test/audio_encoder_factory_template_unittest.cc

@@ -22,6 +22,7 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/mock_audio_encoder.h"
+#include "test/scoped_key_value_config.h"
 
 namespace webrtc {
 
@@ -77,9 +78,11 @@ struct AudioEncoderFakeApi {
 }  // namespace
 
 TEST(AudioEncoderFactoryTemplateTest, NoEncoderTypes) {
+  test::ScopedKeyValueConfig field_trials;
   rtc::scoped_refptr<AudioEncoderFactory> factory(
       rtc::make_ref_counted<
-          audio_encoder_factory_template_impl::AudioEncoderFactoryT<>>());
+          audio_encoder_factory_template_impl::AudioEncoderFactoryT<>>(
+          &field_trials));
   EXPECT_THAT(factory->GetSupportedEncoders(), ::testing::IsEmpty());
   EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
   EXPECT_EQ(nullptr,

api/audio_options.cc

@@ -55,16 +55,11 @@ void AudioOptions::SetAll(const AudioOptions& change) {
   SetFrom(&audio_jitter_buffer_enable_rtx_handling,
           change.audio_jitter_buffer_enable_rtx_handling);
   SetFrom(&typing_detection, change.typing_detection);
-  SetFrom(&experimental_agc, change.experimental_agc);
-  SetFrom(&experimental_ns, change.experimental_ns);
   SetFrom(&residual_echo_detector, change.residual_echo_detector);
-  SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
-  SetFrom(&tx_agc_digital_compression_gain,
-          change.tx_agc_digital_compression_gain);
-  SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
   SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
   SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
   SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
+  SetFrom(&init_recording_on_send, change.init_recording_on_send);
 }
 
 bool AudioOptions::operator==(const AudioOptions& o) const {
@@ -84,15 +79,11 @@ bool AudioOptions::operator==(const AudioOptions& o) const {
          audio_jitter_buffer_enable_rtx_handling ==
              o.audio_jitter_buffer_enable_rtx_handling &&
          typing_detection == o.typing_detection &&
-         experimental_agc == o.experimental_agc &&
-         experimental_ns == o.experimental_ns &&
          residual_echo_detector == o.residual_echo_detector &&
-         tx_agc_target_dbov == o.tx_agc_target_dbov &&
-         tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
-         tx_agc_limiter == o.tx_agc_limiter &&
          combined_audio_video_bwe == o.combined_audio_video_bwe &&
          audio_network_adaptor == o.audio_network_adaptor &&
-         audio_network_adaptor_config == o.audio_network_adaptor_config;
+         audio_network_adaptor_config == o.audio_network_adaptor_config &&
+         init_recording_on_send == o.init_recording_on_send;
 }
 
 std::string AudioOptions::ToString() const {
@@ -117,15 +108,10 @@ std::string AudioOptions::ToString() const {
   ToStringIfSet(&result, "audio_jitter_buffer_enable_rtx_handling",
                 audio_jitter_buffer_enable_rtx_handling);
   ToStringIfSet(&result, "typing", typing_detection);
-  ToStringIfSet(&result, "experimental_agc", experimental_agc);
-  ToStringIfSet(&result, "experimental_ns", experimental_ns);
   ToStringIfSet(&result, "residual_echo_detector", residual_echo_detector);
-  ToStringIfSet(&result, "tx_agc_target_dbov", tx_agc_target_dbov);
-  ToStringIfSet(&result, "tx_agc_digital_compression_gain",
-                tx_agc_digital_compression_gain);
-  ToStringIfSet(&result, "tx_agc_limiter", tx_agc_limiter);
   ToStringIfSet(&result, "combined_audio_video_bwe", combined_audio_video_bwe);
   ToStringIfSet(&result, "audio_network_adaptor", audio_network_adaptor);
+  ToStringIfSet(&result, "init_recording_on_send", init_recording_on_send);
   result << "}";
   return result.str();
 }

api/audio_options.h

@@ -60,15 +60,14 @@ struct RTC_EXPORT AudioOptions {
   absl::optional<int> audio_jitter_buffer_min_delay_ms;
   // Audio receiver jitter buffer (NetEq) should handle retransmitted packets.
   absl::optional<bool> audio_jitter_buffer_enable_rtx_handling;
+  // Deprecated.
+  // TODO(bugs.webrtc.org/11226): Remove.
   // Audio processing to detect typing.
   absl::optional<bool> typing_detection;
-  absl::optional<bool> experimental_agc;
+  // TODO(bugs.webrtc.org/11539): Deprecated, replaced by
-  absl::optional<bool> experimental_ns;
+  // webrtc::CreateEchoDetector() and injection when creating the audio
-  // Note that tx_agc_* only applies to non-experimental AGC.
+  // processing module.
   absl::optional<bool> residual_echo_detector;
-  absl::optional<uint16_t> tx_agc_target_dbov;
-  absl::optional<uint16_t> tx_agc_digital_compression_gain;
-  absl::optional<bool> tx_agc_limiter;
   // Enable combined audio+bandwidth BWE.
   // TODO(pthatcher): This flag is set from the
   // "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
@@ -80,6 +79,10 @@ struct RTC_EXPORT AudioOptions {
   absl::optional<bool> audio_network_adaptor;
   // Config string for audio network adaptor.
   absl::optional<std::string> audio_network_adaptor_config;
+  // Pre-initialize the ADM for recording when starting to send. Default to
+  // true.
+  // TODO(webrtc:13566): Remove this option. See issue for details.
+  absl::optional<bool> init_recording_on_send;
 };
 
 }  // namespace cricket

api/candidate.cc

@@ -27,14 +27,14 @@ Candidate::Candidate()
       network_cost_(0) {}
 
 Candidate::Candidate(int component,
-                     const std::string& protocol,
+                     absl::string_view protocol,
                      const rtc::SocketAddress& address,
                      uint32_t priority,
-                     const std::string& username,
+                     absl::string_view username,
-                     const std::string& password,
+                     absl::string_view password,
-                     const std::string& type,
+                     absl::string_view type,
                      uint32_t generation,
-                     const std::string& foundation,
+                     absl::string_view foundation,
                      uint16_t network_id,
                      uint16_t network_cost)
     : id_(rtc::CreateRandomString(8)),
@@ -101,7 +101,9 @@ uint32_t Candidate::GetPriority(uint32_t type_preference,
   //      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   // NIC Type - Type of the network adapter e.g. 3G/Wifi/Wired.
   // Addr Pref - Address preference value as per RFC 3484.
-  // local preference =  (NIC Type << 8 | Addr_Pref) - relay preference.
+  // local preference =  (NIC Type << 8 | Addr_Pref) + relay preference.
+  // The relay preference is based on the number of TURN servers, the
+  // first TURN server gets the highest preference.
 
   int addr_pref = IPAddressPrecedence(address_.ipaddr());
   int local_preference =
@@ -153,4 +155,11 @@ Candidate Candidate::ToSanitizedCopy(bool use_hostname_address,
   return copy;
 }
 
