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philipel 57ec685446 Unwrap TL0 pic index to avoid having to work with a wrapped number. 
This is to avoid clearing the |gof_info_| map when there are jumps in the
tl0 pic index.

Bug: chromium:855211
Change-Id: I762557070d65b3c535cb9a49498975bcd9c2c485
Reviewed-on: https://webrtc-review.googlesource.com/86943
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23872}
2018-07-06 13:28:38 +00:00
api
AEC3: Slower adaptation of main filter
2018-07-05 14:37:27 +00:00
audio
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
build_overrides
Add phoglund@ to various OWNERS and remove kjellander@
2017-10-19 09:21:12 +00:00
call
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
common_audio
Removing warning suppression flags in common_audio/.
2018-07-04 11:13:09 +00:00
common_video
Add bit depth information to PlanarYuvBuffer
2018-06-26 20:23:37 +00:00
data
WebRTC: Replace ProjectRootPath by ResourcePath
2016-11-22 18:43:05 +00:00
examples
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
infra
Flip luci.webrtc.try to production
2018-05-30 08:30:00 +00:00
logging
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
media
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
modules
Unwrap TL0 pic index to avoid having to work with a wrapped number.
2018-07-06 13:28:38 +00:00
ortc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
p2p
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
pc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
resources
AGC2 RNN VAD: Polishing.
2018-05-15 16:41:02 +00:00
rtc_base
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
rtc_tools
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
sdk
Update capturer/test code to use the new interface VideoCapturer.
2018-07-06 10:42:11 +00:00
stats
Adding "is_standardized" flag to RTCStatsMember.
2018-06-28 00:43:46 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Removing /wd4334 from system_wrappers.
2018-07-05 09:18:34 +00:00
test
Limit input size to iSAC and iLBC decoder fuzzers
2018-07-05 14:30:12 +00:00
tools_webrtc
roll_deps: Accept any prefix (like 'git_revision:'), not only 'version:' for CIPD
2018-07-03 09:41:53 +00:00
video
Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader.
2018-07-05 14:29:07 +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
Roll back checking in the third_party directory
2018-06-27 13:04:08 +00:00
.gn
Set gtest_enable_absl_printers to true.
2018-06-29 09:36:17 +00:00
.vpython
Add vpython dependencies needed to run presubmit tests on LUCI
2018-05-18 08:10:25 +00:00
AUTHORS
Generalize SimulcastEncoderAdapter, use for H264 & VP8.
2018-06-21 15:57:43 +00:00
BUILD.gn
A new PeerConnection level perf test.
2018-06-27 23:19:05 +00:00
CODE_OF_CONDUCT.md
Add code of conduct to WebRTC repo
2017-05-16 12:09:13 +00:00
codereview.settings
Make Gerrit the default for WebRTC changes
2017-09-29 01:38:07 +00:00
common_types.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
DEPS
Roll chromium_revision b2695df06e..f0909821f4 (572823:572927)
2018-07-06 10:16:21 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
…
LICENSE_THIRD_PARTY
Remove third party dependecies that are not more in the source code
2018-06-21 11:33:41 +00:00
native-api.md
Remove legacy VoiceEngine.
2018-01-12 11:31:52 +00:00
OWNERS
Add mbonadei@ to build configs OWNERS.
2018-06-20 12:39:11 +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
Roll chromium_revision 95336cb92b..191d55580e (557816:557824)
2018-05-11 11:17:05 +00:00
PRESUBMIT.py
Adding CheckNoWarningSuppressionFlagsAreAdded.
2018-07-04 09:01:32 +00:00
pylintrc
Removing invalid-name from disabled pylint checks.
2017-10-11 08:06:49 +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
Add style guidance about forward declarations.
2018-03-28 20:58:27 +00:00
typedefs.h
Remove typedefs.h from webrtc/ root (part 1)
2018-05-23 12:07:10 +00:00
WATCHLISTS
Adding mbonadei@ to build_files WATCHLIST.
2018-06-20 12:38:06 +00:00
webrtc.gni
Don't call deprecated FFmpeg API.
2018-06-26 13:57:35 +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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