Update to WebRTC 4389 (e7d9f74)

Contains changes for M86 (4240), M87 (4280), M88 (4324), and M89 (4389).

.gn

@@ -21,7 +21,6 @@ secondary_source = "//build/secondary/"
 # their includes checked for proper dependencies when you run either
 # "gn check" or "gn gen --check".
 check_targets = [
-  ":webrtc_common",
   "//api/*",
   "//audio/*",
   "//backup/*",
@@ -60,6 +59,7 @@ default_args = {
   # component builds.
   is_component_build = false
 
+  # RingRTC change to control MacOS/iOS targets
   mac_sdk_min = "10.12.1"
 
   ios_deployment_target = "11.0"
@@ -76,6 +76,7 @@ default_args = {
   # for unittests, it can be disabled (see third_party/libyuv/BUILD.gn)
   libyuv_use_gflags = false
 
+  # RingRTC change to disable AV1
   enable_libaom = false
 
   gtest_enable_absl_printers = true

.vpython

@@ -66,3 +66,11 @@ 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

@@ -1,110 +1,128 @@
-# Names should be added to this file like so:
+# Names should be added to this file with this pattern:
-# Name or Organization <email address>
+#
+# For individuals:
+#   Name <email address>
+#
+# For organizations:
+#   Organization <fnmatch pattern>
+#
+# See python fnmatch module documentation for more information.
+#
+# Please keep the list sorted.
 
+# BEGIN individuals section.
 Adam Fedor <adam.fedor@gmail.com>
 Akshay Shah <meetakshay99@gmail.com>
-Alex Henrie <alexhenrie24@gmail.com>
 Alexander Brauckmann <a.brauckmann@gmail.com>
 Alexandre Gouaillard <agouaillard@gmail.com>
+Alex Henrie <alexhenrie24@gmail.com>
 Andrew MacDonald <andrew@webrtc.org>
 Andrey Efremov <yoklmnprst@ya.ru>
+Andrew Johnson <ajohnson@draster.com>
 Anil Kumar <an1kumar@gmail.com>
 Ben Strong <bstrong@gmail.com>
 Bob Withers <bwit@pobox.com>
 Bridger Maxwell <bridgeyman@gmail.com>
-Chris Tserng <tserng@amazon.com>
 Christophe Dumez <ch.dumez@samsung.com>
+Chris Tserng <tserng@amazon.com>
 Cody Barnes <conceptgenesis@gmail.com>
 Colin Plumb
 Cyril Lashkevich <notorca@gmail.com>
+CZ Theng <cz.theng@gmail.com>
+Danail Kirov <dkirovbroadsoft@gmail.com>
 David Porter <david@porter.me>
 Dax Booysen <dax@younow.com>
-Danail Kirov <dkirovbroadsoft@gmail.com>
+Dennis Angelo <dennis.angelo@gmail.com>
+Dharmesh Chauhan <dharmesh.r.chauhan@gmail.com>
 Dirk-Jan C. Binnema <djcb@djcbsoftware.nl>
 Dmitry Lizin <sdkdimon@gmail.com>
+Eike Rathke <erathke@redhat.com>
 Eric Rescorla, RTFM Inc. <ekr@rtfm.com>
 Frederik Riedel, Frogg GmbH <frederik.riedel@frogg.io>
 Giji Gangadharan <giji.g@samsung.com>
 Graham Yoakum <gyoakum@skobalt.com>
 Gustavo Garcia <gustavogb@gmail.com>
+Hans Knoechel <hans@hans-knoechel.de>
 Hugues Ekra <hekra01@gmail.com>
 Jake Hilton <jakehilton@gmail.com>
 James H. Brown <jbrown@burgoyne.com>
+Jan Grulich <grulja@gmail.com>
 Jan Kalab <pitlicek@gmail.com>
 Jens Nielsen <jens.nielsen@berotec.se>
 Jiawei Ou <jiawei.ou@gmail.com>
 Jie Mao <maojie0924@gmail.com>
+Jiwon Kim <jwkim0000@gmail.com>
 Jose Antonio Olivera Ortega <josea.olivera@gmail.com>
 Kiran Thind <kiran.thind@gmail.com>
+Korniltsev Anatoly <korniltsev.anatoly@gmail.com>
+Lennart Grahl <lennart.grahl@gmail.com>
 Luke Weber <luke.weber@gmail.com>
 Maksim Khobat <maksimkhobat@gmail.com>
 Mallikarjuna Rao V <vm.arjun@samsung.com>
 Manish Jethani <manish.jethani@gmail.com>
 Martin Storsjo <martin@martin.st>
 Matthias Liebig <matthias.gcode@gmail.com>
+Maxim Pavlov <pavllovmax@gmail.com>
 Maxim Potapov <vopatop.skam@gmail.com>
 Michael Iedema <michael@kapsulate.com>
+Michel Promonet <michel.promonet.1@gmail.com>
+Miguel Paris <mparisdiaz@gmail.com>
 Mike Gilbert <floppymaster@gmail.com>
+Min Wang <mingewang@gmail.com>
 Mo Zanaty <mzanaty@cisco.com>
 Pali Rohar
 Paul Kapustin <pkapustin@gmail.com>
-Philipp Hancke <philipp.hancke@googlemail.com>
 Peng Yu <yupeng323@gmail.com>
+Philipp Hancke <philipp.hancke@googlemail.com>
+Piasy Xu <xz4215@gmail.com>
 Rafael Lopez Diez <rafalopezdiez@gmail.com>
 Ralph Giles <giles@ghostscript.com>
+Raman Budny <budnyjj@gmail.com>
+Ramprakash Jelari <ennajelari@gmail.com>
 Riku Voipio <riku.voipio@linaro.org>
 Robert Bares <robert@bares.me>
 Robert Nagy <robert.nagy@gmail.com>
 Ryan Yoakum <ryoakum@skobalt.com>
-Satender Saroha <ssaroha@yahoo.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>
 Silviu Caragea <silviu.cpp@gmail.com>
 Stefan Gula <steweg@gmail.com>
+Stephan Hartmann <stha09@googlemail.com>
 Steve Reid <sreid@sea-to-sky.net>
 Tarun Chawla <trnkumarchawla@gmail.com>
+Todd Wong <todd.wong.ndq@gmail.com>
+Tomas Popela <tomas.popela@gmail.com>
 Trevor Hayes <trevor.axiom@gmail.com>
 Uladzislau Susha <landby@gmail.com>
-Vladimir Beloborodov <VladimirTechMan@gmail.com>
 Vicken Simonian <vsimon@gmail.com>
 Victor Costan <costan@gmail.com>
+Vladimir Beloborodov <VladimirTechMan@gmail.com>
 Xiaohong Xu <freemine@yeah.net>
 Xiaolei Yu <dreifachstein@gmail.com>
 Yura Yaroshevich <yura.yaroshevich@gmail.com>
 Yuriy Pavlyshak <yuriy@appear.in>
-Hans Knoechel <hans@hans-knoechel.de>
-Korniltsev Anatoly <korniltsev.anatoly@gmail.com>
-Todd Wong <todd.wong.ndq@gmail.com>
-Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
-Maxim Pavlov <pavllovmax@gmail.com>
 Yusuke Suzuki <utatane.tea@gmail.com>
-Piasy Xu <xz4215@gmail.com>
+# END individuals section.
-Tomas Popela <tomas.popela@gmail.com>
-Jan Grulich <grulja@gmail.com>
-Jiwon Kim <jwkim0000@gmail.com>
-Eike Rathke <erathke@redhat.com>
-Michel Promonet <michel.promonet.1@gmail.com>
-Min Wang <mingewang@gmail.com>
-Ramprakash Jelari <ennajelari@gmail.com>
-CZ Theng <cz.theng@gmail.com>
-Miguel Paris <mparisdiaz@gmail.com>
-Raman Budny <budnyjj@gmail.com>
-Stephan Hartmann <stha09@googlemail.com>
 
-&yet LLC <*@andyet.com>
+# BEGIN organizations section.
-8x8 Inc. <*@sip-communicator.org>
 8x8 Inc. <*@8x8.com>
+8x8 Inc. <*@sip-communicator.org>
 Agora IO <*@agora.io>
 ARM Holdings <*@arm.com>
 BroadSoft Inc. <*@broadsoft.com>
+CoSMo Software Consulting, Pte Ltd <*@cosmosoftware.io>
 Facebook Inc. <*@fb.com>
 Google Inc. <*@google.com>
+Highfive, Inc. <*@highfive.com>
 HyperConnect Inc. <*@hpcnt.com>
-Life On Air Inc. <*@lifeonair.com>
 Intel Corporation <*@intel.com>
+Life On Air Inc. <*@lifeonair.com>
 Microsoft Corporation <*@microsoft.com>
 MIPS Technologies <*@mips.com>
 Mozilla Foundation <*@mozilla.com>
+Netgem S.A. <*@netgem.com>
 NVIDIA Corporation <*@nvidia.com>
 Opera Software ASA <*@opera.com>
 Optical Tone Ltd <*@opticaltone.com>
@@ -112,20 +130,19 @@ Pengutronix e.K. <*@pengutronix.de>
 RingCentral, Inc. <*@ringcentral.com>
 Signal Messenger, LLC <*@signal.org>
 Sinch AB <*@sinch.com>
-Signal Messenger, LLC <*signal.org>
 struktur AG <*@struktur.de>
 Telenor Digital AS <*@telenor.com>
 Temasys Communications <*@temasys.io>
 The Chromium Authors <*@chromium.org>
 The WebRTC Authors <*@webrtc.org>
+Threema GmbH <*@threema.ch>
+Tuple, LLC <*@tuple.app>
 Twilio, Inc. <*@twilio.com>
+Vewd Software AS <*@vewd.com>
+Videona Socialmedia <*@videona.com>
 Videxio AS <*@videxio.com>
 Vidyo, Inc. <*@vidyo.com>
 Vonage Holdings Corp. <*@vonage.com>
 Wire Swiss GmbH <*@wire.com>
-Vewd Software AS <*@vewd.com>
+&yet LLC <*@andyet.com>
-Highfive, Inc. <*@highfive.com>
+# END organizations section.
-CoSMo Software Consulting, Pte Ltd <*@cosmosoftware.io>
-Tuple, LLC <*@tuple.app>
-Videona Socialmedia <*@videona.com>
-Threema GmbH <*@threema.ch>

BUILD.gn

@@ -82,12 +82,15 @@ if (!build_with_chromium) {
       }
       if (rtc_enable_protobuf) {
         deps += [
-          # "audio:low_bandwidth_audio_test",
+          "audio:low_bandwidth_audio_test",
           "logging:rtc_event_log_rtp_dump",
           "tools_webrtc/perf:webrtc_dashboard_upload",
         ]
       }
     }
+    if (target_os == "android") {
+      deps += [ "tools_webrtc:binary_version_check" ]
+    }
   }
 }
 
@@ -129,6 +132,10 @@ config("common_inherited_config") {
     defines += [ "RTC_DISABLE_CHECK_MSG" ]
   }
 
+  if (rtc_enable_avx2) {
+    defines += [ "WEBRTC_ENABLE_AVX2" ]
+  }
+
   # Some tests need to declare their own trace event handlers. If this define is
   # not set, the first time TRACE_EVENT_* is called it will store the return
   # value for the current handler in an static variable, so that subsequent
@@ -169,7 +176,7 @@ config("common_inherited_config") {
       "WEBRTC_IOS",
     ]
   }
-  if (is_linux) {
+  if (is_linux || is_chromeos) {
     defines += [ "WEBRTC_LINUX" ]
   }
   if (is_mac) {
@@ -414,7 +421,7 @@ config("common_config") {
 }
 
 config("common_objc") {
-  libs = [ "Foundation.framework" ]
+  frameworks = [ "Foundation.framework" ]
 
   if (rtc_use_metal_rendering) {
     defines = [ "RTC_SUPPORTS_METAL" ]
@@ -436,7 +443,6 @@ if (!build_with_chromium) {
     defines = []
 
     deps = [
-      ":webrtc_common",
       "api:create_peerconnection_factory",
       "api:libjingle_peerconnection_api",
       "api:rtc_error",
@@ -512,15 +518,6 @@ if (!build_with_chromium) {
   }
 }
 
-rtc_source_set("webrtc_common") {
-  # Client code SHOULD NOT USE THIS TARGET, but for now it needs to be public
-  # because there exists client code that uses it.
-  # TODO(bugs.webrtc.org/9808): Move to private visibility as soon as that
-  # client code gets updated.
-  visibility = [ "*" ]
-  sources = [ "common_types.h" ]
-}
-
 if (use_libfuzzer || use_afl) {
   # This target is only here for gn to discover fuzzer build targets under
   # webrtc/test/fuzzers/.
@@ -535,17 +532,18 @@ if (rtc_include_tests) {
     testonly = true
 
     deps = [
-      ":webrtc_common",
       "api:compile_all_headers",
       "api:rtc_api_unittests",
       "api/audio/test:audio_api_unittests",
       "api/audio_codecs/test:audio_codecs_api_unittests",
+      "api/numerics:numerics_unittests",
       "api/transport:stun_unittest",
       "api/video/test:rtc_api_video_unittests",
       "api/video_codecs/test:video_codecs_api_unittests",
       "call:fake_network_pipe_unittests",
       "p2p:libstunprober_unittests",
       "p2p:rtc_p2p_unittests",
+      "rtc_base:callback_list_unittests",
       "rtc_base:rtc_base_approved_unittests",
       "rtc_base:rtc_base_unittests",
       "rtc_base:rtc_json_unittests",
@@ -553,6 +551,7 @@ if (rtc_include_tests) {
       "rtc_base:rtc_operations_chain_unittests",
       "rtc_base:rtc_task_queue_unittests",
       "rtc_base:sigslot_unittest",
+      "rtc_base:untyped_function_unittest",
       "rtc_base:weak_ptr_unittests",
       "rtc_base/experiments:experiments_unittests",
       "rtc_base/synchronization:sequence_checker_unittests",

ENG_REVIEW_OWNERS

@@ -6,6 +6,5 @@
 # review owners to ensure that the added dependency was OK.
 
 danilchap@webrtc.org
-kwiberg@webrtc.org
 mbonadei@webrtc.org
 phoglund@webrtc.org

OWNERS

@@ -1,6 +1,5 @@
 henrika@webrtc.org
 juberti@webrtc.org
-kwiberg@webrtc.org
 mflodman@webrtc.org
 stefan@webrtc.org
 tommi@webrtc.org
@@ -13,10 +12,8 @@ per-file DEPS=*
 per-file pylintrc=phoglund@webrtc.org
 per-file WATCHLISTS=*
 per-file abseil-in-webrtc.md=danilchap@webrtc.org
-per-file abseil-in-webrtc.md=kwiberg@webrtc.org
 per-file abseil-in-webrtc.md=mbonadei@webrtc.org
 per-file style-guide.md=danilchap@webrtc.org
-per-file style-guide.md=kwiberg@webrtc.org
+per-file native-api.md=mbonadei@webrtc.org
-per-file native-api.md=kwiberg@webrtc.org
 
 # COMPONENT: Internals>WebRTC

README.chromium

@@ -1,6 +1,7 @@
 Name: WebRTC
 URL: http://www.webrtc.org
 Version: 90
+CPEPrefix: cpe:/a:webrtc_project:webrtc:90
 License: BSD
 License File: LICENSE
 

README.md

@@ -6,58 +6,21 @@ It currently has the following changes:
 * Various things disabled (RTP header extensions, audio codecs)
 * Various security patches (since the version when the fork branched off)
 
-It began by branching at branch-heads/3987.  Since then, we have merge at the following points:
+See [here][native-dev] for instructions on how to get started
+developing with the native code.
 