+void Candidate::Assign(std::string& s, absl::string_view view) {
+  // Assigning via a temporary object, like s = std::string(view), results in
+  // binary size bloat. To avoid that, extract pointer and size from the
+  // string view, and use std::string::assign method.
+  s.assign(view.data(), view.size());
+}
+
 }  // namespace cricket

api/candidate.h

@@ -17,6 +17,7 @@
 #include <algorithm>
 #include <string>
 
+#include "absl/strings/string_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/network_constants.h"
 #include "rtc_base/socket_address.h"
@@ -33,32 +34,32 @@ class RTC_EXPORT Candidate {
   // TODO(pthatcher): Match the ordering and param list as per RFC 5245
   // candidate-attribute syntax. http://tools.ietf.org/html/rfc5245#section-15.1
   Candidate(int component,
-            const std::string& protocol,
+            absl::string_view protocol,
             const rtc::SocketAddress& address,
             uint32_t priority,
-            const std::string& username,
+            absl::string_view username,
-            const std::string& password,
+            absl::string_view password,
-            const std::string& type,
+            absl::string_view type,
             uint32_t generation,
-            const std::string& foundation,
+            absl::string_view foundation,
             uint16_t network_id = 0,
             uint16_t network_cost = 0);
   Candidate(const Candidate&);
   ~Candidate();
 
   const std::string& id() const { return id_; }
-  void set_id(const std::string& id) { id_ = id; }
+  void set_id(absl::string_view id) { Assign(id_, id); }
 
   int component() const { return component_; }
   void set_component(int component) { component_ = component; }
 
   const std::string& protocol() const { return protocol_; }
-  void set_protocol(const std::string& protocol) { protocol_ = protocol; }
+  void set_protocol(absl::string_view protocol) { Assign(protocol_, protocol); }
 
   // The protocol used to talk to relay.
   const std::string& relay_protocol() const { return relay_protocol_; }
-  void set_relay_protocol(const std::string& protocol) {
+  void set_relay_protocol(absl::string_view protocol) {
-    relay_protocol_ = protocol;
+    Assign(relay_protocol_, protocol);
   }
 
   const rtc::SocketAddress& address() const { return address_; }
@@ -90,17 +91,17 @@ class RTC_EXPORT Candidate {
 
   // TODO(honghaiz): Change to usernameFragment or ufrag.
   const std::string& username() const { return username_; }
-  void set_username(const std::string& username) { username_ = username; }
+  void set_username(absl::string_view username) { Assign(username_, username); }
 
   const std::string& password() const { return password_; }
-  void set_password(const std::string& password) { password_ = password; }
+  void set_password(absl::string_view password) { Assign(password_, password); }
 
   const std::string& type() const { return type_; }
-  void set_type(const std::string& type) { type_ = type; }
+  void set_type(absl::string_view type) { Assign(type_, type); }
 
   const std::string& network_name() const { return network_name_; }
-  void set_network_name(const std::string& network_name) {
+  void set_network_name(absl::string_view network_name) {
-    network_name_ = network_name;
+    Assign(network_name_, network_name);
   }
 
   rtc::AdapterType network_type() const { return network_type_; }
@@ -108,6 +109,13 @@ class RTC_EXPORT Candidate {
     network_type_ = network_type;
   }
 
+  rtc::AdapterType underlying_type_for_vpn() const {
+    return underlying_type_for_vpn_;
+  }
+  void set_underlying_type_for_vpn(rtc::AdapterType network_type) {
+    underlying_type_for_vpn_ = network_type;
+  }
+
   // Candidates in a new generation replace those in the old generation.
   uint32_t generation() const { return generation_; }
   void set_generation(uint32_t generation) { generation_ = generation; }
@@ -126,8 +134,8 @@ class RTC_EXPORT Candidate {
   void set_network_id(uint16_t network_id) { network_id_ = network_id; }
 
   const std::string& foundation() const { return foundation_; }
-  void set_foundation(const std::string& foundation) {
+  void set_foundation(absl::string_view foundation) {
-    foundation_ = foundation;
+    Assign(foundation_, foundation);
   }
 
   const rtc::SocketAddress& related_address() const { return related_address_; }
@@ -135,18 +143,18 @@ class RTC_EXPORT Candidate {
     related_address_ = related_address;
   }
   const std::string& tcptype() const { return tcptype_; }
-  void set_tcptype(const std::string& tcptype) { tcptype_ = tcptype; }
+  void set_tcptype(absl::string_view tcptype) { Assign(tcptype_, tcptype); }
 
   // The name of the transport channel of this candidate.
   // TODO(phoglund): remove.
   const std::string& transport_name() const { return transport_name_; }
-  void set_transport_name(const std::string& transport_name) {
+  void set_transport_name(absl::string_view transport_name) {
-    transport_name_ = transport_name;
+    Assign(transport_name_, transport_name);
   }
 
   // The URL of the ICE server which this candidate is gathered from.
   const std::string& url() const { return url_; }
-  void set_url(const std::string& url) { url_ = url; }
+  void set_url(absl::string_view url) { Assign(url_, url); }
 
   // Determines whether this candidate is equivalent to the given one.
   bool IsEquivalent(const Candidate& c) const;
@@ -177,6 +185,10 @@ class RTC_EXPORT Candidate {
                             bool filter_related_address) const;
 
  private:
+  // TODO(bugs.webrtc.org/13220): With C++17, we get a std::string assignment
+  // operator accepting any object implicitly convertible to std::string_view,
+  // and then we don't need this workaround.
+  static void Assign(std::string& s, absl::string_view view);
   std::string ToStringInternal(bool sensitive) const;
 
   std::string id_;
@@ -190,6 +202,7 @@ class RTC_EXPORT Candidate {
   std::string type_;
   std::string network_name_;
   rtc::AdapterType network_type_;
+  rtc::AdapterType underlying_type_for_vpn_;
   uint32_t generation_;
   std::string foundation_;
   rtc::SocketAddress related_address_;

api/create_peerconnection_factory.cc

@@ -49,6 +49,10 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
       dependencies.task_queue_factory.get());
   dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
 
+  if (network_thread) {
+    // TODO(bugs.webrtc.org/13145): Add an rtc::SocketFactory* argument.
+    dependencies.socket_factory = network_thread->socketserver();
+  }
   cricket::MediaEngineDependencies media_dependencies;
   media_dependencies.task_queue_factory = dependencies.task_queue_factory.get();
   media_dependencies.adm = std::move(default_adm);

api/data_channel_interface.cc

@@ -40,4 +40,8 @@ bool DataChannelInterface::negotiated() const {
   return false;
 }
 
+uint64_t DataChannelInterface::MaxSendQueueSize() {
+  return 16 * 1024 * 1024;  // 16 MiB
+}
+
 }  // namespace webrtc

api/data_channel_interface.h

@@ -174,6 +174,7 @@ class RTC_EXPORT DataChannelInterface : public rtc::RefCountInterface {
   // Returns the number of bytes of application data (UTF-8 text and binary
   // data) that have been queued using Send but have not yet been processed at
   // the SCTP level. See comment above Send below.
+  // Values are less or equal to MaxSendQueueSize().
   virtual uint64_t buffered_amount() const = 0;
 
