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chromium-webrtc-autoroll f9a5561de7 Roll chromium_revision ee5dfb2215..34f99c21a3 (629907:630023) 
Change log: ee5dfb2215..34f99c21a3
Full diff: ee5dfb2215..34f99c21a3

Changed dependencies
* src/base: 1eb0493202..d029fef0da
* src/build: a4d3fbeca9..a721f3327e
* src/ios: e02d2524ee..6a1390bde0
* src/testing: d200b6e601..4b50bc6568
* src/third_party: 1242705f8e..399bdf2333
* src/third_party/libvpx/source/libvpx: cde3da57b9..ce4336c2ab
* src/tools: 9dd2ebe4c4..6d07835ab0
DEPS diff: ee5dfb2215..34f99c21a3/DEPS

No update to Clang.

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

Change-Id: I9623171b736d965610cc87d624e0e6c00e1f725d
Reviewed-on: https://webrtc-review.googlesource.com/c/121900
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@{#26592}
2019-02-07 20:23:43 +00:00
api
Delete deprecated method PeerConnectionFactory::CreateVideoSource
2019-02-07 14:24:02 +00:00
audio
Propagate base minimum delay to audio_receiver_stream
2019-02-06 11:07:42 +00:00
build_overrides
Disable hermetic toolchain when building on macOS 10.14.
2018-12-07 13:57:28 +00:00
call
Propagate base minimum delay to audio_receiver_stream
2019-02-06 11:07:42 +00:00
common_audio
Stop using Googletest legacy APIs.
2019-01-31 13:23:33 +00:00
common_video
Always add/rewrite VUI and set max_num_reorder_frames to 0.
2019-02-04 09:18:46 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Remove unused file.
2019-02-06 10:24:07 +00:00
logging
Add events processing to GetIceEvents.
2019-02-06 14:14:09 +00:00
media
Reland: Remove dead code from stream_params.h
2019-02-06 19:18:15 +00:00
modules
Define RtpGenericFrameDescriptorExtension00
2019-02-07 16:50:18 +00:00
p2p
Remove TCPPort incoming_only_ member.
2019-02-04 19:46:30 +00:00
pc
Also check the pending remote description when generating MIDs for legacy remote offers
2019-02-07 20:22:38 +00:00
resources
Reland of Opus multistream.
2019-01-29 12:16:19 +00:00
rtc_base
Add missing operator= and extra methods to the SamplesStatsCounter.
2019-02-07 12:59:17 +00:00
rtc_tools
Do not build rtp_generator in Chromium builds.
2019-02-04 14:48:58 +00:00
sdk
Delete deprecated method PeerConnectionFactory::CreateVideoSource
2019-02-07 14:24:02 +00:00
stats
Expose video freeze metrics in GetStats.
2019-02-04 09:58:08 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Makes Clock interface fully mutable.
2019-01-31 08:12:05 +00:00
test
Introduce PeerConnectionE2EQualityTestFixture implementation.
2019-02-07 17:07:39 +00:00
tools_webrtc
Pause rolling buildtools
2019-02-07 08:38:52 +00:00
video
Add owned data buffer to EncodedImage
2019-02-07 13:11:47 +00:00
.clang-format
…
.git-blame-ignore-revs
Let git-hyper-blame ignore format commit.
2018-06-20 09:26:44 +00:00
.gitignore
Add Visual Studio Code project folder to gitignore file.
2019-01-21 18:42:33 +00:00
.gn
Re-enable gtest absl pretty printers.
2018-08-13 13:54:05 +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
Allow use of functions in absl/algorithms
2019-01-28 18:42:16 +00:00
AUTHORS
Add ',' between elements in RTCStatsReport::ToJson
2019-01-08 16:05:23 +00:00
BUILD.gn
Remove HAVE_WEBRTC_VOICE.
2019-02-06 18:39:45 +00:00
CODE_OF_CONDUCT.md
…
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
Delete constant RTP_PAYLOAD_NAME_SIZE
2019-01-25 07:59:52 +00:00
DEPS
Roll chromium_revision ee5dfb2215..34f99c21a3 (629907:630023)
2019-02-07 20:23:43 +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
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Clean up root OWNERS.
2018-11-09 14:23:59 +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
Add PRESUBMIT to enforce usage of new Googletest APIs.
2019-02-06 11:06:02 +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
Modify pc/ WATCHLISTS definition
2019-01-31 22:09:16 +00:00
webrtc.gni
Override default manifest from Chromium in WebRTC.
2019-01-24 13:00:49 +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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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