-At branch-heads/4044:
+[Authoritative list](native-api.md) of directories that contain the
-* Update DEPS on Chromium, BoringSSL, FFMPEG, libvpx, ...
+native API header files.
-* Added support for "GOOG_PING" to ICE
-  * Send normal STUN binding requests with a special attribute meaning "I support GOOG_PING" until the first response
-  * Send "I support GOOG_PING" in all STUN binding responses (making them slightly bigger)
-  * If the response has an "I support GOOG_PING" attribute, switch to supporting GOOG_PING.  Else, don't
-  * Is supporting GOOG_PING mode and the last ACKed request is the same as the one just sent, send a GOOG_PING
-  * GOOG_PINGs have an HMAC of 4 bytes (instead of 20) and no other attribute (no fingerpint, username, peer reflex priority, etc), which makes them really small
-  * GOOG_PING responses also have an HMAC of 4 bytes
-* Added support for selecting camera by name to Android SDK
-* Added support for degredation preference to Android and iOS SDKs
-* MessageQueue and Thread merged into one class
-* Audio data includes absolute capture timestamps
-* VideoEngine has different send and receive codec capabilities
-* VideoEncoders report their requested resolution alignment 
-* VideoDecoders allow specifying the size of the decoded frame buffer pool
-* VideoSinks can specify a resolution alignment, which is back 
-* Change default behavior of audio sending when ANA is enabled to dynamically adjust the packet overhead calculation according to the network route used by ICE and the size of the RTP headers.
-* Different behavior for transport-cc1 and transport-cc2 when WebRTC-SendSideBwe-WithOverhead is enabled
-* NetEq can take an injectable Clock (which comes from Call::Create)
-* NetEq no longer supports INTER_ARRIVAL_TIME HistogramMode (it appeared to be unused)
-* Removed AEC(1) (as opposed to AEC2 and AEC3)
-* Removed ICE periodic regathering support
-* Improved drawing over cursor over desktop capture
-* Improved support for WebRTC-RtcpLossNotification ("goog-lntf")
 
-At branch-heads/4103:
+### More info
-- Updated SCTP
-- Added AV1 (although we turned it off to reduce attack surface)
-- Header extensions added (turned off to reduce attack surface)
-- Adaptation refactored a lot
-- Various bug fixes (epoll, non-paced audio, candidates generated on 3G<->4G switching)
-- Tweaks to padding
-- Support for "inseartable streams" added (not used at the moment)
 
-At branch-heads/4147:
+ * Official web site: http://www.webrtc.org
-- Update SCTP
+ * Master source code repo: https://webrtc.googlesource.com/src
-- VideoReceiveStream replaced with VideoReceiveStream2
+ * Samples and reference apps: https://github.com/webrtc
-- Tweaks to GoogCC
+ * Mailing list: http://groups.google.com/group/discuss-webrtc
-- More adaptation refactoring
+ * Continuous build: http://build.chromium.org/p/client.webrtc
-- Support for 5G in network stack
+ * [Coding style guide](style-guide.md)
-- Remove SRTP_AES128_CM_SHA1_80
+ * [Code of conduct](CODE_OF_CONDUCT.md)
-- Allow forInjectable APM?
+ * [Reporting bugs](docs/bug-reporting.md)
 
-At branch-heads/4183:
+[native-dev]: https://webrtc.googlesource.com/src/+/refs/heads/master/docs/native-code/index.md
-- Video adapation refactored
-- Refactored audio frame length (ptime) adapation
-- Support for audio RED payload type added
-- Support for audio RED payload type added
-- Frame Marking RTP header extension removed
-- MediaTransportInterface and DatagramTransportInterface removed
-- Switch to using absl::MutexLock
-- RtpRtcp module moved to RtpRtcpInterface + RtpRtcpModule2

WATCHLISTS

@@ -113,8 +113,9 @@
                      'peah@webrtc.org',
                      'saza@webrtc.org'],
     'audio': ['peah@webrtc.org'],
-    'api': ['kwiberg@webrtc.org','peah@webrtc.org'],
+    'api': ['hta@webrtc.org',
-    'base': ['kwiberg@webrtc.org'],
+            'peah@webrtc.org'],
+    'base': ['hta@webrtc.org'],
     'call': ['mflodman@webrtc.org',
              'stefan@webrtc.org'],
     'video': ['mflodman@webrtc.org',
@@ -134,7 +135,6 @@
     'audio_coding': ['alessiob@webrtc.org',
                      'audio-team@agora.io',
                      'henrik.lundin@webrtc.org',
-                     'kwiberg@webrtc.org',
                      'minyue@webrtc.org',
                      'peah@webrtc.org',
                      'saza@webrtc.org'],
@@ -152,7 +152,6 @@
                          'audio-team@agora.io',
                          'fhernqvist@webrtc.org',
                          'henrik.lundin@webrtc.org',
-                         'kwiberg@webrtc.org',
                          'minyue@webrtc.org',
                          'peah@webrtc.org',
                          'saza@webrtc.org'],

abseil-in-webrtc.md

@@ -9,6 +9,17 @@ adds the first use.
 
 [abseil]: https://abseil.io/about/
 
+
+## How to depend on Abseil
+
+For build targets of type `rtc_library`, `rtc_source_set` and
+`rtc_static_library`, dependencies on Abseil need to be listed in `absl_deps`
+instead of `deps`.
+
+This is needed in order to support the Abseil component build in Chromium. In
+that build mode, WebRTC will depend on a monolithic Abseil build target that
+will generate a shared library.
+
 ## **Allowed**
 
 * `absl::InlinedVector`
@@ -63,12 +74,3 @@ has decided if they will change `absl::Span` to match.
 These are optimized for speed, not binary size. Even `StrCat` calls
 with a modest number of arguments can easily add several hundred bytes
 to the binary.
-
-## How to depend on Abseil
-
-For build targets `rtc_library`, `rtc_source_set` and `rtc_static_library`,
-dependencies on Abseil need to be listed in `absl_deps` instead of `deps`.
-
-This is needed in order to support the Abseil component build in Chromium. In
-such build mode, WebRTC will depend on a unique Abseil build target what will
-generate a shared library.

api/BUILD.gn

@@ -55,6 +55,7 @@ if (!build_with_chromium) {
       "audio:audio_mixer_api",
       "audio_codecs:audio_codecs_api",
       "task_queue:default_task_queue_factory",
+      "transport:field_trial_based_config",
       "video_codecs:video_codecs_api",
     ]
   }
@@ -68,7 +69,6 @@ rtc_library("rtp_headers") {
   ]
   deps = [
     ":array_view",
-    "..:webrtc_common",
     "units:timestamp",
     "video:video_rtp_headers",
   ]
@@ -87,7 +87,6 @@ rtc_library("rtp_packet_info") {
     ":refcountedbase",
     ":rtp_headers",
     ":scoped_refptr",
-    "..:webrtc_common",
     "../rtc_base:rtc_base_approved",
     "../rtc_base/system:rtc_export",
   ]
@@ -99,6 +98,7 @@ rtc_library("media_stream_interface") {
   sources = [
     "media_stream_interface.cc",
     "media_stream_interface.h",
+    "media_stream_track.h",
     "notifier.h",
   ]
   deps = [
@@ -127,6 +127,7 @@ rtc_library("libjingle_peerconnection_api") {
     "dtls_transport_interface.cc",
     "dtls_transport_interface.h",
     "dtmf_sender_interface.h",
+    # RingRTC change to add ICE forking
     "ice_gatherer_interface.h",
     "ice_transport_interface.h",
     "jsep.cc",
@@ -150,6 +151,7 @@ rtc_library("libjingle_peerconnection_api") {
     "rtp_transceiver_interface.h",
     "sctp_transport_interface.cc",
     "sctp_transport_interface.h",
+    "set_local_description_observer_interface.h",
     "set_remote_description_observer_interface.h",
     "stats_types.cc",
     "stats_types.h",
@@ -186,6 +188,7 @@ rtc_library("libjingle_peerconnection_api") {
     "transport:bitrate_settings",
     "transport:enums",
     "transport:network_control",
+    "transport:sctp_transport_factory_interface",
     "transport:webrtc_key_value_config",
     "transport/rtp:rtp_source",
     "units:data_rate",
@@ -197,7 +200,6 @@ rtc_library("libjingle_peerconnection_api") {
     # Basically, don't add stuff here. You might break sensitive downstream
     # targets like pnacl. API should not depend on anything outside of this
     # file, really. All these should arguably go away in time.
-    "..:webrtc_common",
     "../media:rtc_media_base",
     "../media:rtc_media_config",
     "../modules/audio_processing:audio_processing_statistics",
@@ -297,9 +299,10 @@ rtc_source_set("video_quality_analyzer_api") {
   ]
 }
 
-rtc_source_set("track_id_stream_label_map") {
+rtc_source_set("track_id_stream_info_map") {
   visibility = [ "*" ]
-  sources = [ "test/track_id_stream_label_map.h" ]
+  sources = [ "test/track_id_stream_info_map.h" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
 }
 
 rtc_source_set("rtp_transceiver_direction") {
@@ -346,7 +349,7 @@ rtc_source_set("audio_quality_analyzer_api") {
 
   deps = [
     ":stats_observer_interface",
-    ":track_id_stream_label_map",
+    ":track_id_stream_info_map",
   ]
 }
 
@@ -378,9 +381,9 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
     ":rtp_parameters",
     ":simulated_network_api",
     ":stats_observer_interface",
+    ":track_id_stream_info_map",
     ":video_quality_analyzer_api",
     "../media:rtc_media_base",
-    "../rtc_base:deprecation",
     "../rtc_base:rtc_base",
     "rtc_event_log",
     "task_queue",
@@ -399,7 +402,10 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
 rtc_source_set("frame_generator_api") {
   visibility = [ "*" ]
   testonly = true
-  sources = [ "test/frame_generator_interface.h" ]
+  sources = [
+    "test/frame_generator_interface.cc",
+    "test/frame_generator_interface.h",
+  ]
 
   deps = [
     ":scoped_refptr",
@@ -464,6 +470,7 @@ if (rtc_include_tests) {
     deps = [
       ":audio_quality_analyzer_api",
       ":peer_connection_quality_test_fixture_api",
+      ":time_controller",
       ":video_quality_analyzer_api",
       "../test/pc/e2e:peerconnection_quality_test",
     ]
@@ -499,8 +506,8 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
     ":peer_connection_quality_test_fixture_api",
     "../rtc_base:checks",
     "../test:fileutils",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_source_set("libjingle_logging_api") {
@@ -578,10 +585,7 @@ rtc_source_set("bitrate_allocation") {
 rtc_source_set("simulated_network_api") {
   visibility = [ "*" ]
   sources = [ "test/simulated_network.h" ]
-  deps = [
+  deps = [ "../rtc_base" ]
-    "../rtc_base",
-    "../rtc_base:criticalsection",
-  ]
   absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
@@ -593,6 +597,7 @@ rtc_source_set("network_emulation_manager_api") {
     "test/network_emulation_manager.h",
   ]
   deps = [
+    ":array_view",
     ":simulated_network_api",
     ":time_controller",
     "../call:simulated_network",
@@ -755,7 +760,7 @@ if (rtc_include_tests) {
       "test/videocodec_test_stats.h",
     ]
     deps = [
-      "..:webrtc_common",
+      "../media:rtc_h264_profile_id",
       "../modules/video_coding:video_codec_interface",
       "../rtc_base:stringutils",
       "video:video_frame_type",
@@ -789,6 +794,17 @@ if (rtc_include_tests) {
     ]
   }
 
+  rtc_source_set("mock_data_channel") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [ "test/mock_data_channel.h" ]
+
+    deps = [
+      ":libjingle_peerconnection_api",
+      "../test:test_support",
+    ]
+  }
+
   rtc_source_set("mock_fec_controller_override") {
     testonly = true
     sources = [ "test/mock_fec_controller_override.h" ]
@@ -829,7 +845,6 @@ if (rtc_include_tests) {
       ":array_view",
       ":libjingle_peerconnection_api",
       ":rtp_parameters",
-      "..:webrtc_common",
       "../rtc_base:checks",
       "../rtc_base:rtc_base_approved",
       "crypto:frame_encryptor_interface",
@@ -846,13 +861,23 @@ if (rtc_include_tests) {
       ":array_view",
       ":libjingle_peerconnection_api",
       ":rtp_parameters",
-      "..:webrtc_common",
       "../rtc_base:checks",
       "../rtc_base:rtc_base_approved",
       "crypto:frame_decryptor_interface",
     ]
   }
 
+  rtc_source_set("mock_media_stream_interface") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [ "test/mock_media_stream_interface.h" ]
+
+    deps = [
+      ":media_stream_interface",
+      "../test:test_support",
+    ]
+  }
+
   rtc_source_set("dummy_peer_connection") {
     visibility = [ "*" ]
     testonly = true
@@ -889,8 +914,10 @@ if (rtc_include_tests) {
   }
 
   rtc_source_set("mock_rtp") {
+    visibility = [ "*" ]
     testonly = true
     sources = [
+      "test/mock_rtp_transceiver.h",
       "test/mock_rtpreceiver.h",
       "test/mock_rtpsender.h",
     ]
@@ -1036,8 +1063,10 @@ if (rtc_include_tests) {
       ":fake_frame_decryptor",
       ":fake_frame_encryptor",
       ":mock_audio_mixer",
+      ":mock_data_channel",
       ":mock_frame_decryptor",
       ":mock_frame_encryptor",
+      ":mock_media_stream_interface",
       ":mock_peer_connection_factory_interface",
       ":mock_peerconnectioninterface",
       ":mock_rtp",

api/DEPS

@@ -128,7 +128,7 @@ specific_include_rules = {
     "+media/base/media_config.h",
     "+media/base/media_engine.h",
     "+p2p/base/port_allocator.h",
-    "+rtc_base/network.h",
+    "+rtc_base/network_monitor_factory.h",
     "+rtc_base/rtc_certificate.h",
     "+rtc_base/rtc_certificate_generator.h",
     "+rtc_base/socket_address.h",
@@ -172,6 +172,9 @@ specific_include_rules = {
     "+rtc_base/ref_count.h",
   ],
 
+  "set_local_description_observer_interface\.h": [
+    "+rtc_base/ref_count.h",
+  ],
   "set_remote_description_observer_interface\.h": [
     "+rtc_base/ref_count.h",
   ],
@@ -258,7 +261,6 @@ specific_include_rules = {
   ],
 
   "simulated_network\.h": [
-    "+rtc_base/critical_section.h",
     "+rtc_base/random.h",
     "+rtc_base/thread_annotations.h",
   ],

api/OWNERS

@@ -2,7 +2,6 @@ crodbro@webrtc.org
 deadbeef@webrtc.org
 hta@webrtc.org
 juberti@webrtc.org
-kwiberg@webrtc.org
 magjed@webrtc.org
 perkj@webrtc.org
 tkchin@webrtc.org
@@ -11,4 +10,16 @@ tommi@webrtc.org
 per-file peer_connection*=hbos@webrtc.org
 
 per-file DEPS=mbonadei@webrtc.org
-per-file DEPS=kwiberg@webrtc.org
+
+# Please keep this list in sync with Chromium's //base/metrics/OWNERS and
+# send a CL when you notice any difference.
+# Even if people in the list below cannot formally grant +1 on WebRTC, it
+# is good to get their LGTM before sending the CL to one of the folder OWNERS.
+per-file uma_metrics.h=asvitkine@chromium.org
+per-file uma_metrics.h=bcwhite@chromium.org
+per-file uma_metrics.h=caitlinfischer@google.com
+per-file uma_metrics.h=holte@chromium.org
+per-file uma_metrics.h=isherman@chromium.org
+per-file uma_metrics.h=jwd@chromium.org
+per-file uma_metrics.h=mpearson@chromium.org
+per-file uma_metrics.h=rkaplow@chromium.org

api/adaptation/BUILD.gn

@@ -16,6 +16,7 @@ rtc_source_set("resource_adaptation_api") {
   ]
   deps = [
     "../../api:scoped_refptr",
+    "../../rtc_base:checks",
     "../../rtc_base:refcount",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:rtc_export",

api/adaptation/resource.cc

@@ -10,6 +10,8 @@
 
 #include "api/adaptation/resource.h"
 
+#include "rtc_base/checks.h"
+
 namespace webrtc {
 
 const char* ResourceUsageStateToString(ResourceUsageState usage_state) {
@@ -19,6 +21,7 @@ const char* ResourceUsageStateToString(ResourceUsageState usage_state) {
     case ResourceUsageState::kUnderuse:
       return "kUnderuse";
   }
+  RTC_CHECK_NOTREACHED();
 }
 
 ResourceListener::~ResourceListener() {}

api/array_view.h

@@ -13,6 +13,7 @@
 
 #include <algorithm>
 #include <array>
+#include <iterator>
 #include <type_traits>
 