   // Begins the graceful data channel closing procedure. See:
@@ -182,14 +183,16 @@ class RTC_EXPORT DataChannelInterface : public rtc::RefCountInterface {
 
   // Sends `data` to the remote peer. If the data can't be sent at the SCTP
   // level (due to congestion control), it's buffered at the data channel level,
-  // up to a maximum of 16MB. If Send is called while this buffer is full, the
+  // up to a maximum of MaxSendQueueSize().
-  // data channel will be closed abruptly.
+  // Returns false if the data channel is not in open state or if the send
-  //
+  // buffer is full.
-  // So, it's important to use buffered_amount() and OnBufferedAmountChange to
+  // TODO(webrtc:13289): Return an RTCError with information about the failure.
-  // ensure the data channel is used efficiently but without filling this
-  // buffer.
   virtual bool Send(const DataBuffer& buffer) = 0;
 
+  // Amount of bytes that can be queued for sending on the data channel.
+  // Those are bytes that have not yet been processed at the SCTP level.
+  static uint64_t MaxSendQueueSize();
+
  protected:
   ~DataChannelInterface() override = default;
 };

api/dtls_transport_interface.cc

@@ -20,11 +20,27 @@ DtlsTransportInformation::DtlsTransportInformation(DtlsTransportState state)
 
 DtlsTransportInformation::DtlsTransportInformation(
     DtlsTransportState state,
+    absl::optional<DtlsTransportTlsRole> role,
     absl::optional<int> tls_version,
     absl::optional<int> ssl_cipher_suite,
     absl::optional<int> srtp_cipher_suite,
     std::unique_ptr<rtc::SSLCertChain> remote_ssl_certificates)
     : state_(state),
+      role_(role),
+      tls_version_(tls_version),
+      ssl_cipher_suite_(ssl_cipher_suite),
+      srtp_cipher_suite_(srtp_cipher_suite),
+      remote_ssl_certificates_(std::move(remote_ssl_certificates)) {}
+
+// Deprecated version
+DtlsTransportInformation::DtlsTransportInformation(
+    DtlsTransportState state,
+    absl::optional<int> tls_version,
+    absl::optional<int> ssl_cipher_suite,
+    absl::optional<int> srtp_cipher_suite,
+    std::unique_ptr<rtc::SSLCertChain> remote_ssl_certificates)
+    : state_(state),
+      role_(absl::nullopt),
       tls_version_(tls_version),
       ssl_cipher_suite_(ssl_cipher_suite),
       srtp_cipher_suite_(srtp_cipher_suite),
@@ -33,6 +49,7 @@ DtlsTransportInformation::DtlsTransportInformation(
 DtlsTransportInformation::DtlsTransportInformation(
     const DtlsTransportInformation& c)
     : state_(c.state()),
+      role_(c.role_),
       tls_version_(c.tls_version_),
       ssl_cipher_suite_(c.ssl_cipher_suite_),
       srtp_cipher_suite_(c.srtp_cipher_suite_),
@@ -43,6 +60,7 @@ DtlsTransportInformation::DtlsTransportInformation(
 DtlsTransportInformation& DtlsTransportInformation::operator=(
     const DtlsTransportInformation& c) {
   state_ = c.state();
+  role_ = c.role_;
   tls_version_ = c.tls_version_;
   ssl_cipher_suite_ = c.ssl_cipher_suite_;
   srtp_cipher_suite_ = c.srtp_cipher_suite_;

api/dtls_transport_interface.h

@@ -36,6 +36,11 @@ enum class DtlsTransportState {
   kNumValues
 };
 
+enum class DtlsTransportTlsRole {
+  kServer,  // Other end sends CLIENT_HELLO
+  kClient   // This end sends CLIENT_HELLO
+};
+
 // This object gives snapshot information about the changeable state of a
 // DTLSTransport.
 class RTC_EXPORT DtlsTransportInformation {
@@ -44,10 +49,19 @@ class RTC_EXPORT DtlsTransportInformation {
   explicit DtlsTransportInformation(DtlsTransportState state);
   DtlsTransportInformation(
       DtlsTransportState state,
+      absl::optional<DtlsTransportTlsRole> role,
       absl::optional<int> tls_version,
       absl::optional<int> ssl_cipher_suite,
       absl::optional<int> srtp_cipher_suite,
       std::unique_ptr<rtc::SSLCertChain> remote_ssl_certificates);
+  ABSL_DEPRECATED("Use version with role parameter")
+  DtlsTransportInformation(
+      DtlsTransportState state,
+      absl::optional<int> tls_version,
+      absl::optional<int> ssl_cipher_suite,
+      absl::optional<int> srtp_cipher_suite,
+      std::unique_ptr<rtc::SSLCertChain> remote_ssl_certificates);
+
   // Copy and assign
   DtlsTransportInformation(const DtlsTransportInformation& c);
   DtlsTransportInformation& operator=(const DtlsTransportInformation& c);
@@ -57,6 +71,7 @@ class RTC_EXPORT DtlsTransportInformation {
       default;
 
   DtlsTransportState state() const { return state_; }
+  absl::optional<DtlsTransportTlsRole> role() const { return role_; }
   absl::optional<int> tls_version() const { return tls_version_; }
   absl::optional<int> ssl_cipher_suite() const { return ssl_cipher_suite_; }
   absl::optional<int> srtp_cipher_suite() const { return srtp_cipher_suite_; }
@@ -67,6 +82,7 @@ class RTC_EXPORT DtlsTransportInformation {
 
  private:
   DtlsTransportState state_;
+  absl::optional<DtlsTransportTlsRole> role_;
   absl::optional<int> tls_version_;
   absl::optional<int> ssl_cipher_suite_;
   absl::optional<int> srtp_cipher_suite_;

api/ice_transport_factory.cc

@@ -58,18 +58,9 @@ rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
 
 rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
     IceTransportInit init) {
-  if (init.async_resolver_factory()) {
-    // Backwards compatibility mode
-    return rtc::make_ref_counted<IceTransportWithTransportChannel>(
-        std::make_unique<cricket::P2PTransportChannel>(
-            "", cricket::ICE_CANDIDATE_COMPONENT_RTP, init.port_allocator(),
-            init.async_resolver_factory(), init.event_log()));
-  } else {
   return rtc::make_ref_counted<IceTransportWithTransportChannel>(
       cricket::P2PTransportChannel::Create(
-            "", cricket::ICE_CANDIDATE_COMPONENT_RTP, init.port_allocator(),
+          "", cricket::ICE_CANDIDATE_COMPONENT_RTP, std::move(init)));
-            init.async_dns_resolver_factory(), init.event_log()));
-  }
 }
 
 }  // namespace webrtc

api/ice_transport_interface.h

@@ -23,9 +23,11 @@
 namespace cricket {
 class IceTransportInternal;
 class PortAllocator;
+class IceControllerFactoryInterface;
 }  // namespace cricket
 
 namespace webrtc {
+class WebRtcKeyValueConfig;
 
