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chromium-webrtc-autoroll dd6588d7c1 Roll chromium_revision e19703fdfe..b133b38aa4 (794735:794854) 
Change log: e19703fdfe..b133b38aa4
Full diff: e19703fdfe..b133b38aa4

Changed dependencies
* src/base: ee0001e51e..e6c4077b24
* src/build: 6985835c9b..2e781420a9
* src/ios: 09261a3621..47c35b1e27
* src/testing: dc9b1969e7..165367ff94
* src/third_party: 6e4870ba40..a4b288cba7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/cf93e1de9e..ac60992d41
* src/third_party/depot_tools: a0c3f906bc..486f1812ef
* src/third_party/perfetto: 06e30bae5c..f8ccdc63a1
* src/tools: 042bd4c12b..4def294a52
DEPS diff: e19703fdfe..b133b38aa4/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ic39c96f2a53bc614a28455c6893bde669d6208da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180925
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#31855}
2020-08-05 04:54:26 +00:00
api
Reland "Add EncodedImageCallback::OnEncodedImage without RTPFragmentationHeader"
2020-08-04 12:17:06 +00:00
audio
Remove unused critical section includes.
2020-07-16 13:52:28 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-03 15:45:41 +00:00
common_audio
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
common_video
Ignore RTPFragmentationHeader when rewriting H264 SPS
2020-07-27 09:42:25 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
examples
Implemented Android Demo Application for VoIP API
2020-07-21 16:34:22 +00:00
logging
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
media
Reducing threshold for usrsctp "buffer low" callback.
2020-08-04 20:08:06 +00:00
modules
Fix undeclared dependencies on ole32.lib and user32.lib
2020-08-05 01:27:36 +00:00
p2p
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
pc
sdp: reject sdp with malformed b= lines
2020-08-03 18:21:14 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Delete old name ProcessAllMessageQueues
2020-08-03 11:24:17 +00:00
rtc_tools
Revert "Moved VideoReceiveStream::Decoder::decoder_factory to VideoReceiveStream::Config::decoder_factory."
2020-08-03 15:45:41 +00:00
sdk
Move Android network monitor to separate target/directory.
2020-08-04 20:16:07 +00:00
stats
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
test
Add networks stats collector from PeerConnection GetStats API
2020-08-04 09:05:14 +00:00
tools_webrtc
Fix "Assignment had no effect." error during chromium roll.
2020-07-30 12:57:20 +00:00
video
Reland "Only enable conference mode simulcast allocations with flag enabled"
2020-08-04 10:30:08 +00:00
.clang-format
…
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .clangd to .gitignore
2019-10-28 12:27:50 +00:00
.gn
Reenable libaom decoder by default
2020-03-18 18:04:41 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md
Use absl_deps in order to preapre to the Abseil component build release.
2020-06-08 12:59:40 +00:00
AUTHORS
Explicitly set presentation style to full screen, because,
2020-07-17 07:15:36 +00:00
BUILD.gn
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +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
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
DEPS
Roll chromium_revision e19703fdfe..b133b38aa4 (794735:794854)
2020-08-05 04:54:26 +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
…
native-api.md
Make transient suppression optionally excludable via defines
2020-04-02 11:44:07 +00:00
OWNERS
Remove phoglund as root owner.
2020-03-30 12:15:56 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +00:00
presubmit_test.py
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
PRESUBMIT.py
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
Fix link in documentation. (take 2)
2020-04-16 11:08:43 +00:00
style-guide.md
C++ style: We don't allow designated initializers
2020-06-03 09:11:09 +00:00
WATCHLISTS
Remove benwright@webrtc.org from WATCHLISTS
2020-01-31 18:46:52 +00:00
webrtc_lib_link_test.cc
Rewrite the lib link test to just be a binary.
2019-10-18 07:42:20 +00:00
webrtc.gni
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
whitespace.txt
Trigger CI bots.
2020-07-15 17:50:55 +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 here 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

Description
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%