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chromium-webrtc-autoroll 26ab9d6855 Roll chromium_revision db92e07547..8939017df7 (656805:657653) 
Change log: db92e07547..8939017df7
Full diff: db92e07547..8939017df7

Changed dependencies
* src/base: 539846dc1d..6e1e13bf3e
* src/build: ad74ef2f83..ea2296fbae
* src/ios: b76c091b33..e6bde469a1
* src/testing: 90baaa8ad6..20ee8223e1
* src/third_party: e470fd1a92..408254f68d
* src/third_party/android_build_tools/aapt2: XPNW95mgY7ws_5lNsyjlq7DowuughMNsRIGuGCT0basC..j6U3mv7-KG3PSDtVvTwycWzjwvFR1_sSdA540AYxpucC
* src/third_party/android_build_tools/bundletool: Z272op8PxTBt5cUJ8aE0NXam_SO7tp-0T0R1woZ0XN0C..bGlR4jA25RgxNi_eSTqm3lX-DvXyHELRfoYIWkmfY1EC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/cca9447f62..f8847c1a94
* src/third_party/depot_tools: ccd2b4da9a..e7f0b4c62c
* src/third_party/ffmpeg: 4500d7f55f..90cf969d61
* src/third_party/google-truth: 4d6fe892fc3150ab40ef1d619baf0038859eb6d2..0VVeotkT0RWtPio6D5NOjDWUwgzDXEbOlqAdmRZ4ku4C
* src/third_party/libvpx/source/libvpx: 3fd96f7d7d..1cbcb820ac
* src/tools: b6d9e26128..f3fca3630c
DEPS diff: db92e07547..8939017df7/DEPS

Clang version changed 359912:360094
Details: db92e07547..8939017df7/tools/clang/scripts/update.py

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

Change-Id: I7dca4da39bfaf3fe356a7c6e66db3fecc8d94ee1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135647
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27878}
2019-05-08 10:49:49 +00:00
api
Do not use absl::flat_hash_map in DefaultVideoQualityAnalyzer.
2019-05-08 10:23:59 +00:00
audio
Ensure that we always set values for min and max audio bitrate.
2019-05-03 13:45:43 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Remove packets from RtpPacketHistory if acked via TransportFeedback
2019-05-07 18:18:02 +00:00
common_audio
Add ability to play audio in circle for TestAudioDevice wav file capturer
2019-04-16 15:33:03 +00:00
common_video
Remove deprecated interface in I420BufferInterface::GetI420
2019-05-03 12:16:34 +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
Remove deprecated interface in I420BufferInterface::GetI420
2019-05-03 12:16:34 +00:00
logging
Add FILE* constructors to RtcEventLogOutputFile
2019-05-06 08:09:24 +00:00
media
Add missing overrides to QualityTestVideoEncoder
2019-05-08 08:01:49 +00:00
modules
Add lifetime concealment stats to NetEqStatsPlotter.
2019-05-08 08:40:08 +00:00
p2p
Factor out the fake port allocator in build.
2019-04-23 20:47:38 +00:00
pc
Revert "Version 2 "Refactoring DataContentDescription class""
2019-05-05 19:00:13 +00:00
resources
RNN VAD: clean-up unit tests
2019-04-29 12:55:02 +00:00
rtc_base
Add ability to configure quality scaler settings through field trial.
2019-05-06 13:20:27 +00:00
rtc_tools
Deprecating injection of event log into GoogCC factory.
2019-05-02 11:05:17 +00:00
sdk
Add support for ignoring errors encountered while configuring preferred attributes of an audio session.
2019-05-08 07:21:12 +00:00
stats
Expose new audio stats on the API
2019-05-03 10:10:15 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Delete unneeded direct includes of common_types.h
2019-04-01 07:18:13 +00:00
test
Do not use absl::flat_hash_map in DefaultVideoQualityAnalyzer.
2019-05-08 10:23:59 +00:00
tools_webrtc
Remove loopback video quality test from configs
2019-04-12 12:32:42 +00:00
video
Rename "average_freeze_duration" metric to "freeze_duration_average"
2019-05-08 08:22:48 +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 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
Bump iOS min supported version to 10.0
2019-03-07 13:08:17 +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
Ban absl::StrSplit and absl::StrJoin
2019-02-26 00:45:11 +00:00
AUTHORS
Import proto_library.gni when rtc_enable_protobuf is true
2019-02-27 09:56:42 +00:00
BUILD.gn
Replacing SequencedTaskChecker with SequenceChecker.
2019-04-09 12:28:04 +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
Move some RTP-related observers from common_types.h
2019-05-08 06:59:23 +00:00
DEPS
Roll chromium_revision db92e07547..8939017df7 (656805:657653)
2019-05-08 10:49:49 +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
Removing niklas.enbom@webrtc.org from WATCHLIST and OWNERS
2019-04-23 14:51:24 +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
Do not run CheckNoStreamUsageIsAdded on tests.
2019-05-07 15:52:35 +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 peah to more audio watchlists
2019-04-26 12:45:06 +00:00
webrtc.gni
Testing no /DUNICODE assumptions with Win more configs bots.
2019-03-28 08:46:37 +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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