 // An ICE transport, as represented to the outside world.
 // This object is refcounted, and is therefore alive until the
@@ -74,12 +76,27 @@ struct IceTransportInit final {
   RtcEventLog* event_log() { return event_log_; }
   void set_event_log(RtcEventLog* event_log) { event_log_ = event_log; }
 
+  void set_ice_controller_factory(
+      cricket::IceControllerFactoryInterface* ice_controller_factory) {
+    ice_controller_factory_ = ice_controller_factory;
+  }
+  cricket::IceControllerFactoryInterface* ice_controller_factory() {
+    return ice_controller_factory_;
+  }
+
+  const WebRtcKeyValueConfig* field_trials() { return field_trials_; }
+  void set_field_trials(const WebRtcKeyValueConfig* field_trials) {
+    field_trials_ = field_trials;
+  }
+
  private:
   cricket::PortAllocator* port_allocator_ = nullptr;
   AsyncDnsResolverFactoryInterface* async_dns_resolver_factory_ = nullptr;
   // For backwards compatibility. Only one resolver factory can be set.
   AsyncResolverFactory* async_resolver_factory_ = nullptr;
   RtcEventLog* event_log_ = nullptr;
+  cricket::IceControllerFactoryInterface* ice_controller_factory_ = nullptr;
+  const WebRtcKeyValueConfig* field_trials_ = nullptr;
   // TODO(https://crbug.com/webrtc/12657): Redesign to have const members.
 };
 

api/jsep_ice_candidate.h

@@ -22,7 +22,6 @@
 
 #include "api/candidate.h"
 #include "api/jsep.h"
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -64,6 +63,10 @@ class JsepCandidateCollection : public IceCandidateCollection {
   // Move constructor is defined so that a vector of JsepCandidateCollections
   // can be resized.
   JsepCandidateCollection(JsepCandidateCollection&& o);
+
+  JsepCandidateCollection(const JsepCandidateCollection&) = delete;
+  JsepCandidateCollection& operator=(const JsepCandidateCollection&) = delete;
+
   // Returns a copy of the candidate collection.
   JsepCandidateCollection Clone() const;
   size_t count() const override;
@@ -80,8 +83,6 @@ class JsepCandidateCollection : public IceCandidateCollection {
 
  private:
   std::vector<std::unique_ptr<JsepIceCandidate>> candidates_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(JsepCandidateCollection);
 };
 
 }  // namespace webrtc

api/jsep_session_description.h

@@ -22,7 +22,6 @@
 #include "api/candidate.h"
 #include "api/jsep.h"
 #include "api/jsep_ice_candidate.h"
-#include "rtc_base/constructor_magic.h"
 
 namespace cricket {
 class SessionDescription;
@@ -43,6 +42,9 @@ class JsepSessionDescription : public SessionDescriptionInterface {
       absl::string_view session_version);
   virtual ~JsepSessionDescription();
 
+  JsepSessionDescription(const JsepSessionDescription&) = delete;
+  JsepSessionDescription& operator=(const JsepSessionDescription&) = delete;
+
   // Takes ownership of `description`.
   bool Initialize(std::unique_ptr<cricket::SessionDescription> description,
                   const std::string& session_id,
@@ -82,8 +84,6 @@ class JsepSessionDescription : public SessionDescriptionInterface {
   bool GetMediasectionIndex(const IceCandidateInterface* candidate,
                             size_t* index);
   int GetMediasectionIndex(const cricket::Candidate& candidate);
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(JsepSessionDescription);
 };
 
 }  // namespace webrtc

api/media_stream_interface.h

@@ -28,6 +28,7 @@
 #include "api/video/video_frame.h"
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_source_interface.h"
+#include "api/video_track_source_constraints.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
 #include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
@@ -146,8 +147,6 @@ class VideoTrackSourceInterface : public MediaSourceInterface,
   // Add an encoded video sink to the source and additionally cause
   // a key frame to be generated from the source. The sink will be
   // invoked from a decoder queue.
-  // TODO(bugs.webrtc.org/11114): make pure virtual once downstream project
-  // adapts.
   virtual void AddEncodedSink(
       rtc::VideoSinkInterface<RecordableEncodedFrame>* sink) = 0;
 
@@ -155,6 +154,13 @@ class VideoTrackSourceInterface : public MediaSourceInterface,
   virtual void RemoveEncodedSink(
       rtc::VideoSinkInterface<RecordableEncodedFrame>* sink) = 0;
 
+  // Notify about constraints set on the source. The information eventually gets
+  // routed to attached sinks via VideoSinkInterface<>::OnConstraintsChanged.
+  // The call is expected to happen on the network thread.
+  // TODO(crbug/1255737): make pure virtual once downstream project adapts.
+  virtual void ProcessConstraints(
+      const webrtc::VideoTrackSourceConstraints& constraints) {}
+
  protected:
   ~VideoTrackSourceInterface() override = default;
 };
@@ -197,7 +203,7 @@ class AudioTrackSinkInterface {
                       int sample_rate,
                       size_t number_of_channels,
                       size_t number_of_frames) {
-    RTC_NOTREACHED() << "This method must be overridden, or not used.";
+    RTC_DCHECK_NOTREACHED() << "This method must be overridden, or not used.";
   }
 
   // In this method, `absolute_capture_timestamp_ms`, when available, is
@@ -327,6 +333,8 @@ class MediaStreamInterface : public rtc::RefCountInterface,
   virtual rtc::scoped_refptr<VideoTrackInterface> FindVideoTrack(
       const std::string& track_id) = 0;
 
+  // Takes ownership of added tracks.
+  // TODO(hta): Should take scoped_refptr rather than raw pointer.
   virtual bool AddTrack(AudioTrackInterface* track) = 0;
   virtual bool AddTrack(VideoTrackInterface* track) = 0;
   virtual bool RemoveTrack(AudioTrackInterface* track) = 0;

api/media_types.cc

@@ -28,7 +28,7 @@ std::string MediaTypeToString(MediaType type) {
       return kMediaTypeData;
     case MEDIA_TYPE_UNSUPPORTED:
       // Unsupported media stores the m=<mediatype> differently.
-      RTC_NOTREACHED();
+      RTC_DCHECK_NOTREACHED();
       return "";
   }
   RTC_CHECK_NOTREACHED();

api/metronome/BUILD.gn

@@ -1,15 +1,19 @@
-#!/bin/sh
+# Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
-#
-# 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.
-#
+
-# This script is used to launch a simple http server for files in the same
+import("../../webrtc.gni")
-# location as the script itself.
+
-cd "`dirname \"$0\"`"
+rtc_source_set("metronome") {
-echo "Starting http server in port 8080."
+  visibility = [ "*" ]
-exec python -m SimpleHTTPServer 8080
+  sources = [ "metronome.h" ]
+  deps = [
+    "../../rtc_base/system:rtc_export",
+    "../task_queue",
+    "../units:time_delta",
+  ]
+}

