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chromium-webrtc-autoroll be4fcb63a2 Roll chromium_revision 96a2369865..b6340b5158 (680738:681123) 
Change log: 96a2369865..b6340b5158
Full diff: 96a2369865..b6340b5158

Changed dependencies
* src/base: b5492349e9..7faff4463c
* src/build: a558f2657b..1a0a4bc2f1
* src/ios: ddb87399cb..5dea9934e6
* src/testing: 992203660d..9d70163229
* src/third_party: adf7051fd6..55bfbb1970
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/3dd1372096..c581761466
* src/third_party/depot_tools: 34cc05018d..365720e237
* src/third_party/ffmpeg: fff11548e7..e0e3133c40
* src/third_party/libvpx/source/libvpx: bb407a27b2..18d309c127
* src/tools: ce748f252c..82f0e4397b
DEPS diff: 96a2369865..b6340b5158/DEPS

No update to Clang.

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

Change-Id: Ied503855289ffa6ac3cc17077c404e0eced4e056
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146921
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@{#28687}
2019-07-26 03:28:10 +00:00
api
Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
2019-07-24 16:47:13 +00:00
audio
Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
2019-07-24 16:47:13 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Don't crash when simulcast layer count is different from RID count
2019-07-24 15:23:44 +00:00
common_audio
Format almost everything.
2019-07-08 13:45:15 +00:00
common_video
Format almost everything.
2019-07-08 13:45:15 +00:00
crypto
Adding new top-level directory crypto/
2019-03-08 00:35:05 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Revert "Don't use all_dependent_configs for sdk frameworks"
2019-07-22 08:46:36 +00:00
logging
Add usage message to rtc_event_log2rtp_dump.
2019-07-22 12:29:39 +00:00
media
Add a field trial for older applications to reduce the simulcast layer count
2019-07-24 14:41:32 +00:00
modules
Add option to remove transport sequence number from FEC packet calculation
2019-07-25 22:47:27 +00:00
p2p
Move datagram_dtls_adaptor from p2p/base/ to pc/
2019-07-10 18:54:20 +00:00
pc
Add option to remove transport sequence number from FEC packet calculation
2019-07-25 22:47:27 +00:00
resources
Cleanup of resources from removed remote bitrate estimate test framework.
2019-06-18 10:22:01 +00:00
rtc_base
Revert "Don't use all_dependent_configs for sdk frameworks"
2019-07-22 08:46:36 +00:00
rtc_tools
Delete rtc_tools/barcode_tools.
2019-07-25 16:54:36 +00:00
sdk
Revert "Don't use all_dependent_configs for sdk frameworks"
2019-07-22 08:46:36 +00:00
stats
Format almost everything.
2019-07-08 13:45:15 +00:00
style-guide
…
system_wrappers
Format almost everything.
2019-07-08 13:45:15 +00:00
test
Revert "Record audio/video bytes sent in analyzer stream stats."
2019-07-23 13:24:42 +00:00
tools_webrtc
Switch neteq_rtpplay into an executable.
2019-07-25 08:45:21 +00:00
video
Migrate 2 test with Vp9 SVC test on PC framework
2019-07-25 12:51:43 +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 Visual Studio Code project folder to gitignore file.
2019-01-21 18:42:33 +00:00
.gn
Remove last mention of ortc from the codebase.
2019-05-25 07:28:05 +00:00
.vpython
…
abseil-in-webrtc.md
Allowing buffering a LNTF (loss notification) feedback message in RTCPSender
2019-06-03 16:28:34 +00:00
AUTHORS
Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly
2019-06-28 19:12:14 +00:00
BUILD.gn
Migrate WebRTC test infra to ABSL_FLAG.
2019-07-19 06:54:04 +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
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision 96a2369865..b6340b5158 (680738:681123)
2019-07-26 03:28:10 +00:00
ENG_REVIEW_OWNERS
…
LICENSE
…
license_template.txt
…
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
…
presubmit_test_mocks.py
…
presubmit_test.py
…
PRESUBMIT.py
Convert file objects to strings, before passing to PresubmitNotifyResult
2019-06-26 11:31:21 +00:00
pylintrc
…
README.chromium
…
README.md
…
style-guide.md
Remove rule that discourages passing optional by const reference
2019-02-05 11:58:05 +00:00
WATCHLISTS
Remove myself from OWNERS in a few places.
2019-06-10 07:57:46 +00:00
webrtc.gni
Migrate WebRTC test infra to ABSL_FLAG.
2019-07-19 06:54:04 +00:00
whitespace.txt
…

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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