 #include "rtc_base/checks.h"
@@ -258,6 +259,18 @@ class ArrayView final : public impl::ArrayViewBase<T, Size> {
   T* end() const { return this->data() + this->size(); }
   const T* cbegin() const { return this->data(); }
   const T* cend() const { return this->data() + this->size(); }
+  std::reverse_iterator<T*> rbegin() const {
+    return std::make_reverse_iterator(end());
+  }
+  std::reverse_iterator<T*> rend() const {
+    return std::make_reverse_iterator(begin());
+  }
+  std::reverse_iterator<const T*> crbegin() const {
+    return std::make_reverse_iterator(cend());
+  }
+  std::reverse_iterator<const T*> crend() const {
+    return std::make_reverse_iterator(cbegin());
+  }
 
   ArrayView<T> subview(size_t offset, size_t size) const {
     return offset < this->size()

api/array_view_unittest.cc

@@ -451,6 +451,20 @@ TEST(ArrayViewTest, TestIterationEmpty) {
   }
 }
 
+TEST(ArrayViewTest, TestReverseIterationEmpty) {
+  // Variable-size.
+  ArrayView<std::vector<std::vector<std::vector<std::string>>>> av;
+  EXPECT_EQ(av.rbegin(), av.rend());
+  EXPECT_EQ(av.crbegin(), av.crend());
+  EXPECT_TRUE(av.empty());
+
+  // Fixed-size.
+  ArrayView<std::vector<std::vector<std::vector<std::string>>>, 0> af;
+  EXPECT_EQ(af.begin(), af.end());
+  EXPECT_EQ(af.cbegin(), af.cend());
+  EXPECT_TRUE(af.empty());
+}
+
 TEST(ArrayViewTest, TestIterationVariable) {
   char arr[] = "Arrr!";
   ArrayView<char> av(arr);
@@ -472,6 +486,25 @@ TEST(ArrayViewTest, TestIterationVariable) {
   }
 }
 
+TEST(ArrayViewTest, TestReverseIterationVariable) {
+  char arr[] = "Arrr!";
+  ArrayView<char> av(arr);
+  EXPECT_EQ('\0', *av.rbegin());
+  EXPECT_EQ('\0', *av.crbegin());
+  EXPECT_EQ('A', *(av.rend() - 1));
+  EXPECT_EQ('A', *(av.crend() - 1));
+
+  const char* cit = av.cend() - 1;
+  for (auto crit = av.crbegin(); crit != av.crend(); ++crit, --cit) {
+    EXPECT_EQ(*cit, *crit);
+  }
+
+  char* it = av.end() - 1;
+  for (auto rit = av.rbegin(); rit != av.rend(); ++rit, --it) {
+    EXPECT_EQ(*it, *rit);
+  }
+}
+
 TEST(ArrayViewTest, TestIterationFixed) {
   char arr[] = "Arrr!";
   ArrayView<char, 6> av(arr);
@@ -493,6 +526,25 @@ TEST(ArrayViewTest, TestIterationFixed) {
   }
 }
 
+TEST(ArrayViewTest, TestReverseIterationFixed) {
+  char arr[] = "Arrr!";
+  ArrayView<char, 6> av(arr);
+  EXPECT_EQ('\0', *av.rbegin());
+  EXPECT_EQ('\0', *av.crbegin());
+  EXPECT_EQ('A', *(av.rend() - 1));
+  EXPECT_EQ('A', *(av.crend() - 1));
+
+  const char* cit = av.cend() - 1;
+  for (auto crit = av.crbegin(); crit != av.crend(); ++crit, --cit) {
+    EXPECT_EQ(*cit, *crit);
+  }
+
+  char* it = av.end() - 1;
+  for (auto rit = av.rbegin(); rit != av.rend(); ++rit, --it) {
+    EXPECT_EQ(*it, *rit);
+  }
+}
+
 TEST(ArrayViewTest, TestEmpty) {
   EXPECT_TRUE(ArrayView<int>().empty());
   const int a[] = {1, 2, 3};

api/audio/BUILD.gn

@@ -24,6 +24,11 @@ rtc_library("audio_frame_api") {
   ]
 }
 
+rtc_source_set("audio_frame_processor") {
+  visibility = [ "*" ]
+  sources = [ "audio_frame_processor.h" ]
+}
+
 rtc_source_set("audio_mixer_api") {
   visibility = [ "*" ]
   sources = [ "audio_mixer.h" ]

api/audio/audio_frame_processor.h

@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2020 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_AUDIO_AUDIO_FRAME_PROCESSOR_H_
+#define API_AUDIO_AUDIO_FRAME_PROCESSOR_H_
+
+#include <functional>
+#include <memory>
+
+namespace webrtc {
+
+class AudioFrame;
+
+// If passed into PeerConnectionFactory, will be used for additional
+// processing of captured audio frames, performed before encoding.
+// Implementations must be thread-safe.
+class AudioFrameProcessor {
+ public:
+  using OnAudioFrameCallback = std::function<void(std::unique_ptr<AudioFrame>)>;
+  virtual ~AudioFrameProcessor() = default;
+
+  // Processes the frame received from WebRTC, is called by WebRTC off the
+  // realtime audio capturing path. AudioFrameProcessor must reply with
+  // processed frames by calling |sink_callback| if it was provided in SetSink()
+  // call. |sink_callback| can be called in the context of Process().
+  virtual void Process(std::unique_ptr<AudioFrame> frame) = 0;
+
+  // Atomically replaces the current sink with the new one. Before the
+  // first call to this function, or if the provided |sink_callback| is nullptr,
+  // processed frames are simply discarded.
+  virtual void SetSink(OnAudioFrameCallback sink_callback) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_AUDIO_FRAME_PROCESSOR_H_

api/audio/echo_canceller3_config.cc

@@ -153,6 +153,7 @@ bool EchoCanceller3Config::Validate(EchoCanceller3Config* config) {
 
   res = res & Limit(&c->filter.config_change_duration_blocks, 0, 100000);
   res = res & Limit(&c->filter.initial_state_seconds, 0.f, 100.f);
+  res = res & Limit(&c->filter.coarse_reset_hangover_blocks, 0, 2500);
 
   res = res & Limit(&c->erle.min, 1.f, 100000.f);
   res = res & Limit(&c->erle.max_l, 1.f, 100000.f);

api/audio/echo_canceller3_config.h

@@ -86,6 +86,7 @@ struct RTC_EXPORT EchoCanceller3Config {
 
     size_t config_change_duration_blocks = 250;
     float initial_state_seconds = 2.5f;
+    int coarse_reset_hangover_blocks = 25;
     bool conservative_initial_phase = false;
     bool enable_coarse_filter_output_usage = true;
     bool use_linear_filter = true;
@@ -143,6 +144,7 @@ struct RTC_EXPORT EchoCanceller3Config {
     float noise_gate_slope = 0.3f;
     size_t render_pre_window_size = 1;
     size_t render_post_window_size = 1;
+    bool model_reverb_in_nonlinear_mode = true;
   } echo_model;
 
   struct ComfortNoise {
@@ -215,11 +217,12 @@ struct RTC_EXPORT EchoCanceller3Config {
     struct HighBandsSuppression {
       float enr_threshold = 1.f;
       float max_gain_during_echo = 1.f;
-      float anti_howling_activation_threshold = 25.f;
+      float anti_howling_activation_threshold = 400.f;
-      float anti_howling_gain = 0.01f;
+      float anti_howling_gain = 1.f;
     } high_bands_suppression;
 
     float floor_first_increase = 0.00001f;
+    bool conservative_hf_suppression = false;
   } suppressor;
 };
 }  // namespace webrtc

api/audio/echo_canceller3_config_json.cc

@@ -223,6 +223,8 @@ void Aec3ConfigFromJsonString(absl::string_view json_string,
               &cfg.filter.config_change_duration_blocks);
     ReadParam(section, "initial_state_seconds",
               &cfg.filter.initial_state_seconds);
+    ReadParam(section, "coarse_reset_hangover_blocks",
+              &cfg.filter.coarse_reset_hangover_blocks);
     ReadParam(section, "conservative_initial_phase",
               &cfg.filter.conservative_initial_phase);
     ReadParam(section, "enable_coarse_filter_output_usage",
@@ -302,6 +304,8 @@ void Aec3ConfigFromJsonString(absl::string_view json_string,
               &cfg.echo_model.render_pre_window_size);
     ReadParam(section, "render_post_window_size",
               &cfg.echo_model.render_post_window_size);
+    ReadParam(section, "model_reverb_in_nonlinear_mode",
+              &cfg.echo_model.model_reverb_in_nonlinear_mode);
   }
 
   if (rtc::GetValueFromJsonObject(aec3_root, "comfort_noise", &section)) {
@@ -381,6 +385,8 @@ void Aec3ConfigFromJsonString(absl::string_view json_string,
 
     ReadParam(section, "floor_first_increase",
               &cfg.suppressor.floor_first_increase);
+    ReadParam(section, "conservative_hf_suppression",
+              &cfg.suppressor.conservative_hf_suppression);
   }
 }
 
@@ -498,6 +504,8 @@ std::string Aec3ConfigToJsonString(const EchoCanceller3Config& config) {
       << config.filter.config_change_duration_blocks << ",";
   ost << "\"initial_state_seconds\": " << config.filter.initial_state_seconds
       << ",";
+  ost << "\"coarse_reset_hangover_blocks\": "
+      << config.filter.coarse_reset_hangover_blocks << ",";
   ost << "\"conservative_initial_phase\": "
       << (config.filter.conservative_initial_phase ? "true" : "false") << ",";
   ost << "\"enable_coarse_filter_output_usage\": "
@@ -585,7 +593,9 @@ std::string Aec3ConfigToJsonString(const EchoCanceller3Config& config) {
   ost << "\"render_pre_window_size\": "
       << config.echo_model.render_pre_window_size << ",";
   ost << "\"render_post_window_size\": "
-      << config.echo_model.render_post_window_size;
+      << config.echo_model.render_post_window_size << ",";
+  ost << "\"model_reverb_in_nonlinear_mode\": "
+      << (config.echo_model.model_reverb_in_nonlinear_mode ? "true" : "false");
   ost << "},";
 
   ost << "\"comfort_noise\": {";
@@ -672,7 +682,10 @@ std::string Aec3ConfigToJsonString(const EchoCanceller3Config& config) {
   ost << "\"anti_howling_gain\": "
       << config.suppressor.high_bands_suppression.anti_howling_gain;
   ost << "},";
-  ost << "\"floor_first_increase\": " << config.suppressor.floor_first_increase;
+  ost << "\"floor_first_increase\": " << config.suppressor.floor_first_increase
+      << ",";
+  ost << "\"conservative_hf_suppression\": "
+      << config.suppressor.conservative_hf_suppression;
   ost << "}";
   ost << "}";
   ost << "}";

api/audio/test/echo_canceller3_config_json_unittest.cc

@@ -22,6 +22,7 @@ TEST(EchoCanceller3JsonHelpers, ToStringAndParseJson) {
   cfg.filter.refined.error_floor = 2.f;
   cfg.filter.coarse_initial.length_blocks = 3u;
   cfg.comfort_noise.noise_floor_dbfs = 100.f;
+  cfg.echo_model.model_reverb_in_nonlinear_mode = false;
   cfg.suppressor.normal_tuning.mask_hf.enr_suppress = .5f;
   cfg.suppressor.subband_nearend_detection.nearend_average_blocks = 3;
   cfg.suppressor.subband_nearend_detection.subband1 = {1, 3};
@@ -48,6 +49,8 @@ TEST(EchoCanceller3JsonHelpers, ToStringAndParseJson) {
             cfg_transformed.filter.refined.error_floor);
   EXPECT_EQ(cfg.comfort_noise.noise_floor_dbfs,
             cfg_transformed.comfort_noise.noise_floor_dbfs);
+  EXPECT_EQ(cfg.echo_model.model_reverb_in_nonlinear_mode,
+            cfg_transformed.echo_model.model_reverb_in_nonlinear_mode);
   EXPECT_EQ(cfg.suppressor.normal_tuning.mask_hf.enr_suppress,
             cfg_transformed.suppressor.normal_tuning.mask_hf.enr_suppress);
   EXPECT_EQ(cfg.suppressor.subband_nearend_detection.nearend_average_blocks,

api/audio_codecs/L16/audio_decoder_L16.cc

@@ -31,6 +31,7 @@ absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
 
 void AudioDecoderL16::AppendSupportedDecoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // Pcm16BAppendSupportedCodecSpecs(specs);
 }
 

api/audio_codecs/L16/audio_encoder_L16.cc

@@ -43,6 +43,7 @@ absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
 
 void AudioEncoderL16::AppendSupportedEncoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // Pcm16BAppendSupportedCodecSpecs(specs);
 }
 

api/audio_codecs/OWNERS

@@ -1 +1,2 @@
-kwiberg@webrtc.org
+minyue@webrtc.org
+henrik.lundin@webrtc.org

api/audio_codecs/audio_decoder_factory.h

@@ -38,6 +38,8 @@ class AudioDecoderFactory : public rtc::RefCountInterface {
   // communication between the AudioEncoder and AudioDecoder instances, which is
   // needed for some codecs with built-in bandwidth adaptation.)
   //
+  // Returns null if the format isn't supported.
+  //
   // Note: Implementations need to be robust against combinations other than
   // one encoder, one decoder getting the same ID; such decoders must still
   // work.

api/audio_codecs/audio_encoder.h

@@ -115,6 +115,7 @@ class AudioEncoder {
     std::vector<EncodedInfoLeaf> redundant;
   };
 
+  // RingRTC change to enable configuring OPUS
   // Very OPUS-specific
   struct Config {
     // AKA ptime or frame size
@@ -275,6 +276,7 @@ class AudioEncoder {
   virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
       const = 0;
 
+  //  RingRTC Change to configure OPUS
   virtual bool Configure(const Config& config) {
     RTC_LOG(LS_WARNING) << "Default AudioEncoder::Configure(...) does nothing!";
     return false;

api/audio_codecs/audio_encoder_factory.h

@@ -44,6 +44,8 @@ class AudioEncoderFactory : public rtc::RefCountInterface {
   // communication between the AudioEncoder and AudioDecoder instances, which is
   // needed for some codecs with built-in bandwidth adaptation.)
   //
+  // Returns null if the format isn't supported.
+  //
   // Note: Implementations need to be robust against combinations other than
   // one encoder, one decoder getting the same ID; such encoders must still
   // work.