api/metronome/metronome.h

@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2022 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.
+ */
+
+#ifndef API_METRONOME_METRONOME_H_
+#define API_METRONOME_METRONOME_H_
+
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// The Metronome posts OnTick() on task queues provided by its listeners' task
+// queue periodically. The metronome can be used as an alternative to using
+// PostDelayedTask on a thread or task queue for coalescing work and reducing
+// the number of idle-wakeups.
+//
+// Listeners can be added and removed from any sequence, but it is illegal to
+// remove a listener from an OnTick invocation.
+//
+// The metronome concept is still under experimentation, and may not be availble
+// in all platforms or applications. See https://crbug.com/1253787 for more
+// details.
+//
+// Metronome implementations must be thread-safe.
+class RTC_EXPORT Metronome {
+ public:
+  class RTC_EXPORT TickListener {
+   public:
+    virtual ~TickListener() = default;
+
+    // OnTick is run on the task queue provided by OnTickTaskQueue each time the
+    // metronome ticks.
+    virtual void OnTick() = 0;
+
+    // The task queue that OnTick will run on. Must not be null.
+    virtual TaskQueueBase* OnTickTaskQueue() = 0;
+  };
+
+  virtual ~Metronome() = default;
+
+  // Adds a tick listener to the metronome. Once this method has returned
+  // OnTick will be invoked on each metronome tick. A listener may
+  // only be added to the metronome once.
+  virtual void AddListener(TickListener* listener) = 0;
+
+  // Removes the tick listener from the metronome. Once this method has returned
+  // OnTick will never be called again. This method must not be called from
+  // within OnTick.
+  virtual void RemoveListener(TickListener* listener) = 0;
+
+  // Returns the current tick period of the metronome.
+  virtual TimeDelta TickPeriod() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_METRONOME_METRONOME_H_

api/metronome/test/BUILD.gn

@@ -0,0 +1,30 @@
+# Copyright (c) 2022 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")
+
+rtc_library("fake_metronome") {
+  testonly = true
+  sources = [
+    "fake_metronome.cc",
+    "fake_metronome.h",
+  ]
+  deps = [
+    "..:metronome",
+    "../..:priority",
+    "../..:sequence_checker",
+    "../../../rtc_base:macromagic",
+    "../../../rtc_base:rtc_event",
+    "../../../rtc_base:rtc_task_queue",
+    "../../../rtc_base/synchronization:mutex",
+    "../../../rtc_base/task_utils:repeating_task",
+    "../../../rtc_base/task_utils:to_queued_task",
+    "../../task_queue",
+    "../../units:time_delta",
+  ]
+}

api/metronome/test/fake_metronome.cc

@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2022 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.
+ */
+
+#include "api/metronome/test/fake_metronome.h"
+
+#include "api/priority.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/event.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/task_utils/to_queued_task.h"
+
+namespace webrtc::test {
+
+ForcedTickMetronome::ForcedTickMetronome(TimeDelta tick_period)
+    : tick_period_(tick_period) {}
+
+void ForcedTickMetronome::AddListener(TickListener* listener) {
+  listeners_.insert(listener);
+}
+
+void ForcedTickMetronome::RemoveListener(TickListener* listener) {
+  listeners_.erase(listener);
+}
+
+TimeDelta ForcedTickMetronome::TickPeriod() const {
+  return tick_period_;
+}
+
+size_t ForcedTickMetronome::NumListeners() {
+  return listeners_.size();
+}
+
+void ForcedTickMetronome::Tick() {
+  for (auto* listener : listeners_) {
+    listener->OnTickTaskQueue()->PostTask(
+        ToQueuedTask([listener] { listener->OnTick(); }));
+  }
+}
+
+FakeMetronome::FakeMetronome(TaskQueueFactory* factory, TimeDelta tick_period)
+    : tick_period_(tick_period),
+      queue_(factory->CreateTaskQueue("MetronomeQueue",
+                                      TaskQueueFactory::Priority::HIGH)) {}
+
+FakeMetronome::~FakeMetronome() {
+  RTC_DCHECK(listeners_.empty());
+}
+
+void FakeMetronome::AddListener(TickListener* listener) {
+  MutexLock lock(&mutex_);
+  listeners_.insert(listener);
+  if (!started_) {
+    tick_task_ = RepeatingTaskHandle::Start(queue_.Get(), [this] {
+      MutexLock lock(&mutex_);
+      // Stop if empty.
+      if (listeners_.empty())
+        return TimeDelta::PlusInfinity();
+      for (auto* listener : listeners_) {
+        listener->OnTickTaskQueue()->PostTask(
+            ToQueuedTask([listener] { listener->OnTick(); }));
+      }
+      return tick_period_;
+    });
+    started_ = true;
+  }
+}
+
+void FakeMetronome::RemoveListener(TickListener* listener) {
+  MutexLock lock(&mutex_);
+  listeners_.erase(listener);
+}
+
+void FakeMetronome::Stop() {
+  MutexLock lock(&mutex_);
+  RTC_DCHECK(listeners_.empty());
+  if (started_)
+    queue_.PostTask([this] { tick_task_.Stop(); });
+}
+
+TimeDelta FakeMetronome::TickPeriod() const {
+  return tick_period_;
+}
+
+}  // namespace webrtc::test

api/metronome/test/fake_metronome.h

@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2022 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.
+ */
+
+#ifndef API_METRONOME_TEST_FAKE_METRONOME_H_
+#define API_METRONOME_TEST_FAKE_METRONOME_H_
+
+#include <memory>
+#include <set>
+
+#include "api/metronome/metronome.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc::test {
+
+// ForcedTickMetronome is a Metronome that ticks when `Tick()` is invoked.
+// The constructor argument `tick_period` returned in `TickPeriod()`.
+class ForcedTickMetronome : public Metronome {
+ public:
+  explicit ForcedTickMetronome(TimeDelta tick_period);
+
+  // Forces all TickListeners to run `OnTick`.
+  void Tick();
+  size_t NumListeners();
+
+  // Metronome implementation.
+  void AddListener(TickListener* listener) override;
+  void RemoveListener(TickListener* listener) override;
+  TimeDelta TickPeriod() const override;
+
+ private:
+  const TimeDelta tick_period_;
+  std::set<TickListener*> listeners_;
+};
+
+// FakeMetronome is a metronome that ticks based on a repeating task at the
+// `tick_period` provided in the constructor. It is designed for use with
+// simulated task queues for unit tests.
+//
+// `Stop()` must be called before destruction, as it cancels the metronome tick
+// on the proper task queue.
+class FakeMetronome : public Metronome {
+ public:
+  FakeMetronome(TaskQueueFactory* factory, TimeDelta tick_period);
+  ~FakeMetronome() override;
+
+  // Metronome implementation.
+  void AddListener(TickListener* listener) override;
+  void RemoveListener(TickListener* listener) override;
+  TimeDelta TickPeriod() const override;
+
+  void Stop();
+
+ private:
+  const TimeDelta tick_period_;
+  RepeatingTaskHandle tick_task_;
+  bool started_ RTC_GUARDED_BY(mutex_) = false;
+  std::set<TickListener*> listeners_ RTC_GUARDED_BY(mutex_);
+  Mutex mutex_;
+  rtc::TaskQueue queue_;
+};
+
+}  // namespace webrtc::test
+
+#endif  // API_METRONOME_TEST_FAKE_METRONOME_H_

