signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Emircan Uysaler bbdabe50db Rename stereo video codec to multiplex 
This CL only does the rename from"stereo" to multiplex". With this we have a
better name that doesn't clash with audio's usage of stereo.

Bug: webrtc:7671
Change-Id: Iebc3fc20839025f1bc8bcf0e16141bf9744ef652
Reviewed-on: https://webrtc-review.googlesource.com/43242
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21769}
2018-01-25 23:16:04 +00:00
api Change return type of RtpSenderInterface::SetParameters from bool to RTCError 2018-01-25 19:06:04 +00:00
audio Remove logging in audio/* from release builds. 2018-01-25 13:46:54 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Make fec controller plug-able. 2018-01-22 11:48:16 +00:00
common_audio Replace left shift with equivalent multiplication. 2018-01-12 15:29:59 +00:00
common_video Rename stereo video codec to multiplex 2018-01-25 23:16:04 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Reland "Make it possible to run video_quality_loopback_test in swarming." 2018-01-23 13:03:17 +00:00
infra Make new Chromium compile trybots mandatory. 2018-01-12 09:41:51 +00:00
logging Remove dependency on system_wrappers from rtc_event_log. 2018-01-19 18:04:03 +00:00
media Rename stereo video codec to multiplex 2018-01-25 23:16:04 +00:00
modules Rename stereo video codec to multiplex 2018-01-25 23:16:04 +00:00
ortc Configure media flow correctly with Unified Plan 2018-01-18 19:01:38 +00:00
p2p Don't use link-local networks to determine the lowest cost of networks 2018-01-24 00:38:01 +00:00
pc Change return type of RtpSenderInterface::SetParameters from bool to RTCError 2018-01-25 19:06:04 +00:00
resources Add a new NetEq decoding unit test for Opus with DTX 2017-11-28 10:45:38 +00:00
rtc_base Stop undefining EACCES. 2018-01-25 19:12:14 +00:00
rtc_tools Making rtc_tools:unpack_aecdump publicly visible. 2018-01-11 09:04:30 +00:00
sdk Change return type of RtpSenderInterface::SetParameters from bool to RTCError 2018-01-25 19:06:04 +00:00
stats Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" 2018-01-10 15:55:04 +00:00
system_wrappers Reland Use runtime enabled features API to enable dual stream mode 2018-01-18 12:22:49 +00:00
test Rename stereo video codec to multiplex 2018-01-25 23:16:04 +00:00
tools_webrtc Parameterize PeerConnection end to end tests for Unified Plan 2018-01-25 04:03:18 +00:00
video Rename stereo video codec to multiplex 2018-01-25 23:16:04 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Create .git-blame-ignore-revs and add Java format CL to it. 2016-10-20 09:20:39 +00:00
.gitignore Reland "Make it possible to run video_quality_loopback_test in swarming." 2018-01-23 13:03:17 +00:00
.gn Pinning NDK api levels. 2018-01-18 16:55:58 +00:00
.vpython Add psutil to vpython dependencies (used on builder bots) 2017-09-04 08:04:16 +00:00
AUTHORS RemoteBitrateEstimatorAbsSendTime: check clock is a valid ref 2018-01-16 01:27:11 +00:00
BUILD.gn Reland Use runtime enabled features API to enable dual stream mode 2018-01-18 12:22:49 +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 Rename stereo video codec to multiplex 2018-01-25 23:16:04 +00:00
common_types.h Rename stereo video codec to multiplex 2018-01-25 23:16:04 +00:00
DEPS Roll chromium_revision e173942fcd..5d01e2667f (531623:531725) 2018-01-24 23:27:08 +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
LICENSE_THIRD_PARTY Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
native-api.md Remove legacy VoiceEngine. 2018-01-12 11:31:52 +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 Remove legacy VoiceEngine. 2018-01-12 11:31:52 +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 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" 2018-01-10 15:55:04 +00:00
typedefs.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
WATCHLISTS Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
webrtc.gni Update gn files to support Mozilla build 2018-01-16 07:51:23 +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