api/audio_codecs/g711/audio_decoder_g711.cc

@@ -37,6 +37,7 @@ absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
 
 void AudioDecoderG711::AppendSupportedDecoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // for (const char* type : {"PCMU", "PCMA"}) {
   //   specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
   // }

api/audio_codecs/g711/audio_encoder_g711.cc

@@ -47,6 +47,7 @@ absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
 
 void AudioEncoderG711::AppendSupportedEncoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // for (const char* type : {"PCMU", "PCMA"}) {
   //   specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
   // }

api/audio_codecs/g722/audio_decoder_g722.cc

@@ -31,6 +31,7 @@ absl::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
 
 void AudioDecoderG722::AppendSupportedDecoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // specs->push_back({{"G722", 8000, 1}, {16000, 1, 64000}});
 }
 

api/audio_codecs/g722/audio_encoder_g722.cc

@@ -44,6 +44,7 @@ absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
 
 void AudioEncoderG722::AppendSupportedEncoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // const SdpAudioFormat fmt = {"G722", 8000, 1};
   // const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
   // specs->push_back({fmt, info});

api/audio_codecs/ilbc/audio_decoder_ilbc.cc

@@ -28,6 +28,7 @@ absl::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
 
 void AudioDecoderIlbc::AppendSupportedDecoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // specs->push_back({{"ILBC", 8000, 1}, {8000, 1, 13300}});
 }
 

api/audio_codecs/ilbc/audio_encoder_ilbc.cc

@@ -32,7 +32,7 @@ int GetIlbcBitrate(int ptime) {
       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
       return 13333;
     default:
-      FATAL();
+      RTC_CHECK_NOTREACHED();
   }
 }
 }  // namespace
@@ -59,6 +59,7 @@ absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
 
 void AudioEncoderIlbc::AppendSupportedEncoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // const SdpAudioFormat fmt = {"ILBC", 8000, 1};
   // const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
   // specs->push_back({fmt, info});

api/audio_codecs/isac/audio_decoder_isac_fix.cc

@@ -27,6 +27,7 @@ absl::optional<AudioDecoderIsacFix::Config> AudioDecoderIsacFix::SdpToConfig(
 
 void AudioDecoderIsacFix::AppendSupportedDecoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // specs->push_back({{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}});
 }
 

api/audio_codecs/isac/audio_decoder_isac_float.cc

@@ -32,6 +32,7 @@ AudioDecoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
 
 void AudioDecoderIsacFloat::AppendSupportedDecoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // specs->push_back({{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}});
   // specs->push_back({{"ISAC", 32000, 1}, {32000, 1, 56000, 10000, 56000}});
 }

api/audio_codecs/isac/audio_encoder_isac_fix.cc

@@ -38,6 +38,7 @@ absl::optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
 
 void AudioEncoderIsacFix::AppendSupportedEncoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // const SdpAudioFormat fmt = {"ISAC", 16000, 1};
   // const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
   // specs->push_back({fmt, info});

api/audio_codecs/isac/audio_encoder_isac_float.cc

@@ -45,6 +45,7 @@ AudioEncoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
 
 void AudioEncoderIsacFloat::AppendSupportedEncoders(
     std::vector<AudioCodecSpec>* specs) {
+  // RingRTC change to unused audio codecs
   // for (int sample_rate_hz : {16000, 32000}) {
   //   const SdpAudioFormat fmt = {"ISAC", sample_rate_hz, 1};
   //   const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));

api/create_peerconnection_factory.cc

@@ -18,6 +18,7 @@
 #include "api/rtc_event_log/rtc_event_log_factory.h"
 #include "api/scoped_refptr.h"
 #include "api/task_queue/default_task_queue_factory.h"
+#include "api/transport/field_trial_based_config.h"
 #include "media/base/media_engine.h"
 #include "media/engine/webrtc_media_engine.h"
 #include "modules/audio_device/include/audio_device.h"
@@ -36,7 +37,8 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
     std::unique_ptr<VideoEncoderFactory> video_encoder_factory,
     std::unique_ptr<VideoDecoderFactory> video_decoder_factory,
     rtc::scoped_refptr<AudioMixer> audio_mixer,
-    rtc::scoped_refptr<AudioProcessing> audio_processing) {
+    rtc::scoped_refptr<AudioProcessing> audio_processing,
+    AudioFrameProcessor* audio_frame_processor) {
   PeerConnectionFactoryDependencies dependencies;
   dependencies.network_thread = network_thread;
   dependencies.worker_thread = worker_thread;
@@ -45,12 +47,14 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
   dependencies.call_factory = CreateCallFactory();
   dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>(
       dependencies.task_queue_factory.get());
+  dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
 
   cricket::MediaEngineDependencies media_dependencies;
   media_dependencies.task_queue_factory = dependencies.task_queue_factory.get();
   media_dependencies.adm = std::move(default_adm);
   media_dependencies.audio_encoder_factory = std::move(audio_encoder_factory);
   media_dependencies.audio_decoder_factory = std::move(audio_decoder_factory);
+  media_dependencies.audio_frame_processor = audio_frame_processor;
   if (audio_processing) {
     media_dependencies.audio_processing = std::move(audio_processing);
   } else {
@@ -59,6 +63,7 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
   media_dependencies.audio_mixer = std::move(audio_mixer);
   media_dependencies.video_encoder_factory = std::move(video_encoder_factory);
   media_dependencies.video_decoder_factory = std::move(video_decoder_factory);
+  media_dependencies.trials = dependencies.trials.get();
   dependencies.media_engine =
       cricket::CreateMediaEngine(std::move(media_dependencies));
 

api/create_peerconnection_factory.h

@@ -31,6 +31,7 @@ class Thread;
 namespace webrtc {
 
 class AudioDeviceModule;
+class AudioFrameProcessor;
 class AudioProcessing;
 
 // Create a new instance of PeerConnectionFactoryInterface with optional video
@@ -47,7 +48,8 @@ CreatePeerConnectionFactory(
     std::unique_ptr<VideoEncoderFactory> video_encoder_factory,
     std::unique_ptr<VideoDecoderFactory> video_decoder_factory,
     rtc::scoped_refptr<AudioMixer> audio_mixer,
-    rtc::scoped_refptr<AudioProcessing> audio_processing);
+    rtc::scoped_refptr<AudioProcessing> audio_processing,
+    AudioFrameProcessor* audio_frame_processor = nullptr);
 
 }  // namespace webrtc
 

api/crypto/crypto_options.cc

@@ -23,6 +23,7 @@ CryptoOptions::CryptoOptions(const CryptoOptions& other) {
 
 CryptoOptions::~CryptoOptions() {}
 
+// RingRTC change to use GCM by default
 // static
 CryptoOptions CryptoOptions::Default() {
   CryptoOptions options;

api/crypto/crypto_options.h

@@ -25,6 +25,7 @@ struct RTC_EXPORT CryptoOptions {
   CryptoOptions(const CryptoOptions& other);
   ~CryptoOptions();
 
+  // RingRTC change to use GCM by default
   // Helper method to return an instance of the CryptoOptions with GCM crypto
   // suites disabled. This method should be used instead of depending on current
   // default values set by the constructor.

api/crypto/frame_decryptor_interface.h

@@ -27,7 +27,6 @@ namespace webrtc {
 // without it. You may assume that this interface will have the same lifetime
 // as the RTPReceiver it is attached to. It must only be attached to one
 // RTPReceiver. Additional data may be null.
-// Note: This interface is not ready for production use.
 class FrameDecryptorInterface : public rtc::RefCountInterface {
  public:
   // The Status enum represents all possible states that can be

api/crypto/frame_encryptor_interface.h

@@ -24,7 +24,6 @@ namespace webrtc {
 // addition to the standard SRTP mechanism and is not intended to be used
 // without it. Implementations of this interface will have the same lifetime as
 // the RTPSenders it is attached to. Additional data may be null.
-// Note: This interface is not ready for production use.
 class FrameEncryptorInterface : public rtc::RefCountInterface {
  public:
   ~FrameEncryptorInterface() override {}

api/jsep.h

@@ -136,6 +136,13 @@ class RTC_EXPORT SessionDescriptionInterface {
 
   virtual ~SessionDescriptionInterface() {}
 
+  // Create a new SessionDescriptionInterface object
+  // with the same values as the old object.
+  // TODO(bugs.webrtc.org:12215): Remove default implementation
+  virtual std::unique_ptr<SessionDescriptionInterface> Clone() const {
+    return nullptr;
+  }
+
   // Only for use internally.
   virtual cricket::SessionDescription* description() = 0;
   virtual const cricket::SessionDescription* description() const = 0;

api/jsep_ice_candidate.h

@@ -34,6 +34,8 @@ class RTC_EXPORT JsepIceCandidate : public IceCandidateInterface {
   JsepIceCandidate(const std::string& sdp_mid,
                    int sdp_mline_index,
                    const cricket::Candidate& candidate);
+  JsepIceCandidate(const JsepIceCandidate&) = delete;
+  JsepIceCandidate& operator=(const JsepIceCandidate&) = delete;
   ~JsepIceCandidate() override;
   // |err| may be null.
   bool Initialize(const std::string& sdp, SdpParseError* err);
@@ -53,8 +55,6 @@ class RTC_EXPORT JsepIceCandidate : public IceCandidateInterface {
   std::string sdp_mid_;
   int sdp_mline_index_;
   cricket::Candidate candidate_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(JsepIceCandidate);
 };
 
 // Implementation of IceCandidateCollection which stores JsepIceCandidates.
@@ -64,6 +64,8 @@ class JsepCandidateCollection : public IceCandidateCollection {
   // Move constructor is defined so that a vector of JsepCandidateCollections
   // can be resized.
   JsepCandidateCollection(JsepCandidateCollection&& o);
+  // Returns a copy of the candidate collection.
+  JsepCandidateCollection Clone() const;
   size_t count() const override;
   bool HasCandidate(const IceCandidateInterface* candidate) const override;
   // Adds and takes ownership of the JsepIceCandidate.

api/jsep_session_description.h

@@ -49,6 +49,8 @@ class JsepSessionDescription : public SessionDescriptionInterface {
                   const std::string& session_id,
                   const std::string& session_version);
 
+  virtual std::unique_ptr<SessionDescriptionInterface> Clone() const;
+
   virtual cricket::SessionDescription* description() {
     return description_.get();
   }

api/media_stream_interface.h

@@ -216,6 +216,11 @@ class AudioTrackSinkInterface {
                   number_of_frames);
   }
 
+  // Returns the number of channels encoded by the sink. This can be less than
+  // the number_of_channels if down-mixing occur. A value of -1 means an unknown
+  // number.
+  virtual int NumPreferredChannels() const { return -1; }
+
  protected:
   virtual ~AudioTrackSinkInterface() {}
 };

api/media_stream_proxy.h

@@ -22,7 +22,7 @@ namespace webrtc {
 // are called on is an implementation detail.
 BEGIN_SIGNALING_PROXY_MAP(MediaStream)
 PROXY_SIGNALING_THREAD_DESTRUCTOR()
-PROXY_CONSTMETHOD0(std::string, id)
+BYPASS_PROXY_CONSTMETHOD0(std::string, id)
 PROXY_METHOD0(AudioTrackVector, GetAudioTracks)
 PROXY_METHOD0(VideoTrackVector, GetVideoTracks)
 PROXY_METHOD1(rtc::scoped_refptr<AudioTrackInterface>,

api/media_stream_track.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef PC_MEDIA_STREAM_TRACK_H_
+#ifndef API_MEDIA_STREAM_TRACK_H_
-#define PC_MEDIA_STREAM_TRACK_H_
+#define API_MEDIA_STREAM_TRACK_H_
 
 #include <string>
 
@@ -38,6 +38,7 @@ class MediaStreamTrack : public Notifier<T> {
     }
     return fire_on_change;
   }
+  void set_ended() { set_state(MediaStreamTrackInterface::TrackState::kEnded); }
 
  protected:
   explicit MediaStreamTrack(const std::string& id)
@@ -53,10 +54,10 @@ class MediaStreamTrack : public Notifier<T> {
 
  private:
   bool enabled_;
-  std::string id_;
+  const std::string id_;
   MediaStreamTrackInterface::TrackState state_;
 };
 
 }  // namespace webrtc
 
-#endif  // PC_MEDIA_STREAM_TRACK_H_
+#endif  // API_MEDIA_STREAM_TRACK_H_

api/media_stream_track_proxy.h

@@ -26,8 +26,8 @@ namespace webrtc {
 
 BEGIN_SIGNALING_PROXY_MAP(AudioTrack)
 PROXY_SIGNALING_THREAD_DESTRUCTOR()
-PROXY_CONSTMETHOD0(std::string, kind)
+BYPASS_PROXY_CONSTMETHOD0(std::string, kind)
-PROXY_CONSTMETHOD0(std::string, id)
+BYPASS_PROXY_CONSTMETHOD0(std::string, id)
 PROXY_CONSTMETHOD0(TrackState, state)
 PROXY_CONSTMETHOD0(bool, enabled)
 PROXY_CONSTMETHOD0(AudioSourceInterface*, GetSource)
@@ -42,8 +42,8 @@ END_PROXY_MAP()
 
 BEGIN_PROXY_MAP(VideoTrack)
 PROXY_SIGNALING_THREAD_DESTRUCTOR()
-PROXY_CONSTMETHOD0(std::string, kind)
+BYPASS_PROXY_CONSTMETHOD0(std::string, kind)
-PROXY_CONSTMETHOD0(std::string, id)
+BYPASS_PROXY_CONSTMETHOD0(std::string, id)
 PROXY_CONSTMETHOD0(TrackState, state)
 PROXY_CONSTMETHOD0(bool, enabled)
 PROXY_METHOD1(bool, set_enabled, bool)

api/media_types.cc

@@ -26,10 +26,12 @@ std::string MediaTypeToString(MediaType type) {
       return kMediaTypeVideo;
     case MEDIA_TYPE_DATA:
       return kMediaTypeData;
+    case MEDIA_TYPE_UNSUPPORTED:
+      // Unsupported media stores the m=<mediatype> differently.
+      RTC_NOTREACHED();
+      return "";
   }
-  FATAL();
+  RTC_CHECK_NOTREACHED();
-  // Not reachable; avoids compile warning.
-  return "";
 }
 
 }  // namespace cricket

api/media_types.h

@@ -20,7 +20,12 @@
 
 namespace cricket {
 
-enum MediaType { MEDIA_TYPE_AUDIO, MEDIA_TYPE_VIDEO, MEDIA_TYPE_DATA };
+enum MediaType {
+  MEDIA_TYPE_AUDIO,
+  MEDIA_TYPE_VIDEO,
+  MEDIA_TYPE_DATA,
+  MEDIA_TYPE_UNSUPPORTED
+};
 
 extern const char kMediaTypeAudio[];
 extern const char kMediaTypeVideo[];

api/neteq/neteq.h

@@ -36,7 +36,6 @@ struct NetEqNetworkStatistics {
   uint16_t preferred_buffer_size_ms;  // Target buffer size in ms.
   uint16_t jitter_peaks_found;        // 1 if adding extra delay due to peaky
                                       // jitter; 0 otherwise.
-  uint16_t packet_loss_rate;          // Loss rate (network + late) in Q14.
   uint16_t expand_rate;         // Fraction (of original stream) of synthesized
                                 // audio inserted through expansion (in Q14).
   uint16_t speech_expand_rate;  // Fraction (of original stream) of synthesized
@@ -49,7 +48,6 @@ struct NetEqNetworkStatistics {
                                       // decoding (in Q14).
   uint16_t secondary_discarded_rate;  // Fraction of discarded FEC/RED data (in
                                       // Q14).
-  size_t added_zero_samples;  // Number of zero samples added in "off" mode.
   // Statistics for packet waiting times, i.e., the time between a packet
   // arrives until it is decoded.
   int mean_waiting_time_ms;
@@ -274,6 +272,9 @@ class NetEq {
   // after the call.
   virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;
 
+  // Current values only, not resetting any state.
+  virtual NetEqNetworkStatistics CurrentNetworkStatistics() const = 0;
+
   // Returns a copy of this class's lifetime statistics. These statistics are
   // never reset.
   virtual NetEqLifetimeStatistics GetLifetimeStatistics() const = 0;

api/neteq/neteq_controller.h

@@ -97,6 +97,15 @@ class NetEqController {
     size_t sync_buffer_samples;
   };
 
+  struct PacketArrivedInfo {
+    size_t packet_length_samples;
+    uint32_t main_timestamp;
+    uint16_t main_sequence_number;
+    bool is_cng_or_dtmf;
+    bool is_dtx;
+    bool buffer_flush;
+  };
+
   virtual ~NetEqController() = default;
 
   // Resets object to a clean state.
@@ -152,16 +161,17 @@ class NetEqController {
   virtual void AddSampleMemory(int32_t value) = 0;
 
   // Returns the target buffer level in ms.
-  virtual int TargetLevelMs() = 0;
+  virtual int TargetLevelMs() const = 0;
 
   // Notify the NetEqController that a packet has arrived. Returns the relative
   // arrival delay, if it can be computed.
-  virtual absl::optional<int> PacketArrived(bool last_cng_or_dtmf,
+  virtual absl::optional<int> PacketArrived(int fs_hz,
-                                            size_t packet_length_samples,
                                             bool should_update_stats,
-                                            uint16_t main_sequence_number,
+                                            const PacketArrivedInfo& info) = 0;
-                                            uint32_t main_timestamp,
+
-                                            int fs_hz) = 0;
+  // Notify the NetEqController that we are currently in muted state.
+  // TODO(ivoc): Make pure virtual when downstream is updated.
+  virtual void NotifyMutedState() {}
 
   // Returns true if a peak was found.
   virtual bool PeakFound() const = 0;

api/numerics/BUILD.gn

@@ -0,0 +1,41 @@
+# Copyright (c) 2020 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("numerics") {
+  visibility = [ "*" ]
+
+  sources = [
+    "samples_stats_counter.cc",
+    "samples_stats_counter.h",
+  ]
+  deps = [
+    "..:array_view",
+    "../../rtc_base:checks",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:timeutils",
+    "../units:timestamp",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("numerics_unittests") {
+    visibility = [ "*" ]
+    testonly = true
+
+    sources = [ "samples_stats_counter_unittest.cc" ]
+
+    deps = [
+      ":numerics",
+      "../../test:test_support",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
+  }
+}

api/numerics/DEPS

@@ -0,0 +1,6 @@
+specific_include_rules = {
+  # Some internal headers are allowed even in API headers:
+  "samples_stats_counter\.h": [
+    "+rtc_base/numerics/running_statistics.h",
+  ]
+}

api/numerics/samples_stats_counter.cc

@@ -8,8 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "rtc_base/numerics/samples_stats_counter.h"
+#include "api/numerics/samples_stats_counter.h"
 