api/neteq/neteq.cc

@@ -30,8 +30,7 @@ std::string NetEq::Config::ToString() const {
      << ", min_delay_ms=" << min_delay_ms << ", enable_fast_accelerate="
      << (enable_fast_accelerate ? "true" : "false")
      << ", enable_muted_state=" << (enable_muted_state ? "true" : "false")
-     << ", enable_rtx_handling=" << (enable_rtx_handling ? "true" : "false")
+     << ", enable_rtx_handling=" << (enable_rtx_handling ? "true" : "false");
-     << ", extra_output_delay_ms=" << extra_output_delay_ms;
   return ss.str();
 }
 

api/neteq/neteq.h

@@ -136,10 +136,6 @@ class NetEq {
     bool enable_rtx_handling = false;
     absl::optional<AudioCodecPairId> codec_pair_id;
     bool for_test_no_time_stretching = false;  // Use only for testing.
-    // Adds extra delay to the output of NetEq, without affecting jitter or
-    // loss behavior. This is mainly for testing. Value must be a non-negative
-    // multiple of 10 ms.
-    int extra_output_delay_ms = 0;
   };
 
   enum ReturnCodes { kOK = 0, kFail = -1 };
@@ -183,14 +179,6 @@ class NetEq {
     SdpAudioFormat sdp_format;
   };
 
-  // Creates a new NetEq object, with parameters set in `config`. The `config`
-  // object will only have to be valid for the duration of the call to this
-  // method.
-  static NetEq* Create(
-      const NetEq::Config& config,
-      Clock* clock,
-      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
-
   virtual ~NetEq() {}
 
   // Inserts a new packet into NetEq.

api/notifier.h

@@ -14,7 +14,9 @@
 #include <list>
 
 #include "api/media_stream_interface.h"
+#include "api/sequence_checker.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/system/no_unique_address.h"
 
 namespace webrtc {
 
@@ -23,14 +25,16 @@ namespace webrtc {
 template <class T>
 class Notifier : public T {
  public:
-  Notifier() {}
+  Notifier() { sequence_checker_.Detach(); }
 
   virtual void RegisterObserver(ObserverInterface* observer) {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
     RTC_DCHECK(observer != nullptr);
     observers_.push_back(observer);
   }
 
   virtual void UnregisterObserver(ObserverInterface* observer) {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
     for (std::list<ObserverInterface*>::iterator it = observers_.begin();
          it != observers_.end(); it++) {
       if (*it == observer) {
@@ -41,6 +45,7 @@ class Notifier : public T {
   }
 
   void FireOnChanged() {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
     // Copy the list of observers to avoid a crash if the observer object
     // unregisters as a result of the OnChanged() call. If the same list is used
     // UnregisterObserver will affect the list make the iterator invalid.
@@ -52,7 +57,10 @@ class Notifier : public T {
   }
 
  protected:
-  std::list<ObserverInterface*> observers_;
+  std::list<ObserverInterface*> observers_ RTC_GUARDED_BY(sequence_checker_);
+
+ private:
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
 };
 
 }  // namespace webrtc

api/packet_socket_factory.h

@@ -11,9 +11,12 @@
 #ifndef API_PACKET_SOCKET_FACTORY_H_
 #define API_PACKET_SOCKET_FACTORY_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
+#include "api/async_dns_resolver.h"
+#include "api/wrapping_async_dns_resolver.h"
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/proxy_info.h"
 #include "rtc_base/system/rtc_export.h"
@@ -56,7 +59,7 @@ class RTC_EXPORT PacketSocketFactory {
   virtual AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address,
                                              uint16_t min_port,
                                              uint16_t max_port) = 0;
-  virtual AsyncPacketSocket* CreateServerTcpSocket(
+  virtual AsyncListenSocket* CreateServerTcpSocket(
       const SocketAddress& local_address,
       uint16_t min_port,
       uint16_t max_port,
@@ -69,7 +72,23 @@ class RTC_EXPORT PacketSocketFactory {
       const std::string& user_agent,
       const PacketSocketTcpOptions& tcp_options) = 0;
 
-  virtual AsyncResolverInterface* CreateAsyncResolver() = 0;
+  // The AsyncResolverInterface is deprecated; users are encouraged
+  // to switch to the AsyncDnsResolverInterface.
+  // TODO(bugs.webrtc.org/12598): Remove once all downstream users
+  // are converted.
+  virtual AsyncResolverInterface* CreateAsyncResolver() {
+    // Default implementation, so that downstream users can remove this
+    // immediately after changing to CreateAsyncDnsResolver
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+
+  virtual std::unique_ptr<webrtc::AsyncDnsResolverInterface>
+  CreateAsyncDnsResolver() {
+    // Default implementation, to aid in transition to AsyncDnsResolverInterface
+    return std::make_unique<webrtc::WrappingAsyncDnsResolver>(
+        CreateAsyncResolver());
+  }
 
  private:
   PacketSocketFactory(const PacketSocketFactory&) = delete;

api/peer_connection_interface.cc

@@ -41,12 +41,6 @@ PeerConnectionInterface::RTCConfiguration::RTCConfiguration(
 
 PeerConnectionInterface::RTCConfiguration::~RTCConfiguration() = default;
 
-RTCError PeerConnectionInterface::RemoveTrackNew(
-    rtc::scoped_refptr<RtpSenderInterface> sender) {
-  return RTCError(RemoveTrack(sender) ? RTCErrorType::NONE
-                                      : RTCErrorType::INTERNAL_ERROR);
-}
-
 RTCError PeerConnectionInterface::SetConfiguration(
     const PeerConnectionInterface::RTCConfiguration& config) {
   return RTCError();

api/peer_connection_interface.h

@@ -76,6 +76,7 @@
 #include <vector>
 
 #include "absl/base/attributes.h"
+#include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
 #include "api/adaptation/resource.h"
 #include "api/async_dns_resolver.h"
@@ -95,6 +96,7 @@
 #include "api/jsep.h"
 #include "api/media_stream_interface.h"
 #include "api/media_types.h"
+#include "api/metronome/metronome.h"
 #include "api/neteq/neteq_factory.h"
 #include "api/network_state_predictor.h"
 #include "api/packet_socket_factory.h"
@@ -171,7 +173,12 @@ class StatsObserver : public rtc::RefCountInterface {
   ~StatsObserver() override = default;
 };
 
-enum class SdpSemantics { kPlanB, kUnifiedPlan };
+enum class SdpSemantics {
+  // TODO(https://crbug.com/webrtc/13528): Remove support for kPlanB.
+  kPlanB_DEPRECATED,
+  kPlanB [[deprecated]] = kPlanB_DEPRECATED,
+  kUnifiedPlan,
+};
 
 class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
  public:
@@ -184,6 +191,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     kHaveRemotePrAnswer,
     kClosed,
   };
+  static constexpr absl::string_view AsString(SignalingState);
 
