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chromium-webrtc-autoroll 2d438a864a Roll chromium_revision e0496bdb29..4e516821cf (750459:750566) 
Change log: e0496bdb29..4e516821cf
Full diff: e0496bdb29..4e516821cf

Changed dependencies
* src/base: d7ebae1454..8f95ba94f4
* src/build: 32206c0b16..15cf15f2f8
* src/ios: f12d0470cd..be96ee7f0d
* src/testing: 7a7204ef2e..47e23d6c9d
* src/third_party: a1f080c827..0364b335e4
* src/third_party/depot_tools: c48fb8427c..8cf12290ba
* src/tools: 3db8a5e811..20f9187197
DEPS diff: e0496bdb29..4e516821cf/DEPS

No update to Clang.

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

Change-Id: I8468d77d9c5a2cc619509433ad72aad3c0303a6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170582
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@{#30804}
2020-03-16 16:38:27 +00:00
api
Adds UpdateConfig to SimulatedNetwork
2020-03-16 15:58:43 +00:00
audio
Clamp stable target bitrate to min/max allocated bitrate.
2020-03-12 13:51:10 +00:00
build_overrides
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
call
Adds UpdateConfig to SimulatedNetwork
2020-03-16 15:58:43 +00:00
common_audio
Optimizations and refactoring of the APM 3-band split filter
2020-02-24 13:19:14 +00:00
common_video
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Rename index.md to README.md to make it automatically show up
2020-03-06 10:43:51 +00:00
examples
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
logging
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
media
RtpTransceiverInterface: add header_extensions_to_offer()
2020-03-16 13:16:42 +00:00
modules
APM: Replace most usages of AudioFrame with a stream interface
2020-03-16 11:51:47 +00:00
p2p
Make Connection::id() const
2020-03-05 09:30:18 +00:00
pc
remove mslabel and mslabel ssrc-specific attributes
2020-03-16 14:01:24 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Revert "Leverage dispatch_queue_create_with_target when possible."
2020-03-13 08:02:34 +00:00
rtc_tools
Write protos as binary.
2020-03-12 09:43:57 +00:00
sdk
remove mslabel and mslabel ssrc-specific attributes
2020-03-16 14:01:24 +00:00
stats
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +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 RTC_EXPORT to webrtc::Clock
2020-03-10 23:00:21 +00:00
test
Ensures that all simulated TCP packets are at least 4 bytes.
2020-03-12 17:42:13 +00:00
tools_webrtc
Use luci-go isolate client in mb.py
2020-03-16 07:44:53 +00:00
video
Make video quality tests to always take a fixed duration
2020-03-16 15:38:27 +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
Add Av1 Decoder wrapper behind a build flag
2020-03-11 11:20:56 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md
Fix typo in abseil-in-webrtc.md.
2019-12-18 14:27:34 +00:00
AUTHORS
Fix: rename ms_per_buffer to buffer_duration
2020-03-16 11:04:20 +00:00
BUILD.gn
Fix 'all' build on non Android platforms.
2020-03-11 08:24:27 +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
Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"
2020-02-07 08:23:58 +00:00
DEPS
Roll chromium_revision e0496bdb29..4e516821cf (750459:750566)
2020-03-16 16:38:27 +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
…
license_template.txt
…
native-api.md
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Add #COMPONENT to WebRTC.
2019-10-08 12:20:39 +00:00
PATENTS
…
presubmit_test_mocks.py
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +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
Fix public_deps presubmit and gn format fighting each other.
2020-01-30 11:22:46 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
…
README.md
Tell users where they can find the native API headers
2017-11-14 10:36:46 +00:00
style-guide.md
Add guidance to style guide how to reference a bug in a TODO
2019-12-11 11:55:52 +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
Reformat GN files.
2020-01-21 12:13:11 +00:00
whitespace.txt
Whitespace change
2020-01-28 07:53:15 +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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