+#include <algorithm>
 #include <cmath>
 
 #include "absl/algorithm/container.h"

api/numerics/samples_stats_counter.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef RTC_BASE_NUMERICS_SAMPLES_STATS_COUNTER_H_
+#ifndef API_NUMERICS_SAMPLES_STATS_COUNTER_H_
-#define RTC_BASE_NUMERICS_SAMPLES_STATS_COUNTER_H_
+#define API_NUMERICS_SAMPLES_STATS_COUNTER_H_
 
 #include <vector>
 
@@ -45,6 +45,8 @@ class SamplesStatsCounter {
 
   // Returns if there are any values in O(1) time.
   bool IsEmpty() const { return samples_.empty(); }
+  // Returns the amount of samples added into counter in O(1) time.
+  int64_t NumSamples() const { return stats_.Size(); }
 
   // Returns min in O(1) time. This function may not be called if there are no
   // samples.
@@ -98,7 +100,7 @@ class SamplesStatsCounter {
   }
 
  private:
-  RunningStatistics<double> stats_;
+  webrtc_impl::RunningStatistics<double> stats_;
   std::vector<StatsSample> samples_;
   bool sorted_ = false;
 };
@@ -116,4 +118,4 @@ SamplesStatsCounter operator/(const SamplesStatsCounter& counter, double value);
 
 }  // namespace webrtc
 
-#endif  // RTC_BASE_NUMERICS_SAMPLES_STATS_COUNTER_H_
+#endif  // API_NUMERICS_SAMPLES_STATS_COUNTER_H_

api/numerics/samples_stats_counter_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "rtc_base/numerics/samples_stats_counter.h"
+#include "api/numerics/samples_stats_counter.h"
 
 #include <math.h>
 

api/peer_connection_factory_proxy.h

@@ -36,6 +36,10 @@ PROXY_METHOD2(rtc::scoped_refptr<PeerConnectionInterface>,
               CreatePeerConnection,
               const PeerConnectionInterface::RTCConfiguration&,
               PeerConnectionDependencies)
+PROXY_METHOD2(RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>>,
+              CreatePeerConnectionOrError,
+              const PeerConnectionInterface::RTCConfiguration&,
+              PeerConnectionDependencies)
 PROXY_CONSTMETHOD1(webrtc::RtpCapabilities,
                    GetRtpSenderCapabilities,
                    cricket::MediaType)

api/peer_connection_interface.cc

@@ -54,54 +54,38 @@ RTCError PeerConnectionInterface::SetConfiguration(
   return RTCError();
 }
 
-RTCError PeerConnectionInterface::SetBitrate(const BitrateSettings& bitrate) {
+// RingRTC change to add ICE forking
-  BitrateParameters bitrate_parameters;
-  bitrate_parameters.min_bitrate_bps = bitrate.min_bitrate_bps;
-  bitrate_parameters.current_bitrate_bps = bitrate.start_bitrate_bps;
-  bitrate_parameters.max_bitrate_bps = bitrate.max_bitrate_bps;
-  return SetBitrate(bitrate_parameters);
-}
-
-RTCError PeerConnectionInterface::SetBitrate(
-    const BitrateParameters& bitrate_parameters) {
-  BitrateSettings bitrate;
-  bitrate.min_bitrate_bps = bitrate_parameters.min_bitrate_bps;
-  bitrate.start_bitrate_bps = bitrate_parameters.current_bitrate_bps;
-  bitrate.max_bitrate_bps = bitrate_parameters.max_bitrate_bps;
-  return SetBitrate(bitrate);
-}
-
 rtc::scoped_refptr<webrtc::IceGathererInterface>
 PeerConnectionInterface::CreateSharedIceGatherer() {
   RTC_LOG(LS_ERROR) << "No shared ICE gatherer in dummy implementation";
   return nullptr;
 }
 
+// RingRTC change to add ICE forking
 bool PeerConnectionInterface::UseSharedIceGatherer(
     rtc::scoped_refptr<webrtc::IceGathererInterface> shared_ice_gatherer) {
   RTC_LOG(LS_ERROR) << "No shared ICE gatherer in dummy implementation";
   return false;
 }
 
+// RingRTC change to RTP from being processed before the call is accepted
 bool PeerConnectionInterface::SetIncomingRtpEnabled(bool enabled) {
   RTC_LOG(LS_ERROR) << "No enabling of incoming RTP in dummy implementation";
   return false;
 }
 
+// RingRTC change to send RTP data
 bool PeerConnectionInterface::SendRtp(std::unique_ptr<RtpPacket> rtp_packet) {
   RTC_LOG(LS_ERROR) << "No SendRtp in dummy implementation";
   return false;
 }
 
+// RingRTC change to receive RTP data
 bool PeerConnectionInterface::ReceiveRtp(uint8_t pt) {
   RTC_LOG(LS_ERROR) << "No SendRtp in dummy implementation";
   return false;
 }
 
-PeerConnectionInterface::BitrateParameters::BitrateParameters() = default;
-
-PeerConnectionInterface::BitrateParameters::~BitrateParameters() = default;
-
 PeerConnectionDependencies::PeerConnectionDependencies(
     PeerConnectionObserver* observer_in)
     : observer(observer_in) {}
@@ -136,6 +120,13 @@ PeerConnectionFactoryInterface::CreatePeerConnection(
   return nullptr;
 }
 
+RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>>
+PeerConnectionFactoryInterface::CreatePeerConnectionOrError(
+    const PeerConnectionInterface::RTCConfiguration& configuration,
+    PeerConnectionDependencies dependencies) {
+  return RTCError(RTCErrorType::INTERNAL_ERROR);
+}
+
 RtpCapabilities PeerConnectionFactoryInterface::GetRtpSenderCapabilities(
     cricket::MediaType kind) const {
   return {};

api/peer_connection_interface.h

@@ -98,6 +98,7 @@
 #include "api/rtp_sender_interface.h"
 #include "api/rtp_transceiver_interface.h"
 #include "api/sctp_transport_interface.h"
+#include "api/set_local_description_observer_interface.h"
 #include "api/set_remote_description_observer_interface.h"
 #include "api/stats/rtc_stats_collector_callback.h"
 #include "api/stats_types.h"
@@ -105,6 +106,7 @@
 #include "api/transport/bitrate_settings.h"
 #include "api/transport/enums.h"
 #include "api/transport/network_control.h"
+#include "api/transport/sctp_transport_factory_interface.h"
 #include "api/transport/webrtc_key_value_config.h"
 #include "api/turn_customizer.h"
 #include "call/rtp_packet_sink_interface.h"
@@ -116,7 +118,7 @@
 // inject a PacketSocketFactory and/or NetworkManager, and not expose
 // PortAllocator in the PeerConnection api.
 #include "p2p/base/port_allocator.h"  // nogncheck
-#include "rtc_base/network.h"
+#include "rtc_base/network_monitor_factory.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/rtc_certificate_generator.h"
 #include "rtc_base/socket_address.h"
@@ -642,8 +644,9 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     // Whether network condition based codec switching is allowed.
     absl::optional<bool> allow_codec_switching;
 
-    bool enable_simulcast_stats = true;
+    // The delay before doing a usage histogram report for long-lived
-
+    // PeerConnections. Used for testing only.
+    absl::optional<int> report_usage_pattern_delay_ms;
     //
     // Don't forget to update operator== if adding something.
     //
@@ -914,6 +917,10 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
       const std::string& label,
       const DataChannelInit* config) = 0;
 
+  // NOTE: For the following 6 methods, it's only safe to dereference the
+  // SessionDescriptionInterface on signaling_thread() (for example, calling
+  // ToString).
+
   // Returns the more recently applied description; "pending" if it exists, and
   // otherwise "current". See below.
   virtual const SessionDescriptionInterface* local_description() const = 0;
@@ -953,26 +960,66 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
                             const RTCOfferAnswerOptions& options) = 0;
 
   // Sets the local session description.
-  // The PeerConnection takes the ownership of |desc| even if it fails.
+  //
-  // The |observer| callback will be called when done.
+  // According to spec, the local session description MUST be the same as was
-  // TODO(deadbeef): Change |desc| to be a unique_ptr, to make it clear
+  // returned by CreateOffer() or CreateAnswer() or else the operation should
-  // that this method always takes ownership of it.
+  // fail. Our implementation however allows some amount of "SDP munging", but
+  // please note that this is HIGHLY DISCOURAGED. If you do not intent to munge
+  // SDP, the method below that doesn't take |desc| as an argument will create
+  // the offer or answer for you.
+  //
+  // The observer is invoked as soon as the operation completes, which could be
+  // before or after the SetLocalDescription() method has exited.
+  virtual void SetLocalDescription(
+      std::unique_ptr<SessionDescriptionInterface> desc,
+      rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {}
+  // Creates an offer or answer (depending on current signaling state) and sets
+  // it as the local session description.
+  //
+  // The observer is invoked as soon as the operation completes, which could be
+  // before or after the SetLocalDescription() method has exited.
+  virtual void SetLocalDescription(
+      rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {}
+  // Like SetLocalDescription() above, but the observer is invoked with a delay
+  // after the operation completes. This helps avoid recursive calls by the
+  // observer but also makes it possible for states to change in-between the
+  // operation completing and the observer getting called. This makes them racy
+  // for synchronizing peer connection states to the application.
+  // TODO(https://crbug.com/webrtc/11798): Delete these methods in favor of the
+  // ones taking SetLocalDescriptionObserverInterface as argument.
   virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
                                    SessionDescriptionInterface* desc) = 0;
-  // Implicitly creates an offer or answer (depending on the current signaling
-  // state) and performs SetLocalDescription() with the newly generated session
-  // description.
-  // TODO(hbos): Make pure virtual when implemented by downstream projects.
   virtual void SetLocalDescription(SetSessionDescriptionObserver* observer) {}
+
   // Sets the remote session description.
-  // The PeerConnection takes the ownership of |desc| even if it fails.
+  //
-  // The |observer| callback will be called when done.
+  // (Unlike "SDP munging" before SetLocalDescription(), modifying a remote
-  // TODO(hbos): Remove when Chrome implements the new signature.
+  // offer or answer is allowed by the spec.)
-  virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
+  //
-                                    SessionDescriptionInterface* desc) {}
+  // The observer is invoked as soon as the operation completes, which could be
+  // before or after the SetRemoteDescription() method has exited.
   virtual void SetRemoteDescription(
       std::unique_ptr<SessionDescriptionInterface> desc,
       rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) = 0;
+  // Like SetRemoteDescription() above, but the observer is invoked with a delay
+  // after the operation completes. This helps avoid recursive calls by the
+  // observer but also makes it possible for states to change in-between the
+  // operation completing and the observer getting called. This makes them racy
+  // for synchronizing peer connection states to the application.
+  // TODO(https://crbug.com/webrtc/11798): Delete this method in favor of the
+  // ones taking SetRemoteDescriptionObserverInterface as argument.
+  virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
+                                    SessionDescriptionInterface* desc) {}
+
+  // According to spec, we must only fire "negotiationneeded" if the Operations
+  // Chain is empty. This method takes care of validating an event previously
+  // generated with PeerConnectionObserver::OnNegotiationNeededEvent() to make
+  // sure that even if there was a delay (e.g. due to a PostTask) between the
+  // event being generated and the time of firing, the Operations Chain is empty
+  // and the event is still valid to be fired.
+  virtual bool ShouldFireNegotiationNeededEvent(uint32_t event_id) {
+    return true;
+  }
 
   virtual PeerConnectionInterface::RTCConfiguration GetConfiguration() = 0;
 
@@ -1021,10 +1068,13 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   virtual bool RemoveIceCandidates(
       const std::vector<cricket::Candidate>& candidates) = 0;
 
+
+  // RingRTC change to add ICE forking
   // Creates an IceGatherer that can be shared/used with UseSharedIceGatherer
   virtual rtc::scoped_refptr<webrtc::IceGathererInterface>
   CreateSharedIceGatherer();
 
+  // RingRTC change to add ICE forking
   // SetGatherer with the same IceGatherer on many ICE transports to get
   // ICE forking behavior.  For example, like so:
   // auto gatherer = pc1->CreateSharedIceGatherer();
@@ -1036,13 +1086,16 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   virtual bool UseSharedIceGatherer(
       rtc::scoped_refptr<webrtc::IceGathererInterface> shared_ice_gatherer);
 
+  // RingRTC change to RTP from being processed before the call is accepted
   // If false, all RTP and RTCP packets will be dropped before being processed.
   virtual bool SetIncomingRtpEnabled(bool enabled);
 
+  // RingRTC change to send RTP data
   // Make sure that you don't reuse (SSRC, seqnum) combinations except when rolling
   // over.  Otherwise, SRTP won't work properly.
   virtual bool SendRtp(std::unique_ptr<RtpPacket> rtp_packet);
 
+  // RingRTC change to receive RTP data
   // Packets will go to the PeerConnectionObserver
   virtual bool ReceiveRtp(uint8_t pt);
 
@@ -1050,15 +1103,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     RTC_LOG(LS_WARNING) << "Default PeerConnectionInterface::ConfigureAudioEncoders(...) does nothing!";
   }
 
-  // 0 <= min <= current <= max should hold for set parameters.
-  struct BitrateParameters {
-    BitrateParameters();
-    ~BitrateParameters();
-
-    absl::optional<int> min_bitrate_bps;
-    absl::optional<int> current_bitrate_bps;
-    absl::optional<int> max_bitrate_bps;
-  };
 
   // SetBitrate limits the bandwidth allocated for all RTP streams sent by
   // this PeerConnection. Other limitations might affect these limits and
@@ -1066,12 +1110,7 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   //
   // Setting |current_bitrate_bps| will reset the current bitrate estimate
   // to the provided value.
-  virtual RTCError SetBitrate(const BitrateSettings& bitrate);
+  virtual RTCError SetBitrate(const BitrateSettings& bitrate) = 0;
-
-  // TODO(nisse): Deprecated - use version above. These two default
-  // implementations require subclasses to implement one or the other
-  // of the methods.
-  virtual RTCError SetBitrate(const BitrateParameters& bitrate_parameters);
 
   // Enable/disable playout of received audio streams. Enabled by default. Note
   // that even if playout is enabled, streams will only be played out if the
@@ -1155,6 +1194,14 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
   // thus the observer object can be safely destroyed.
   virtual void Close() = 0;
 
+  // The thread on which all PeerConnectionObserver callbacks will be invoked,
+  // as well as callbacks for other classes such as DataChannelObserver.
+  //
+  // Also the only thread on which it's safe to use SessionDescriptionInterface
+  // pointers.
+  // TODO(deadbeef): Make pure virtual when all subclasses implement it.
+  virtual rtc::Thread* signaling_thread() const { return nullptr; }
+
  protected:
   // Dtor protected as objects shouldn't be deleted via this interface.
   ~PeerConnectionInterface() override = default;
@@ -1183,7 +1230,17 @@ class PeerConnectionObserver : public RtpPacketSinkInterface {
 
   // Triggered when renegotiation is needed. For example, an ICE restart
   // has begun.
-  virtual void OnRenegotiationNeeded() = 0;
+  // TODO(hbos): Delete in favor of OnNegotiationNeededEvent() when downstream
+  // projects have migrated.
+  virtual void OnRenegotiationNeeded() {}
+  // Used to fire spec-compliant onnegotiationneeded events, which should only
+  // fire when the Operations Chain is empty. The observer is responsible for
+  // queuing a task (e.g. Chromium: jump to main thread) to maybe fire the
+  // event. The event identified using |event_id| must only fire if
+  // PeerConnection::ShouldFireNegotiationNeededEvent() returns true since it is
+  // possible for the event to become invalidated by operations subsequently
+  // chained.
+  virtual void OnNegotiationNeededEvent(uint32_t event_id) {}
 
   // Called any time the legacy IceConnectionState changes.
   //
@@ -1347,7 +1404,12 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final {
   std::unique_ptr<NetworkStatePredictorFactoryInterface>
       network_state_predictor_factory;
   std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
+  // This will only be used if CreatePeerConnection is called without a
+  // |port_allocator|, causing the default allocator and network manager to be
+  // used.
+  std::unique_ptr<rtc::NetworkMonitorFactory> network_monitor_factory;
   std::unique_ptr<NetEqFactory> neteq_factory;
+  std::unique_ptr<SctpTransportFactoryInterface> sctp_factory;
   std::unique_ptr<WebRtcKeyValueConfig> trials;
 };
 