   // See https://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate
   enum IceGatheringState {
@@ -191,6 +199,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     kIceGatheringGathering,
     kIceGatheringComplete
   };
+  static constexpr absl::string_view AsString(IceGatheringState state);
 
   // See https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate
   enum class PeerConnectionState {
@@ -201,6 +210,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     kFailed,
     kClosed,
   };
+  static constexpr absl::string_view AsString(PeerConnectionState state);
 
   // See https://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate
   enum IceConnectionState {
@@ -213,6 +223,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     kIceConnectionClosed,
     kIceConnectionMax,
   };
+  static constexpr absl::string_view AsString(IceConnectionState state);
 
   // TLS certificate policy.
   enum TlsCertPolicy {
@@ -294,6 +305,13 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
 
   enum ContinualGatheringPolicy { GATHER_ONCE, GATHER_CONTINUALLY };
 
+  struct PortAllocatorConfig {
+    // For min_port and max_port, 0 means not specified.
+    int min_port = 0;
+    int max_port = 0;
+    uint32_t flags = 0;  // Same as kDefaultPortAllocatorFlags.
+  };
+
   enum class RTCConfigurationType {
     // A configuration that is safer to use, despite not having the best
     // performance. Currently this is the default configuration.
@@ -371,6 +389,18 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
           video_rtcp_report_interval_ms;
     }
 
+    // Settings for the port allcoator. Applied only if the port allocator is
+    // created by PeerConnectionFactory, not if it is injected with
+    // PeerConnectionDependencies
+    int min_port() const { return port_allocator_config.min_port; }
+    void set_min_port(int port) { port_allocator_config.min_port = port; }
+    int max_port() const { return port_allocator_config.max_port; }
+    void set_max_port(int port) { port_allocator_config.max_port = port; }
+    uint32_t port_allocator_flags() { return port_allocator_config.flags; }
+    void set_port_allocator_flags(uint32_t flags) {
+      port_allocator_config.flags = flags;
+    }
+
     static const int kUndefined = -1;
     // Default maximum number of packets in the audio jitter buffer.
     static const int kAudioJitterBufferMaxPackets = 200;
@@ -598,28 +628,27 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     // cost.
     absl::optional<rtc::AdapterType> network_preference;
 
-    // Configure the SDP semantics used by this PeerConnection. Note that the
+    // Configure the SDP semantics used by this PeerConnection. By default, this
-    // WebRTC 1.0 specification requires kUnifiedPlan semantics. The
+    // is Unified Plan which is compliant to the WebRTC 1.0 specification. It is
-    // RtpTransceiver API is only available with kUnifiedPlan semantics.
+    // possible to overrwite this to the deprecated Plan B SDP format, but note
+    // that kPlanB will be deleted at some future date, see
+    // https://crbug.com/webrtc/13528.
     //
-    // kPlanB will cause PeerConnection to create offers and answers with at
+    // kUnifiedPlan will cause the PeerConnection to create offers and answers
+    // with multiple m= sections where each m= section maps to one RtpSender and
+    // one RtpReceiver (an RtpTransceiver), either both audio or both video.
+    // This will also cause the PeerConnection to ignore all but the first
+    // a=ssrc lines that form a Plan B streams (if the PeerConnection is given
+    // Plan B SDP to process).
+    //
+    // kPlanB will cause the PeerConnection to create offers and answers with at
     // most one audio and one video m= section with multiple RtpSenders and
     // RtpReceivers specified as multiple a=ssrc lines within the section. This
     // will also cause PeerConnection to ignore all but the first m= section of
-    // the same media type.
+    // the same media type (if the PeerConnection is given Unified Plan SDP to
-    //
+    // process).
-    // kUnifiedPlan will cause PeerConnection to create offers and answers with
+    // RingRTC Change to use "Plan B"
-    // multiple m= sections where each m= section maps to one RtpSender and one
+    SdpSemantics sdp_semantics = SdpSemantics::kPlanB_DEPRECATED;
-    // RtpReceiver (an RtpTransceiver), either both audio or both video. This
-    // will also cause PeerConnection to ignore all but the first a=ssrc lines
-    // that form a Plan B stream.
-    //
-    // For users who wish to send multiple audio/video streams and need to stay
-    // interoperable with legacy WebRTC implementations or use legacy APIs,
-    // specify kPlanB.
-    //
-    // For all other users, specify kUnifiedPlan.
-    SdpSemantics sdp_semantics = SdpSemantics::kPlanB;
 
     // TODO(bugs.webrtc.org/9891) - Move to crypto_options or remove.
     // Actively reset the SRTP parameters whenever the DTLS transports
@@ -669,6 +698,8 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     // VPN (in case webrtc fails to auto detect them).
     std::vector<rtc::NetworkMask> vpn_list;
 
+    PortAllocatorConfig port_allocator_config;
+
     //
     // Don't forget to update operator== if adding something.
     //
@@ -780,23 +811,25 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
       rtc::scoped_refptr<MediaStreamTrackInterface> track,
       const std::vector<std::string>& stream_ids) = 0;
 
-  // Remove an RtpSender from this PeerConnection.
+  // Removes the connection between a MediaStreamTrack and the PeerConnection.
-  // Returns true on success.
+  // Stops sending on the RtpSender and marks the
-  // TODO(steveanton): Replace with signature that returns RTCError.
-  virtual bool RemoveTrack(RtpSenderInterface* sender) = 0;
-
-  // Plan B semantics: Removes the RtpSender from this PeerConnection.
-  // Unified Plan semantics: Stop sending on the RtpSender and mark the
   // corresponding RtpTransceiver direction as no longer sending.
+  // https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-removetrack
   //
   // Errors:
   // - INVALID_PARAMETER: `sender` is null or (Plan B only) the sender is not
   //       associated with this PeerConnection.
   // - INVALID_STATE: PeerConnection is closed.
+  //
+  // Plan B semantics: Removes the RtpSender from this PeerConnection.
+  //
   // TODO(bugs.webrtc.org/9534): Rename to RemoveTrack once the other signature
-  // is removed.
+  // is removed; remove default implementation once upstream is updated.
-  virtual RTCError RemoveTrackNew(
+  virtual RTCError RemoveTrackOrError(
-      rtc::scoped_refptr<RtpSenderInterface> sender);
+      rtc::scoped_refptr<RtpSenderInterface> sender) {
+    RTC_CHECK_NOTREACHED();
+    return RTCError();
+  }
 
   // AddTransceiver creates a new RtpTransceiver and adds it to the set of
   // transceivers. Adding a transceiver will cause future calls to CreateOffer
@@ -1311,14 +1344,6 @@ class PeerConnectionObserver : public RtpPacketSinkInterface {
   // A new ICE candidate has been gathered.
   virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
 
