webrtc_m130

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Steve Anton 5634427c09 Remove unused properties from MediaContentDescription

Bug: webrtc:8620
Change-Id: I0597495a93aa7507177cf7bcbb8c6d35ee90ae0d
Reviewed-on: https://webrtc-review.googlesource.com/30500
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21150}

2017-12-07 23:03:19 +00:00

api

Fix clang style errors in api/jsep.h

2017-12-06 18:12:06 +00:00

audio

Do not add audio bitrate observer if TWCC sending is not supported

2017-12-07 20:46:38 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +00:00

call

Deliver packet to Call as rtc::CopyOnWriteBuffer

2017-12-07 17:09:07 +00:00

common_audio

Avoid integer-overflow in WebRtcSpl_FilterARFastQ12

2017-12-07 16:56:47 +00:00

common_video

Stop using public_deps to depend on libyuv.

2017-12-04 14:16:08 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

2016-11-22 18:43:05 +00:00

examples

Replace string type with SdpType enum

2017-12-07 19:08:47 +00:00

infra

Adding win_asan to CQ.

2017-11-22 09:12:18 +00:00

logging

Stop using public_deps in modules/rtp_rtcp.

2017-12-06 07:37:52 +00:00

media

Deliver packet to Call as rtc::CopyOnWriteBuffer

2017-12-07 17:09:07 +00:00

modules

Revert "Make RTCP cumulative_lost be a signed value"

2017-12-07 18:12:48 +00:00

ortc

Stop using public_deps in pc/.

2017-12-07 13:57:57 +00:00

p2p

Fix cpplint errors in p2p/

2017-12-06 21:33:36 +00:00

Remove unused properties from MediaContentDescription

2017-12-07 23:03:19 +00:00

resources

Add a new NetEq decoding unit test for Opus with DTX

2017-11-28 10:45:38 +00:00

rtc_base

Delete Filesystem::TempFilename.

2017-12-07 20:05:57 +00:00

rtc_tools

Fix awk gensub call in author_line_count.sh script.

2017-12-06 08:16:12 +00:00

sdk

Replace string type with SdpType enum

2017-12-07 19:08:47 +00:00

stats

Stop using public_deps in stats.

2017-12-04 15:42:18 +00:00

system_wrappers

Stop using public_deps in system_wrappers.

2017-12-06 08:56:52 +00:00

test

Deliver packet to Call as rtc::CopyOnWriteBuffer

2017-12-07 17:09:07 +00:00

tools_webrtc

WebRTC: Add iOS perf trybots (3).

2017-12-05 16:13:12 +00:00

video

Deliver packet to Call as rtc::CopyOnWriteBuffer

2017-12-07 17:09:07 +00:00

voice_engine

Stop using public_deps in system_wrappers.

2017-12-06 08:56:52 +00:00

.clang-format

Tune ObjC clang-format configuration

2017-05-11 09:14:18 +00:00

.git-blame-ignore-revs

…

.gitignore

Roll chromium_revision f93b8b19f2..adf969a7cb (513366:514871) and more.

2017-11-08 21:42:48 +00:00

.gn

Roll chromium_revision 5df4e1bfe7..5bd5874cbf (518692:519731) + iOS fix

2017-11-28 18:42:38 +00:00

.vpython

Add psutil to vpython dependencies (used on builder bots)

2017-09-04 08:04:16 +00:00

AUTHORS

Delete wrapper API ConvertToI420 for YUV conversion to I420

2017-11-15 11:10:20 +00:00

BUILD.gn

Remove webrtc_tests.

2017-12-04 10:15:18 +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.cc

Add a new function to BitrateAllocation: HasBitrate.

2017-11-23 15:00:08 +00:00

common_types.h

Delete MediaFile support for unused fileformats.

2017-12-05 11:37:23 +00:00

DEPS

Roll chromium_revision 5799f48f47..5d9089080e (522145:522183)

2017-12-06 21:28:46 +00:00

LICENSE

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

license_template.txt

…

LICENSE_THIRD_PARTY

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

native-api.md

Tell users where they can find the native API headers

2017-11-14 10:36:46 +00:00

OWNERS

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +00:00

PATENTS

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

presubmit_test_mocks.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

presubmit_test.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

PRESUBMIT.py

Fix cpplint errors in p2p/

2017-12-06 21:33:36 +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

Style guide: Attempt to make the L2 and L3 headings more visually distinct

2017-09-09 03:52:23 +00:00

typedefs.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

WATCHLISTS

Add myself to the watchlist for webrtc/api/ and webrtc/base/

2017-05-04 13:22:46 +00:00

webrtc.gni

Removing assert in rtc_test template.

2017-11-22 08:26:37 +00:00

whitespace.txt

Moving src/webrtc into src/.

2017-09-15 04:25:06 +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

Official web site: http://www.webrtc.org
Master source code repo: https://webrtc.googlesource.com/src
Samples and reference apps: https://github.com/webrtc
Mailing list: http://groups.google.com/group/discuss-webrtc
Continuous build: http://build.chromium.org/p/client.webrtc
Coding style guide
Code of conduct

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C 2.2%

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Python 1.3%

Other 1%