@@ -1408,6 +1470,12 @@ class RTC_EXPORT PeerConnectionFactoryInterface
   // configuration and a PeerConnectionDependencies structure.
   // TODO(benwright): Make pure virtual once downstream mock PC factory classes
   // are updated.
+  virtual RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>>
+  CreatePeerConnectionOrError(
+      const PeerConnectionInterface::RTCConfiguration& configuration,
+      PeerConnectionDependencies dependencies);
+  // Deprecated creator - does not return an error code on error.
+  // TODO(bugs.webrtc.org:12238): Deprecate and remove.
   virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
       const PeerConnectionInterface::RTCConfiguration& configuration,
       PeerConnectionDependencies dependencies);

api/peer_connection_proxy.h

@@ -96,19 +96,27 @@ PROXY_METHOD2(void,
               CreateAnswer,
               CreateSessionDescriptionObserver*,
               const RTCOfferAnswerOptions&)
+PROXY_METHOD2(void,
+              SetLocalDescription,
+              std::unique_ptr<SessionDescriptionInterface>,
+              rtc::scoped_refptr<SetLocalDescriptionObserverInterface>)
+PROXY_METHOD1(void,
+              SetLocalDescription,
+              rtc::scoped_refptr<SetLocalDescriptionObserverInterface>)
 PROXY_METHOD2(void,
               SetLocalDescription,
               SetSessionDescriptionObserver*,
               SessionDescriptionInterface*)
 PROXY_METHOD1(void, SetLocalDescription, SetSessionDescriptionObserver*)
-PROXY_METHOD2(void,
-              SetRemoteDescription,
-              SetSessionDescriptionObserver*,
-              SessionDescriptionInterface*)
 PROXY_METHOD2(void,
               SetRemoteDescription,
               std::unique_ptr<SessionDescriptionInterface>,
               rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>)
+PROXY_METHOD2(void,
+              SetRemoteDescription,
+              SetSessionDescriptionObserver*,
+              SessionDescriptionInterface*)
+PROXY_METHOD1(bool, ShouldFireNegotiationNeededEvent, uint32_t)
 PROXY_METHOD0(PeerConnectionInterface::RTCConfiguration, GetConfiguration)
 PROXY_METHOD1(RTCError,
               SetConfiguration,
@@ -155,6 +163,7 @@ PROXY_METHOD2(bool,
 PROXY_METHOD1(bool, StartRtcEventLog, std::unique_ptr<RtcEventLogOutput>)
 PROXY_METHOD0(void, StopRtcEventLog)
 PROXY_METHOD0(void, Close)
+BYPASS_PROXY_CONSTMETHOD0(rtc::Thread*, signaling_thread)
 END_PROXY_MAP()
 
 }  // namespace webrtc

api/proxy.cc

@@ -10,28 +10,3 @@
 
 #include "api/proxy.h"
 
-namespace webrtc {
-namespace internal {
-
-SynchronousMethodCall::SynchronousMethodCall(rtc::MessageHandler* proxy)
-    : proxy_(proxy) {}
-
-SynchronousMethodCall::~SynchronousMethodCall() = default;
-
-void SynchronousMethodCall::Invoke(const rtc::Location& posted_from,
-                                   rtc::Thread* t) {
-  if (t->IsCurrent()) {
-    proxy_->OnMessage(nullptr);
-  } else {
-    t->Post(posted_from, this, 0);
-    e_.Wait(rtc::Event::kForever);
-  }
-}
-
-void SynchronousMethodCall::OnMessage(rtc::Message*) {
-  proxy_->OnMessage(nullptr);
-  e_.Set();
-}
-
-}  // namespace internal
-}  // namespace webrtc

api/proxy.h

@@ -59,6 +59,8 @@
 #include <utility>
 
 #include "api/scoped_refptr.h"
+#include "api/task_queue/queued_task.h"
+#include "api/task_queue/task_queue_base.h"
 #include "rtc_base/event.h"
 #include "rtc_base/message_handler.h"
 #include "rtc_base/ref_counted_object.h"
@@ -96,27 +98,8 @@ class ReturnType<void> {
   void moved_result() {}
 };
 
-namespace internal {
-
-class RTC_EXPORT SynchronousMethodCall : public rtc::MessageData,
-                                         public rtc::MessageHandler {
- public:
-  explicit SynchronousMethodCall(rtc::MessageHandler* proxy);
-  ~SynchronousMethodCall() override;
-
-  void Invoke(const rtc::Location& posted_from, rtc::Thread* t);
-
- private:
-  void OnMessage(rtc::Message*) override;
-
-  rtc::Event e_;
-  rtc::MessageHandler* proxy_;
-};
-
-}  // namespace internal
-
 template <typename C, typename R, typename... Args>
-class MethodCall : public rtc::Message, public rtc::MessageHandler {
+class MethodCall : public QueuedTask {
  public:
   typedef R (C::*Method)(Args...);
   MethodCall(C* c, Method m, Args&&... args)
@@ -125,12 +108,21 @@ class MethodCall : public rtc::Message, public rtc::MessageHandler {
         args_(std::forward_as_tuple(std::forward<Args>(args)...)) {}
 
   R Marshal(const rtc::Location& posted_from, rtc::Thread* t) {
-    internal::SynchronousMethodCall(this).Invoke(posted_from, t);
+    if (t->IsCurrent()) {
+      Invoke(std::index_sequence_for<Args...>());
+    } else {
+      t->PostTask(std::unique_ptr<QueuedTask>(this));
+      event_.Wait(rtc::Event::kForever);
+    }
     return r_.moved_result();
   }
 
  private:
-  void OnMessage(rtc::Message*) { Invoke(std::index_sequence_for<Args...>()); }
+  bool Run() override {
+    Invoke(std::index_sequence_for<Args...>());
+    event_.Set();
+    return false;
+  }
 
   template <size_t... Is>
   void Invoke(std::index_sequence<Is...>) {
@@ -141,10 +133,11 @@ class MethodCall : public rtc::Message, public rtc::MessageHandler {
   Method m_;
   ReturnType<R> r_;
   std::tuple<Args&&...> args_;
+  rtc::Event event_;
 };
 
 template <typename C, typename R, typename... Args>
-class ConstMethodCall : public rtc::Message, public rtc::MessageHandler {
+class ConstMethodCall : public QueuedTask {
  public:
   typedef R (C::*Method)(Args...) const;
   ConstMethodCall(const C* c, Method m, Args&&... args)
@@ -153,12 +146,21 @@ class ConstMethodCall : public rtc::Message, public rtc::MessageHandler {
         args_(std::forward_as_tuple(std::forward<Args>(args)...)) {}
 
   R Marshal(const rtc::Location& posted_from, rtc::Thread* t) {
-    internal::SynchronousMethodCall(this).Invoke(posted_from, t);
+    if (t->IsCurrent()) {
+      Invoke(std::index_sequence_for<Args...>());
+    } else {
+      t->PostTask(std::unique_ptr<QueuedTask>(this));
+      event_.Wait(rtc::Event::kForever);
+    }
     return r_.moved_result();
   }
 
  private:
-  void OnMessage(rtc::Message*) { Invoke(std::index_sequence_for<Args...>()); }
+  bool Run() override {
+    Invoke(std::index_sequence_for<Args...>());
+    event_.Set();
+    return false;
+  }
 
   template <size_t... Is>
   void Invoke(std::index_sequence<Is...>) {
@@ -169,6 +171,7 @@ class ConstMethodCall : public rtc::Message, public rtc::MessageHandler {
   Method m_;
   ReturnType<R> r_;
   std::tuple<Args&&...> args_;
+  rtc::Event event_;
 };
 
 // Helper macros to reduce code duplication.
@@ -400,11 +403,13 @@ class ConstMethodCall : public rtc::Message, public rtc::MessageHandler {
 // For use when returning purely const state (set during construction).
 // Use with caution. This method should only be used when the return value will
 // always be the same.
-#define BYPASS_PROXY_CONSTMETHOD0(r, method)                            \
+#define BYPASS_PROXY_CONSTMETHOD0(r, method)                                \
-  r method() const override {                                           \
+  r method() const override {                                               \
-    static_assert(!std::is_pointer<r>::value, "Type is a pointer");     \
+    static_assert(                                                          \
-    static_assert(!std::is_reference<r>::value, "Type is a reference"); \
+        std::is_same<r, rtc::Thread*>::value || !std::is_pointer<r>::value, \
-    return c_->method();                                                \
+        "Type is a pointer");                                               \
+    static_assert(!std::is_reference<r>::value, "Type is a reference");     \
+    return c_->method();                                                    \
   }
 
 }  // namespace webrtc

api/rtc_error.h

@@ -11,9 +11,9 @@
 #ifndef API_RTC_ERROR_H_
 #define API_RTC_ERROR_H_
 
-#ifdef UNIT_TEST
+#ifdef WEBRTC_UNIT_TEST
 #include <ostream>
-#endif  // UNIT_TEST
+#endif  // WEBRTC_UNIT_TEST
 #include <string>
 #include <utility>  // For std::move.
 
@@ -137,7 +137,7 @@ class RTC_EXPORT RTCError {
 
   RTCErrorDetailType error_detail() const { return error_detail_; }
   void set_error_detail(RTCErrorDetailType detail) { error_detail_ = detail; }
-  absl::optional<uint16_t> sctp_cause_code() { return sctp_cause_code_; }
+  absl::optional<uint16_t> sctp_cause_code() const { return sctp_cause_code_; }
   void set_sctp_cause_code(uint16_t cause_code) {
     sctp_cause_code_ = cause_code;
   }
@@ -161,7 +161,7 @@ class RTC_EXPORT RTCError {
 RTC_EXPORT const char* ToString(RTCErrorType error);
 RTC_EXPORT const char* ToString(RTCErrorDetailType error);
 
-#ifdef UNIT_TEST
+#ifdef WEBRTC_UNIT_TEST
 inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
     std::ostream& stream,         // no-presubmit-check TODO(webrtc:8982)
     RTCErrorType error) {
@@ -173,7 +173,7 @@ inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
     RTCErrorDetailType error) {
   return stream << ToString(error);
 }
-#endif  // UNIT_TEST
+#endif  // WEBRTC_UNIT_TEST
 
 // Helper macro that can be used by implementations to create an error with a
 // message and log it. |message| should be a string literal or movable

api/rtc_event_log/BUILD.gn

@@ -37,6 +37,7 @@ rtc_library("rtc_event_log_factory") {
     ":rtc_event_log",
     "../../rtc_base:checks",
     "../../rtc_base/system:rtc_export",
+    "../../system_wrappers:field_trial",
     "../task_queue",
   ]
 

api/rtc_event_log/rtc_event.h

@@ -52,7 +52,8 @@ class RtcEvent {
     VideoSendStreamConfig,
     GenericPacketSent,
     GenericPacketReceived,
-    GenericAckReceived
+    GenericAckReceived,
+    FrameDecoded
   };
 
   RtcEvent();

api/rtc_event_log/rtc_event_log_factory.cc

@@ -14,6 +14,7 @@
 #include <utility>
 
 #include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
 
 #ifdef WEBRTC_ENABLE_RTC_EVENT_LOG
 #include "logging/rtc_event_log/rtc_event_log_impl.h"
@@ -29,6 +30,9 @@ RtcEventLogFactory::RtcEventLogFactory(TaskQueueFactory* task_queue_factory)
 std::unique_ptr<RtcEventLog> RtcEventLogFactory::CreateRtcEventLog(
     RtcEventLog::EncodingType encoding_type) {
 #ifdef WEBRTC_ENABLE_RTC_EVENT_LOG
+  if (field_trial::IsEnabled("WebRTC-RtcEventLogKillSwitch")) {
+    return std::make_unique<RtcEventLogNull>();
+  }
   return std::make_unique<RtcEventLogImpl>(encoding_type, task_queue_factory_);
 #else
   return std::make_unique<RtcEventLogNull>();

api/rtp_headers.h

@@ -23,7 +23,6 @@
 #include "api/video/video_content_type.h"
 #include "api/video/video_rotation.h"
 #include "api/video/video_timing.h"
-#include "common_types.h"  // NOLINT (build/include)
 
 namespace webrtc {
 
@@ -142,7 +141,7 @@ struct RTPHeaderExtension {
   bool has_video_timing;
   VideoSendTiming video_timing;
 
-  PlayoutDelay playout_delay = {-1, -1};
+  VideoPlayoutDelay playout_delay;
 
   // For identification of a stream when ssrc is not signaled. See
   // https://tools.ietf.org/html/draft-ietf-avtext-rid-09

api/rtp_parameters.cc

@@ -30,6 +30,7 @@ const char* DegradationPreferenceToString(
     case DegradationPreference::BALANCED:
       return "balanced";
   }
+  RTC_CHECK_NOTREACHED();
 }
 
 const double kDefaultBitratePriority = 1.0;
@@ -121,6 +122,7 @@ constexpr char RtpExtension::kVideoContentTypeUri[];
 constexpr char RtpExtension::kVideoTimingUri[];
 constexpr char RtpExtension::kGenericFrameDescriptorUri00[];
 constexpr char RtpExtension::kDependencyDescriptorUri[];
+constexpr char RtpExtension::kVideoLayersAllocationUri[];
 constexpr char RtpExtension::kTransportSequenceNumberUri[];
 constexpr char RtpExtension::kTransportSequenceNumberV2Uri[];
 constexpr char RtpExtension::kPlayoutDelayUri[];
@@ -161,7 +163,8 @@ bool RtpExtension::IsSupportedForVideo(absl::string_view uri) {
          uri == webrtc::RtpExtension::kDependencyDescriptorUri ||
          uri == webrtc::RtpExtension::kColorSpaceUri ||
          uri == webrtc::RtpExtension::kRidUri ||
-         uri == webrtc::RtpExtension::kRepairedRidUri;
+         uri == webrtc::RtpExtension::kRepairedRidUri ||
+         uri == webrtc::RtpExtension::kVideoLayersAllocationUri;
 }
 
 bool RtpExtension::IsEncryptionSupported(absl::string_view uri) {
@@ -183,7 +186,8 @@ bool RtpExtension::IsEncryptionSupported(absl::string_view uri) {
          uri == webrtc::RtpExtension::kVideoContentTypeUri ||
          uri == webrtc::RtpExtension::kMidUri ||
          uri == webrtc::RtpExtension::kRidUri ||
-         uri == webrtc::RtpExtension::kRepairedRidUri;
+         uri == webrtc::RtpExtension::kRepairedRidUri ||
+         uri == webrtc::RtpExtension::kVideoLayersAllocationUri;
 }
 
 const RtpExtension* RtpExtension::FindHeaderExtensionByUri(

api/rtp_parameters.h

@@ -318,6 +318,10 @@ struct RTC_EXPORT RtpExtension {
       "https://aomediacodec.github.io/av1-rtp-spec/"
       "#dependency-descriptor-rtp-header-extension";
 
+  // Experimental extension for signalling target bitrate per layer.
+  static constexpr char kVideoLayersAllocationUri[] =
+      "http://www.webrtc.org/experiments/rtp-hdrext/video-layers-allocation00";
+
   // Header extension for transport sequence number, see url for details:
   // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
   static constexpr char kTransportSequenceNumberUri[] =
@@ -461,6 +465,9 @@ struct RTC_EXPORT RtpEncodingParameters {
   // For video, scale the resolution down by this factor.
   absl::optional<double> scale_resolution_down_by;
 
+  // https://w3c.github.io/webrtc-svc/#rtcrtpencodingparameters
+  absl::optional<std::string> scalability_mode;
+
   // For an RtpSender, set to true to cause this encoding to be encoded and
   // sent, and false for it not to be encoded and sent. This allows control
   // across multiple encodings of a sender for turning simulcast layers on and

api/rtp_receiver_interface.h

@@ -128,8 +128,8 @@ PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
 PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
 PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>,
                    streams)
-PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
+BYPASS_PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
-PROXY_CONSTMETHOD0(std::string, id)
+BYPASS_PROXY_CONSTMETHOD0(std::string, id)
 PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
 PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*)
 PROXY_METHOD1(void, SetJitterBufferMinimumDelay, absl::optional<double>)

api/rtp_sender_interface.h

@@ -110,8 +110,8 @@ PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
 PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
 PROXY_CONSTMETHOD0(uint32_t, ssrc)
-PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
+BYPASS_PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
-PROXY_CONSTMETHOD0(std::string, id)
+BYPASS_PROXY_CONSTMETHOD0(std::string, id)
 PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
 PROXY_CONSTMETHOD0(std::vector<RtpEncodingParameters>, init_send_encodings)
 PROXY_CONSTMETHOD0(RtpParameters, GetParameters)

api/rtp_transceiver_interface.cc

@@ -25,6 +25,23 @@ RtpTransceiverInterface::fired_direction() const {
   return absl::nullopt;
 }
 