-  // Gathering of an ICE candidate failed.
-  // See https://w3c.github.io/webrtc-pc/#event-icecandidateerror
-  // `host_candidate` is a stringified socket address.
-  virtual void OnIceCandidateError(const std::string& host_candidate,
-                                   const std::string& url,
-                                   int error_code,
-                                   const std::string& error_text) {}
-
   // Gathering of an ICE candidate failed.
   // See https://w3c.github.io/webrtc-pc/#event-icecandidateerror
   virtual void OnIceCandidateError(const std::string& address,
@@ -1443,6 +1468,7 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final {
   rtc::Thread* network_thread = nullptr;
   rtc::Thread* worker_thread = nullptr;
   rtc::Thread* signaling_thread = nullptr;
+  rtc::SocketFactory* socket_factory = nullptr;
   std::unique_ptr<TaskQueueFactory> task_queue_factory;
   std::unique_ptr<cricket::MediaEngineInterface> media_engine;
   std::unique_ptr<CallFactoryInterface> call_factory;
@@ -1460,6 +1486,7 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final {
   std::unique_ptr<WebRtcKeyValueConfig> trials;
   std::unique_ptr<RtpTransportControllerSendFactoryInterface>
       transport_controller_send_factory;
+  std::unique_ptr<Metronome> metronome;
 };
 
 // PeerConnectionFactoryInterface is the factory interface used for creating
@@ -1616,6 +1643,93 @@ RTC_EXPORT rtc::scoped_refptr<PeerConnectionFactoryInterface>
 CreateModularPeerConnectionFactory(
     PeerConnectionFactoryDependencies dependencies);
 
+// https://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate
+inline constexpr absl::string_view PeerConnectionInterface::AsString(
+    SignalingState state) {
+  switch (state) {
+    case SignalingState::kStable:
+      return "stable";
+    case SignalingState::kHaveLocalOffer:
+      return "have-local-offer";
+    case SignalingState::kHaveLocalPrAnswer:
+      return "have-local-pranswer";
+    case SignalingState::kHaveRemoteOffer:
+      return "have-remote-offer";
+    case SignalingState::kHaveRemotePrAnswer:
+      return "have-remote-pranswer";
+    case SignalingState::kClosed:
+      return "closed";
+  }
+  // This cannot happen.
+  // Not using "RTC_CHECK_NOTREACHED()" because AsString() is constexpr.
+  return "";
+}
+
+// https://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate
+inline constexpr absl::string_view PeerConnectionInterface::AsString(
+    IceGatheringState state) {
+  switch (state) {
+    case IceGatheringState::kIceGatheringNew:
+      return "new";
+    case IceGatheringState::kIceGatheringGathering:
+      return "gathering";
+    case IceGatheringState::kIceGatheringComplete:
+      return "complete";
+  }
+  // This cannot happen.
+  // Not using "RTC_CHECK_NOTREACHED()" because AsString() is constexpr.
+  return "";
+}
+
+// https://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate
+inline constexpr absl::string_view PeerConnectionInterface::AsString(
+    PeerConnectionState state) {
+  switch (state) {
+    case PeerConnectionState::kNew:
+      return "new";
+    case PeerConnectionState::kConnecting:
+      return "connecting";
+    case PeerConnectionState::kConnected:
+      return "connected";
+    case PeerConnectionState::kDisconnected:
+      return "disconnected";
+    case PeerConnectionState::kFailed:
+      return "failed";
+    case PeerConnectionState::kClosed:
+      return "closed";
+  }
+  // This cannot happen.
+  // Not using "RTC_CHECK_NOTREACHED()" because AsString() is constexpr.
+  return "";
+}
+
+inline constexpr absl::string_view PeerConnectionInterface::AsString(
+    IceConnectionState state) {
+  switch (state) {
+    case kIceConnectionNew:
+      return "new";
+    case kIceConnectionChecking:
+      return "checking";
+    case kIceConnectionConnected:
+      return "connected";
+    case kIceConnectionCompleted:
+      return "completed";
+    case kIceConnectionFailed:
+      return "failed";
+    case kIceConnectionDisconnected:
+      return "disconnected";
+    case kIceConnectionClosed:
+      return "closed";
+    case kIceConnectionMax:
+      // This cannot happen.
+      // Not using "RTC_CHECK_NOTREACHED()" because AsString() is constexpr.
+      return "";
+  }
+  // This cannot happen.
+  // Not using "RTC_CHECK_NOTREACHED()" because AsString() is constexpr.
+  return "";
+}
+
 }  // namespace webrtc
 
 #endif  // API_PEER_CONNECTION_INTERFACE_H_

api/ref_counted_base.h

@@ -12,7 +12,6 @@
 
 #include <type_traits>
 
-#include "rtc_base/constructor_magic.h"
 #include "rtc_base/ref_counter.h"
 
 namespace rtc {
@@ -21,6 +20,9 @@ class RefCountedBase {
  public:
   RefCountedBase() = default;
 
+  RefCountedBase(const RefCountedBase&) = delete;
+  RefCountedBase& operator=(const RefCountedBase&) = delete;
+
   void AddRef() const { ref_count_.IncRef(); }
   RefCountReleaseStatus Release() const {
     const auto status = ref_count_.DecRef();
@@ -39,8 +41,6 @@ class RefCountedBase {
 
  private:
   mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RefCountedBase);
 };
 
 // Template based version of `RefCountedBase` for simple implementations that do
@@ -61,6 +61,9 @@ class RefCountedNonVirtual {
  public:
   RefCountedNonVirtual() = default;
 
+  RefCountedNonVirtual(const RefCountedNonVirtual&) = delete;
+  RefCountedNonVirtual& operator=(const RefCountedNonVirtual&) = delete;
+
   void AddRef() const { ref_count_.IncRef(); }
   RefCountReleaseStatus Release() const {
     // If you run into this assert, T has virtual methods. There are two
@@ -88,8 +91,6 @@ class RefCountedNonVirtual {
 
  private:
   mutable webrtc::webrtc_impl::RefCounter ref_count_{0};
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(RefCountedNonVirtual);
 };
 
 }  // namespace rtc

api/rtc_error.h

@@ -181,8 +181,8 @@ inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
 #define LOG_AND_RETURN_ERROR_EX(type, message, severity)                     \
   {                                                                          \
     RTC_DCHECK(type != RTCErrorType::NONE);                                  \
-    RTC_LOG(severity) << message << " (" << ToString(type) << ")"; \
+    RTC_LOG(severity) << message << " (" << ::webrtc::ToString(type) << ")"; \
-    return webrtc::RTCError(type, message);                        \
+    return ::webrtc::RTCError(type, message);                                \
   }
 
 #define LOG_AND_RETURN_ERROR(type, message) \
@@ -244,7 +244,7 @@ class RTCErrorOr {
   //
   // REQUIRES: !error.ok(). This requirement is DCHECKed.
   RTCErrorOr(RTCError&& error) : error_(std::move(error)) {  // NOLINT
-    RTC_DCHECK(!error.ok());
+    RTC_DCHECK(!error_.ok());
   }
 
   // Constructs a new RTCErrorOr with the given value. After calling this