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Harald Alvestrand fdbaeda003 Improve structuring of test for audio glitches. 
Bug: webrtc:12361
Change-Id: Ieddc3dafbb638b3bd73dd79bcafa499290fa4340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201723
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32973}
2021-01-14 10:29:58 +00:00
api
Reland "ChannelStatistics used for RTP stats in VoipStatistics."
2021-01-13 16:57:22 +00:00
audio
Reland "ChannelStatistics used for RTP stats in VoipStatistics."
2021-01-13 16:57:22 +00:00
build_overrides
[build] Remove obsolete gn flag
2021-01-11 17:57:44 +00:00
call
Update WebRTC code version (2021-01-13T04:03:21).
2021-01-13 05:43:55 +00:00
common_audio
Organize iLBC headers as per style guide
2020-12-31 20:57:18 +00:00
common_video
Delete KeepBufferRefs helpers, and use of rtc::Bind.
2021-01-12 12:35:38 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
doc: update working-with-native-branches information
2020-12-16 09:02:46 +00:00
examples
Refactor rtc_base build targets.
2021-01-11 18:32:30 +00:00
logging
Move logged types for RTC event log into event headers.
2021-01-12 16:45:40 +00:00
media
Add av1 to lower range IDs.
2021-01-13 20:48:57 +00:00
modules
Fix AudioProcessing::Config::ToString() implementation.
2021-01-14 10:27:18 +00:00
p2p
Refactor rtc_base build targets.
2021-01-11 18:32:30 +00:00
pc
Improve structuring of test for audio glitches.
2021-01-14 10:29:58 +00:00
resources
Erle metric: avoid the decreasing of the metric when there is no render activity.
2020-12-09 11:48:11 +00:00
rtc_base
Delete KeepBufferRefs helpers, and use of rtc::Bind.
2021-01-12 12:35:38 +00:00
rtc_tools
Refactor rtc_base build targets.
2021-01-11 18:32:30 +00:00
sdk
Handle case createShader throws an exception.
2021-01-13 15:50:12 +00:00
stats
stats: add transportId to codec stats
2020-11-17 12:34:39 +00:00
style-guide
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
system_wrappers
Delete use of RWLockWrapper from SimulatedClock
2020-11-04 08:01:08 +00:00
test
Fix inconsistencies in network BUILD.gn file
2021-01-14 09:43:26 +00:00
tools_webrtc
Replace all uses of the word 'master' with 'builder_group' in //tools/mb
2021-01-13 16:16:02 +00:00
video
Add new empty build targets rtp_rtcp_legacy and video_legacy.
2021-01-14 09:45:26 +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
Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
2020-09-07 08:37:14 +00:00
.vpython
Reland "Add protobuf-py2_py3 3.13.0 to .vpython."
2020-11-20 07:52:26 +00:00
abseil-in-webrtc.md
Polish the "Using Abseil in WebRTC" docs
2020-10-16 13:42:00 +00:00
AUTHORS
Fix "control reaches end of non-void function" warnings
2020-10-27 10:22:23 +00:00
BUILD.gn
Add WebRTC code freshness version string.
2020-12-14 16:22:35 +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
DEPS
Roll chromium_revision c0feffab8f..189823ba75 (842900:843026)
2021-01-13 16:53:42 +00:00
ENG_REVIEW_OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
…
native-api.md
Make the remote_bitrate_estimator build target private
2020-11-26 12:21:22 +00:00
OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
presubmit_test.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
PRESUBMIT.py
Fix Authors Check presubmit.
2020-12-15 11:59:03 +00:00
pylintrc
Undo enforcing of PEP-8 pylint changes for method and function names.
2020-11-10 18:26:25 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
doc: move bug reporting instructions to the repository
2020-10-21 14:47:49 +00:00
style-guide.md
C++ style: We don't allow designated initializers
2020-06-03 09:11:09 +00:00
WATCHLISTS
Add hta@ to rtc_base/ and api/ WATCHLISTS.
2021-01-06 09:43:34 +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
Get rid of recently added rtc_openssl_is_boringssl build argument.
2020-12-15 08:34:42 +00:00
whitespace.txt
Trigger bots for FYI waterfall.
2020-11-20 09:35:54 +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 here 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
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Readme 255 MiB
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