+bool RtpTransceiverInterface::stopping() const {
+  return false;
+}
+
+void RtpTransceiverInterface::Stop() {
+  StopInternal();
+}
+
+RTCError RtpTransceiverInterface::StopStandard() {
+  RTC_NOTREACHED() << "DEBUG: RtpTransceiverInterface::StopStandard called";
+  return RTCError::OK();
+}
+
+void RtpTransceiverInterface::StopInternal() {
+  RTC_NOTREACHED() << "DEBUG: RtpTransceiverInterface::StopInternal called";
+}
+
 RTCError RtpTransceiverInterface::SetCodecPreferences(
     rtc::ArrayView<RtpCodecCapability>) {
   RTC_NOTREACHED() << "Not implemented";
@@ -47,4 +64,22 @@ webrtc::RTCError RtpTransceiverInterface::SetOfferedRtpHeaderExtensions(
   return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_OPERATION);
 }
 
+std::vector<RtpHeaderExtensionCapability>
+RtpTransceiverInterface::HeaderExtensionsNegotiated() const {
+  return {};
+}
+
+// TODO(bugs.webrtc.org/11839) Remove default implementations when clients
+// are updated.
+void RtpTransceiverInterface::SetDirection(
+    RtpTransceiverDirection new_direction) {
+  SetDirectionWithError(new_direction);
+}
+
+RTCError RtpTransceiverInterface::SetDirectionWithError(
+    RtpTransceiverDirection new_direction) {
+  RTC_NOTREACHED() << "Default implementation called";
+  return RTCError::OK();
+}
+
 }  // namespace webrtc

api/rtp_transceiver_interface.h

@@ -89,6 +89,16 @@ class RTC_EXPORT RtpTransceiverInterface : public rtc::RefCountInterface {
   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
   virtual bool stopped() const = 0;
 
+  // The stopping attribute indicates that the user has indicated that the
+  // sender of this transceiver will stop sending, and that the receiver will
+  // no longer receive. It is always true if stopped() is true.
+  // If stopping() is true and stopped() is false, it means that the
+  // transceiver's stop() method has been called, but the negotiation with
+  // the other end for shutting down the transceiver is not yet done.
+  // https://w3c.github.io/webrtc-pc/#dfn-stopping-0
+  // TODO(hta): Remove default implementation.
+  virtual bool stopping() const;
+
   // The direction attribute indicates the preferred direction of this
   // transceiver, which will be used in calls to CreateOffer and CreateAnswer.
   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
@@ -99,7 +109,11 @@ class RTC_EXPORT RtpTransceiverInterface : public rtc::RefCountInterface {
   // CreateOffer and CreateAnswer mark the corresponding media descriptions as
   // sendrecv, sendonly, recvonly, or inactive.
   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
-  virtual void SetDirection(RtpTransceiverDirection new_direction) = 0;
+  // TODO(hta): Deprecate SetDirection without error and rename
+  // SetDirectionWithError to SetDirection, remove default implementations.
+  RTC_DEPRECATED virtual void SetDirection(
+      RtpTransceiverDirection new_direction);
+  virtual RTCError SetDirectionWithError(RtpTransceiverDirection new_direction);
 
   // The current_direction attribute indicates the current direction negotiated
   // for this transceiver. If this transceiver has never been represented in an
@@ -114,10 +128,19 @@ class RTC_EXPORT RtpTransceiverInterface : public rtc::RefCountInterface {
   // Exposed in the public interface for use by Chromium.
   virtual absl::optional<RtpTransceiverDirection> fired_direction() const;
 
-  // The Stop method irreversibly stops the RtpTransceiver. The sender of this
+  // Initiates a stop of the transceiver.
-  // transceiver will no longer send, the receiver will no longer receive.
+  // The stop is complete when stopped() returns true.
+  // A stopped transceiver can be reused for a different track.
   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
-  virtual void Stop() = 0;
+  // TODO(hta): Rename to Stop() when users of the non-standard Stop() are
+  // updated.
+  virtual RTCError StopStandard();
+
+  // Stops a transceiver immediately, without waiting for signalling.
+  // This is an internal function, and is exposed for historical reasons.
+  // https://w3c.github.io/webrtc-pc/#dfn-stop-the-rtcrtptransceiver
+  virtual void StopInternal();
+  RTC_DEPRECATED virtual void Stop();
 
   // The SetCodecPreferences method overrides the default codec preferences used
   // by WebRTC for this transceiver.
@@ -133,6 +156,12 @@ class RTC_EXPORT RtpTransceiverInterface : public rtc::RefCountInterface {
   virtual std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToOffer()
       const;
 
+  // Readonly attribute which is either empty if negotation has not yet
+  // happened, or a vector of the negotiated header extensions.
+  // https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface
+  virtual std::vector<RtpHeaderExtensionCapability> HeaderExtensionsNegotiated()
+      const;
+
   // The SetOfferedRtpHeaderExtensions method modifies the next SDP negotiation
   // so that it negotiates use of header extensions which are not kStopped.
   // https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface

api/set_local_description_observer_interface.h

@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2020 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_SET_LOCAL_DESCRIPTION_OBSERVER_INTERFACE_H_
+#define API_SET_LOCAL_DESCRIPTION_OBSERVER_INTERFACE_H_
+
+#include "api/rtc_error.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+// OnSetLocalDescriptionComplete() invokes as soon as
+// PeerConnectionInterface::SetLocalDescription() operation completes, allowing
+// the observer to examine the effects of the operation without delay.
+class SetLocalDescriptionObserverInterface : public rtc::RefCountInterface {
+ public:
+  // On success, |error.ok()| is true.
+  virtual void OnSetLocalDescriptionComplete(RTCError error) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_SET_LOCAL_DESCRIPTION_OBSERVER_INTERFACE_H_

api/stats/rtc_stats_report.h

@@ -84,8 +84,8 @@ class RTC_EXPORT RTCStatsReport : public rtc::RefCountInterface {
   // Removes the stats object from the report, returning ownership of it or null
   // if there is no object with |id|.
   std::unique_ptr<const RTCStats> Take(const std::string& id);
-  // Takes ownership of all the stats in |victim|, leaving it empty.
+  // Takes ownership of all the stats in |other|, leaving it empty.
-  void TakeMembersFrom(rtc::scoped_refptr<RTCStatsReport> victim);
+  void TakeMembersFrom(rtc::scoped_refptr<RTCStatsReport> other);
 
   // Stats iterators. Stats are ordered lexicographically on |RTCStats::id|.
   ConstIterator begin() const;

api/stats/rtcstats_objects.h

@@ -115,6 +115,7 @@ class RTC_EXPORT RTCCodecStats final : public RTCStats {
   RTCCodecStats(const RTCCodecStats& other);
   ~RTCCodecStats() override;
 
+  RTCStatsMember<std::string> transport_id;
   RTCStatsMember<uint32_t> payload_type;
   RTCStatsMember<std::string> mime_type;
   RTCStatsMember<uint32_t> clock_rate;
@@ -419,6 +420,18 @@ class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
   // TODO(hbos): Collect and populate this value for both "audio" and "video",
   // currently not collected for "video". https://bugs.webrtc.org/7065
   RTCStatsMember<double> jitter;
+  RTCStatsMember<double> jitter_buffer_delay;
+  RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
+  RTCStatsMember<uint64_t> total_samples_received;
+  RTCStatsMember<uint64_t> concealed_samples;
+  RTCStatsMember<uint64_t> silent_concealed_samples;
+  RTCStatsMember<uint64_t> concealment_events;
+  RTCStatsMember<uint64_t> inserted_samples_for_deceleration;
+  RTCStatsMember<uint64_t> removed_samples_for_acceleration;
+  RTCStatsMember<double> audio_level;
+  RTCStatsMember<double> total_audio_energy;
+  RTCStatsMember<double> total_samples_duration;
+  RTCStatsMember<int32_t> frames_received;
   // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<double> round_trip_time;
   // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
@@ -441,8 +454,13 @@ class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
   RTCStatsMember<double> gap_loss_rate;
   // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
   RTCStatsMember<double> gap_discard_rate;
+  RTCStatsMember<uint32_t> frame_width;
+  RTCStatsMember<uint32_t> frame_height;
+  RTCStatsMember<uint32_t> frame_bit_depth;
+  RTCStatsMember<double> frames_per_second;
   RTCStatsMember<uint32_t> frames_decoded;
   RTCStatsMember<uint32_t> key_frames_decoded;
+  RTCStatsMember<uint32_t> frames_dropped;
   RTCStatsMember<double> total_decode_time;
   RTCStatsMember<double> total_inter_frame_delay;
   RTCStatsMember<double> total_squared_inter_frame_delay;
@@ -602,7 +620,9 @@ class RTC_EXPORT RTCTransportStats final : public RTCStats {
   ~RTCTransportStats() override;
 
   RTCStatsMember<uint64_t> bytes_sent;
+  RTCStatsMember<uint64_t> packets_sent;
   RTCStatsMember<uint64_t> bytes_received;
+  RTCStatsMember<uint64_t> packets_received;
   RTCStatsMember<std::string> rtcp_transport_stats_id;
   // TODO(hbos): Support enum types? "RTCStatsMember<RTCDtlsTransportState>"?
   RTCStatsMember<std::string> dtls_state;

api/test/DEPS

@@ -11,12 +11,6 @@ specific_include_rules = {
   "fake_constraints\.h": [
     "+rtc_base/string_encode.h",
   ],
-  "loopback_media_transport\.h": [
-    "+rtc_base/async_invoker.h",
-    "+rtc_base/critical_section.h",
-    "+rtc_base/thread.h",
-    "+rtc_base/thread_checker.h",
-  ],
   "neteq_factory_with_codecs\.h": [
     "+system_wrappers/include/clock.h",
   ],
@@ -41,4 +35,7 @@ specific_include_rules = {
   "create_frame_generator\.h": [
     "+system_wrappers/include/clock.h",
   ],
+  "videocodec_test_fixture\.h": [
+    "+media/base/h264_profile_level_id.h"
+  ],
 }

api/test/OWNERS

@@ -1,2 +1,5 @@
+mbonadei@webrtc.org
+sprang@webrtc.org
 srte@webrtc.org
+titovartem@webrtc.org
 

api/test/audio_quality_analyzer_interface.h

@@ -14,7 +14,7 @@
 #include <string>
 
 #include "api/test/stats_observer_interface.h"
-#include "api/test/track_id_stream_label_map.h"
+#include "api/test/track_id_stream_info_map.h"
 
 namespace webrtc {
 namespace webrtc_pc_e2e {
@@ -31,7 +31,7 @@ class AudioQualityAnalyzerInterface : public StatsObserverInterface {
   // stream_id matching. The caller is responsible for ensuring the
   // AnalyzerHelper outlives the instance of the AudioQualityAnalyzerInterface.
   virtual void Start(std::string test_case_name,
-                     TrackIdStreamLabelMap* analyzer_helper) = 0;
+                     TrackIdStreamInfoMap* analyzer_helper) = 0;
 
   // Will be called by the framework at the end of the test. The analyzer
   // has to finalize all its stats and it should report them.

api/test/compile_all_headers.cc

@@ -27,13 +27,17 @@
 // "api/test/videocodec_test_fixture.h"
 // "api/test/videocodec_test_stats.h"
 
+#include "api/test/dummy_peer_connection.h"
 #include "api/test/fake_frame_decryptor.h"
 #include "api/test/fake_frame_encryptor.h"
 #include "api/test/mock_audio_mixer.h"
+#include "api/test/mock_data_channel.h"
 #include "api/test/mock_frame_decryptor.h"
 #include "api/test/mock_frame_encryptor.h"
+#include "api/test/mock_media_stream_interface.h"
 #include "api/test/mock_peer_connection_factory_interface.h"
 #include "api/test/mock_peerconnectioninterface.h"
+#include "api/test/mock_rtp_transceiver.h"
 #include "api/test/mock_rtpreceiver.h"
 #include "api/test/mock_rtpsender.h"
 #include "api/test/mock_transformable_video_frame.h"

api/test/create_peerconnection_quality_test_fixture.cc

@@ -13,6 +13,7 @@
 #include <memory>
 #include <utility>
 
+#include "api/test/time_controller.h"
 #include "test/pc/e2e/peer_connection_quality_test.h"
 
 namespace webrtc {
@@ -21,11 +22,12 @@ namespace webrtc_pc_e2e {
 std::unique_ptr<PeerConnectionE2EQualityTestFixture>
 CreatePeerConnectionE2EQualityTestFixture(
     std::string test_case_name,
+    TimeController& time_controller,
     std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer,
     std::unique_ptr<VideoQualityAnalyzerInterface> video_quality_analyzer) {
   return std::make_unique<PeerConnectionE2EQualityTest>(
-      std::move(test_case_name), std::move(audio_quality_analyzer),
+      std::move(test_case_name), time_controller,
-      std::move(video_quality_analyzer));
+      std::move(audio_quality_analyzer), std::move(video_quality_analyzer));
 }
 
 }  // namespace webrtc_pc_e2e

api/test/create_peerconnection_quality_test_fixture.h

@@ -15,20 +15,25 @@
 
 #include "api/test/audio_quality_analyzer_interface.h"
 #include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/test/time_controller.h"
 #include "api/test/video_quality_analyzer_interface.h"
 
 namespace webrtc {
 namespace webrtc_pc_e2e {
 
 // API is in development. Can be changed/removed without notice.
+
 // Create test fixture to establish test call between Alice and Bob.
 // During the test Alice will be caller and Bob will answer the call.
 // |test_case_name| is a name of test case, that will be used for all metrics
 // reporting.
+// |time_controller| is used to manage all rtc::Thread's and TaskQueue
+// instances. Instance of |time_controller| have to outlive created fixture.
 // Returns a non-null PeerConnectionE2EQualityTestFixture instance.
 std::unique_ptr<PeerConnectionE2EQualityTestFixture>
 CreatePeerConnectionE2EQualityTestFixture(
     std::string test_case_name,
+    TimeController& time_controller,
     std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer,
     std::unique_ptr<VideoQualityAnalyzerInterface> video_quality_analyzer);
 

api/test/create_time_controller.cc

@@ -37,7 +37,8 @@ std::unique_ptr<CallFactoryInterface> CreateTimeControllerBasedCallFactory(
     Call* CreateCall(const Call::Config& config) override {
       if (!module_thread_) {
         module_thread_ = SharedModuleThread::Create(
-            "CallModules", [this]() { module_thread_ = nullptr; });
+            time_controller_->CreateProcessThread("CallModules"),
+            [this]() { module_thread_ = nullptr; });
       }
       return Call::Create(config, time_controller_->GetClock(), module_thread_,
                           time_controller_->CreateProcessThread("Pacer"));

api/test/dummy_peer_connection.h

@@ -36,7 +36,7 @@ class DummyPeerConnection : public PeerConnectionInterface {
 
   bool AddStream(MediaStreamInterface* stream) override { return false; }
   void RemoveStream(MediaStreamInterface* stream) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
 
   RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
@@ -100,17 +100,17 @@ class DummyPeerConnection : public PeerConnectionInterface {
   }
 
   void GetStats(RTCStatsCollectorCallback* callback) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
   void GetStats(
       rtc::scoped_refptr<RtpSenderInterface> selector,
       rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
   void GetStats(
       rtc::scoped_refptr<RtpReceiverInterface> selector,
       rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
   void ClearStatsCache() override {}
 
@@ -145,33 +145,33 @@ class DummyPeerConnection : public PeerConnectionInterface {
     return nullptr;
   }
 
-  void RestartIce() override { FATAL() << "Not implemented"; }
+  void RestartIce() override { RTC_CHECK_NOTREACHED(); }
 
