signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Per Kjellander f7cb16ff51 Check input parameter in RemoteEstimatorProxy::IncomingPacket without lock 
Also inlined RemoteEstimatorProxy::OnPacketArrival

BUG=NONE

Change-Id: I61c94eafb41ea269baeeb0ef13add79672a1488d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151909
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29109}
2019-09-09 11:51:12 +00:00
api Prepare to move packet_socket_factory to api/. 2019-09-06 09:09:02 +00:00
audio Deprecate SingleThreadedTaskQueueForTesting class. 2019-09-03 10:31:30 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Deprecate SingleThreadedTaskQueueForTesting class. 2019-09-03 10:31:30 +00:00
common_audio Remove rtc_use_lto GN arg. 2019-08-20 14:00:49 +00:00
common_video Add new FrameRateEstimator utility class for more precis FPS estimation. 2019-08-14 12:15:06 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM 2019-09-09 08:10:41 +00:00
logging Delete deprecated rtc_event_log_factory_interface.h 2019-09-05 08:57:36 +00:00
media Reland "New build target p2p:stun_types" 2019-09-06 10:14:38 +00:00
modules Check input parameter in RemoteEstimatorProxy::IncomingPacket without lock 2019-09-09 11:51:12 +00:00
p2p Use obfuscated IPs in logging in p2p/ and pc/. 2019-09-06 21:51:56 +00:00
pc Make payload type demux conditional on media direction 2019-09-07 00:08:51 +00:00
resources Use the AEC3 high-pass filter for the whole APM 2019-08-23 20:04:10 +00:00
rtc_base Implement stable rate support in SimulcastRateAllocator 2019-09-06 11:17:30 +00:00
rtc_tools Revert "Always create output_dir in setup_apprtc.py." 2019-09-04 18:19:12 +00:00
sdk Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM 2019-09-09 08:10:41 +00:00
stats Add support for RTCTransportStats.selectedCandidatePairChanges 2019-08-28 13:22:08 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Add helper functions to convert between integer milliseconds and fixed-point seconds. 2019-08-16 14:49:46 +00:00
test Fix NetworkEmulationManagerTest.ThroughputStats flakiness. 2019-09-06 15:07:01 +00:00
tools_webrtc Add license for android_ndk 2019-08-28 14:57:28 +00:00
video New class VideoReceiver2, a trimmed-down vcm::VideoReceiver 2019-09-09 11:23:54 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Let git-hyper-blame ignore new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH. 2019-09-04 18:49:28 +00:00
.gn Remove last mention of ortc from the codebase. 2019-05-25 07:28:05 +00:00
.vpython Add vpython dependencies needed to run presubmit tests on LUCI 2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md Allowing buffering a LNTF (loss notification) feedback message in RTCPSender 2019-06-03 16:28:34 +00:00
AUTHORS Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5." 2019-09-06 05:36:23 +00:00
BUILD.gn Make the RtpHeaderParserImpl available to tests and tools only. 2019-08-29 15:56:40 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit 2018-10-25 08:19:53 +00:00
common_types.h Format almost everything. 2019-07-08 13:45:15 +00:00
DEPS Roll chromium_revision b67ee864bb..82910f739a (694439:694601) 2019-09-08 00:34:51 +00:00
ENG_REVIEW_OWNERS Enforce LGTM from owners of depends-on paths in DEPS via presubmit. 2018-09-28 12:49:54 +00:00
LICENSE Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
license_template.txt Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
native-api.md Delete unused I420 "codec" 2018-12-18 12:30:58 +00:00
OWNERS Add juberti@ to webrtc root owners 2019-05-17 18:11:58 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +00:00
PRESUBMIT.py Disable RunPythonTests on rtc_tools. 2019-08-06 12:48:33 +00:00
pylintrc Fixing py lint errors 2018-07-23 15:28:48 +00:00
README.chromium Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README.md Tell users where they can find the native API headers 2017-11-14 10:36:46 +00:00
style-guide.md Remove rule that discourages passing optional by const reference 2019-02-05 11:58:05 +00:00
WATCHLISTS Add saza to audio watchlists 2019-09-03 14:55:43 +00:00
webrtc.gni Remove rtc_use_lto GN arg. 2019-08-20 14:00:49 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info