   // Create a new offer.
   // The CreateSessionDescriptionObserver callback will be called when done.
   void CreateOffer(CreateSessionDescriptionObserver* observer,
                    const RTCOfferAnswerOptions& options) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
 
   void CreateAnswer(CreateSessionDescriptionObserver* observer,
                     const RTCOfferAnswerOptions& options) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
 
   void SetLocalDescription(SetSessionDescriptionObserver* observer,
                            SessionDescriptionInterface* desc) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
   void SetRemoteDescription(SetSessionDescriptionObserver* observer,
                             SessionDescriptionInterface* desc) override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
   void SetRemoteDescription(
       std::unique_ptr<SessionDescriptionInterface> desc,
       rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer)
       override {
-    FATAL() << "Not implemented";
+    RTC_CHECK_NOTREACHED();
   }
 
   PeerConnectionInterface::RTCConfiguration GetConfiguration() override {
@@ -194,14 +194,8 @@ class DummyPeerConnection : public PeerConnectionInterface {
     return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
   }
 
-  RTCError SetBitrate(const BitrateParameters& bitrate_parameters) override {
+  void SetAudioPlayout(bool playout) override { RTC_CHECK_NOTREACHED(); }
-    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
+  void SetAudioRecording(bool recording) override { RTC_CHECK_NOTREACHED(); }
-  }
-
-  void SetAudioPlayout(bool playout) override { FATAL() << "Not implemented"; }
-  void SetAudioRecording(bool recording) override {
-    FATAL() << "Not implemented";
-  }
 
   rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid(
       const std::string& mid) override {
@@ -239,9 +233,13 @@ class DummyPeerConnection : public PeerConnectionInterface {
     return false;
   }
 
-  void StopRtcEventLog() { FATAL() << "Not implemented"; }
+  void StopRtcEventLog() { RTC_CHECK_NOTREACHED(); }
 
-  void Close() {}
+  void Close() override {}
+
+  rtc::Thread* signaling_thread() const override {
+    return rtc::Thread::Current();
+  }
 };
 
 static_assert(

api/test/frame_generator_interface.cc

@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2020 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/test/frame_generator_interface.h"
+
+namespace webrtc {
+namespace test {
+
+// static
+const char* FrameGeneratorInterface::OutputTypeToString(
+    FrameGeneratorInterface::OutputType type) {
+  switch (type) {
+    case OutputType::kI420:
+      return "I420";
+    case OutputType::kI420A:
+      return "I420A";
+    case OutputType::kI010:
+      return "I010";
+    case OutputType::kNV12:
+      return "NV12";
+    default:
+      RTC_NOTREACHED();
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc

api/test/frame_generator_interface.h

@@ -32,7 +32,8 @@ class FrameGeneratorInterface {
     absl::optional<VideoFrame::UpdateRect> update_rect;
   };
 
-  enum class OutputType { kI420, kI420A, kI010 };
+  enum class OutputType { kI420, kI420A, kI010, kNV12 };
+  static const char* OutputTypeToString(OutputType type);
 
   virtual ~FrameGeneratorInterface() = default;
 

api/test/mock_data_channel.h

@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2020 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_TEST_MOCK_DATA_CHANNEL_H_
+#define API_TEST_MOCK_DATA_CHANNEL_H_
+
+#include <string>
+
+#include "api/data_channel_interface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDataChannelInterface final
+    : public rtc::RefCountedObject<webrtc::DataChannelInterface> {
+ public:
+  static rtc::scoped_refptr<MockDataChannelInterface> Create() {
+    return new MockDataChannelInterface();
+  }
+
+  MOCK_METHOD(void,
+              RegisterObserver,
+              (DataChannelObserver * observer),
+              (override));
+  MOCK_METHOD(void, UnregisterObserver, (), (override));
+  MOCK_METHOD(std::string, label, (), (const, override));
+  MOCK_METHOD(bool, reliable, (), (const, override));
+  MOCK_METHOD(bool, ordered, (), (const, override));
+  MOCK_METHOD(uint16_t, maxRetransmitTime, (), (const, override));
+  MOCK_METHOD(uint16_t, maxRetransmits, (), (const, override));
+  MOCK_METHOD(absl::optional<int>, maxRetransmitsOpt, (), (const, override));
+  MOCK_METHOD(absl::optional<int>, maxPacketLifeTime, (), (const, override));
+  MOCK_METHOD(std::string, protocol, (), (const, override));
+  MOCK_METHOD(bool, negotiated, (), (const, override));
+  MOCK_METHOD(int, id, (), (const, override));
+  MOCK_METHOD(Priority, priority, (), (const, override));
+  MOCK_METHOD(DataState, state, (), (const, override));
+  MOCK_METHOD(RTCError, error, (), (const, override));
+  MOCK_METHOD(uint32_t, messages_sent, (), (const, override));
+  MOCK_METHOD(uint64_t, bytes_sent, (), (const, override));
+  MOCK_METHOD(uint32_t, messages_received, (), (const, override));
+  MOCK_METHOD(uint64_t, bytes_received, (), (const, override));
+  MOCK_METHOD(uint64_t, buffered_amount, (), (const, override));
+  MOCK_METHOD(void, Close, (), (override));
+  MOCK_METHOD(bool, Send, (const DataBuffer& buffer), (override));
+
+ protected:
+  MockDataChannelInterface() = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_TEST_MOCK_DATA_CHANNEL_H_

api/test/mock_media_stream_interface.h

@@ -0,0 +1,89 @@
+/*
+ *  Copyright 2020 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_TEST_MOCK_MEDIA_STREAM_INTERFACE_H_
+#define API_TEST_MOCK_MEDIA_STREAM_INTERFACE_H_
+
+#include <string>
+
+#include "api/media_stream_interface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioSource final
+    : public rtc::RefCountedObject<AudioSourceInterface> {
+ public:
+  static rtc::scoped_refptr<MockAudioSource> Create() {
+    return new MockAudioSource();
+  }
+
+  MOCK_METHOD(void,
+              RegisterObserver,
+              (ObserverInterface * observer),
+              (override));
+  MOCK_METHOD(void,
+              UnregisterObserver,
+              (ObserverInterface * observer),
+              (override));
+  MOCK_METHOD(SourceState, state, (), (const, override));
+  MOCK_METHOD(bool, remote, (), (const, override));
+  MOCK_METHOD(void, SetVolume, (double volume), (override));
+  MOCK_METHOD(void,
+              RegisterAudioObserver,
+              (AudioObserver * observer),
+              (override));
+  MOCK_METHOD(void,
+              UnregisterAudioObserver,
+              (AudioObserver * observer),
+              (override));
+  MOCK_METHOD(void, AddSink, (AudioTrackSinkInterface * sink), (override));
+  MOCK_METHOD(void, RemoveSink, (AudioTrackSinkInterface * sink), (override));
+  MOCK_METHOD(const cricket::AudioOptions, options, (), (const, override));
+
+ private:
+  MockAudioSource() = default;
+};
+
+class MockAudioTrack final : public rtc::RefCountedObject<AudioTrackInterface> {
+ public:
+  static rtc::scoped_refptr<MockAudioTrack> Create() {
+    return new MockAudioTrack();
+  }
+
+  MOCK_METHOD(void,
+              RegisterObserver,
+              (ObserverInterface * observer),
+              (override));
+  MOCK_METHOD(void,
+              UnregisterObserver,
+              (ObserverInterface * observer),
+              (override));
+  MOCK_METHOD(std::string, kind, (), (const, override));
+  MOCK_METHOD(std::string, id, (), (const override));
+  MOCK_METHOD(bool, enabled, (), (const, override));
+  MOCK_METHOD(bool, set_enabled, (bool enable), (override));
+  MOCK_METHOD(TrackState, state, (), (const, override));
+  MOCK_METHOD(AudioSourceInterface*, GetSource, (), (const, override));
+  MOCK_METHOD(void, AddSink, (AudioTrackSinkInterface * sink), (override));
+  MOCK_METHOD(void, RemoveSink, (AudioTrackSinkInterface * sink), (override));
+  MOCK_METHOD(bool, GetSignalLevel, (int* level), (override));
+  MOCK_METHOD(rtc::scoped_refptr<AudioProcessorInterface>,
+              GetAudioProcessor,
+              (),
+              (override));
+
+ private:
+  MockAudioTrack() = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_TEST_MOCK_MEDIA_STREAM_INTERFACE_H_

api/test/mock_peer_connection_factory_interface.h

@@ -22,7 +22,7 @@ namespace webrtc {
 class MockPeerConnectionFactoryInterface final
     : public rtc::RefCountedObject<webrtc::PeerConnectionFactoryInterface> {
  public:
-  rtc::scoped_refptr<MockPeerConnectionFactoryInterface> Create() {
+  static rtc::scoped_refptr<MockPeerConnectionFactoryInterface> Create() {
     return new MockPeerConnectionFactoryInterface();
   }
 
@@ -32,6 +32,11 @@ class MockPeerConnectionFactoryInterface final
               (const PeerConnectionInterface::RTCConfiguration&,
                PeerConnectionDependencies),
               (override));
+  MOCK_METHOD(RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>>,
+              CreatePeerConnectionOrError,
+              (const PeerConnectionInterface::RTCConfiguration&,
+               PeerConnectionDependencies),
+              (override));
   MOCK_METHOD(rtc::scoped_refptr<PeerConnectionInterface>,
               CreatePeerConnection,
               (const PeerConnectionInterface::RTCConfiguration&,

api/test/mock_peerconnectioninterface.h

@@ -167,7 +167,6 @@ class MockPeerConnectionInterface
               (const std::vector<cricket::Candidate>&),
               (override));
   MOCK_METHOD(RTCError, SetBitrate, (const BitrateSettings&), (override));
-  MOCK_METHOD(RTCError, SetBitrate, (const BitrateParameters&), (override));
   MOCK_METHOD(void, SetAudioPlayout, (bool), (override));
   MOCK_METHOD(void, SetAudioRecording, (bool), (override));
   MOCK_METHOD(rtc::scoped_refptr<DtlsTransportInterface>,
@@ -194,6 +193,7 @@ class MockPeerConnectionInterface
   MOCK_METHOD(void, StopRtcEventLog, (), (override));
   MOCK_METHOD(void, Close, (), (override));
 
+  // RingRTC changes for added methods
   MOCK_METHOD0(CreateSharedIceGatherer,
                rtc::scoped_refptr<webrtc::IceGathererInterface>());
   MOCK_METHOD1(UseSharedIceGatherer,

api/test/mock_rtp_transceiver.h

@@ -0,0 +1,85 @@
+/*
+ *  Copyright 2020 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_TEST_MOCK_RTP_TRANSCEIVER_H_
+#define API_TEST_MOCK_RTP_TRANSCEIVER_H_
+
+#include <string>
+#include <vector>
+
+#include "api/rtp_transceiver_interface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockRtpTransceiver final
+    : public rtc::RefCountedObject<RtpTransceiverInterface> {
+ public:
+  static rtc::scoped_refptr<MockRtpTransceiver> Create() {
+    return new MockRtpTransceiver();
+  }
+
+  MOCK_METHOD(cricket::MediaType, media_type, (), (const, override));
+  MOCK_METHOD(absl::optional<std::string>, mid, (), (const, override));
+  MOCK_METHOD(rtc::scoped_refptr<RtpSenderInterface>,
+              sender,
+              (),
+              (const, override));
+  MOCK_METHOD(rtc::scoped_refptr<RtpReceiverInterface>,
+              receiver,
+              (),
+              (const, override));
+  MOCK_METHOD(bool, stopped, (), (const, override));
+  MOCK_METHOD(bool, stopping, (), (const, override));
+  MOCK_METHOD(RtpTransceiverDirection, direction, (), (const, override));
+  MOCK_METHOD(void,
+              SetDirection,
+              (RtpTransceiverDirection new_direction),
+              (override));
+  MOCK_METHOD(RTCError,
+              SetDirectionWithError,
+              (RtpTransceiverDirection new_direction),
+              (override));
+  MOCK_METHOD(absl::optional<RtpTransceiverDirection>,
+              current_direction,
+              (),
+              (const, override));
+  MOCK_METHOD(absl::optional<RtpTransceiverDirection>,
+              fired_direction,
+              (),
+              (const, override));
+  MOCK_METHOD(RTCError, StopStandard, (), (override));
+  MOCK_METHOD(void, StopInternal, (), (override));
+  MOCK_METHOD(void, Stop, (), (override));
+  MOCK_METHOD(RTCError,
+              SetCodecPreferences,
+              (rtc::ArrayView<RtpCodecCapability> codecs),
+              (override));
+  MOCK_METHOD(std::vector<RtpCodecCapability>,
+              codec_preferences,
+              (),
+              (const, override));
+  MOCK_METHOD(std::vector<RtpHeaderExtensionCapability>,
+              HeaderExtensionsToOffer,
+              (),
+              (const, override));
+  MOCK_METHOD(webrtc::RTCError,
+              SetOfferedRtpHeaderExtensions,
+              (rtc::ArrayView<const RtpHeaderExtensionCapability>
+                   header_extensions_to_offer),
+              (override));
+
+ private:
+  MockRtpTransceiver() = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_TEST_MOCK_RTP_TRANSCEIVER_H_

api/test/mock_video_encoder.h

@@ -22,9 +22,7 @@ class MockEncodedImageCallback : public EncodedImageCallback {
  public:
   MOCK_METHOD(Result,
               OnEncodedImage,
-              (const EncodedImage& encodedImage,
+              (const EncodedImage&, const CodecSpecificInfo*),
-               const CodecSpecificInfo*,
-               const RTPFragmentationHeader*),
               (override));
   MOCK_METHOD(void, OnDroppedFrame, (DropReason reason), (override));
 };

api/test/neteq_simulator_factory.cc

@@ -21,6 +21,24 @@
 
 namespace webrtc {
 namespace test {
+namespace {
+NetEqTestFactory::Config convertConfig(
+    const NetEqSimulatorFactory::Config& simulation_config,
+    absl::string_view replacement_audio_filename) {
+  NetEqTestFactory::Config config;
+  config.replacement_audio_file = std::string(replacement_audio_filename);
+  config.max_nr_packets_in_buffer = simulation_config.max_nr_packets_in_buffer;
+  config.initial_dummy_packets = simulation_config.initial_dummy_packets;
+  config.skip_get_audio_events = simulation_config.skip_get_audio_events;
+  config.field_trial_string = simulation_config.field_trial_string;
+  config.output_audio_filename = simulation_config.output_audio_filename;
+  config.pythonplot = simulation_config.python_plot_filename.has_value();
+  config.plot_scripts_basename = simulation_config.python_plot_filename;
+  config.textlog = simulation_config.text_log_filename.has_value();
+  config.textlog_filename = simulation_config.text_log_filename;
+  return config;
+}
+}  // namespace
 
 NetEqSimulatorFactory::NetEqSimulatorFactory()
     : factory_(std::make_unique<NetEqTestFactory>()) {}
@@ -31,13 +49,8 @@ std::unique_ptr<NetEqSimulator> NetEqSimulatorFactory::CreateSimulatorFromFile(
     absl::string_view event_log_filename,
     absl::string_view replacement_audio_filename,
     Config simulation_config) {
-  NetEqTestFactory::Config config;
+  NetEqTestFactory::Config config =
-  config.replacement_audio_file = std::string(replacement_audio_filename);
+      convertConfig(simulation_config, replacement_audio_filename);
-  config.max_nr_packets_in_buffer = simulation_config.max_nr_packets_in_buffer;
-  config.initial_dummy_packets = simulation_config.initial_dummy_packets;
-  config.skip_get_audio_events = simulation_config.skip_get_audio_events;
-  config.field_trial_string = simulation_config.field_trial_string;
-  config.output_audio_filename = simulation_config.output_audio_filename;
   return factory_->InitializeTestFromFile(
       std::string(event_log_filename), simulation_config.neteq_factory, config);
 }
@@ -47,12 +60,8 @@ NetEqSimulatorFactory::CreateSimulatorFromString(
     absl::string_view event_log_file_contents,
     absl::string_view replacement_audio_filename,
     Config simulation_config) {
-  NetEqTestFactory::Config config;
+  NetEqTestFactory::Config config =
-  config.replacement_audio_file = std::string(replacement_audio_filename);
+      convertConfig(simulation_config, replacement_audio_filename);
-  config.max_nr_packets_in_buffer = simulation_config.max_nr_packets_in_buffer;
-  config.initial_dummy_packets = simulation_config.initial_dummy_packets;
-  config.skip_get_audio_events = simulation_config.skip_get_audio_events;
-  config.field_trial_string = simulation_config.field_trial_string;
   return factory_->InitializeTestFromString(
       std::string(event_log_file_contents), simulation_config.neteq_factory,
       config);