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Åsa Persson 1d2b22e193 Use pixels from single active stream if set for balanced degradation settings. 
Bug: none
Change-Id: Id9d8dd5a447ea99f080fc597b28afd9fbbe90db2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220922
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34213}
2021-06-03 10:20:56 +00:00
api
Add ability to adjust the suppressor smoothing in AEC3
2021-06-02 15:21:35 +00:00
audio
AudioSendStream: s/worker_queue_/rtp_transport_queue_/g
2021-06-01 08:00:22 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
Use pixels from single active stream if set for balanced degradation settings.
2021-06-03 10:20:56 +00:00
common_audio
Avoid undefined behavior in a division operation.
2021-04-23 07:49:24 +00:00
common_video
Update BitBuffer methods to style guide
2021-05-18 11:10:27 +00:00
data
…
docs
Create a VideoFrameTrackingId RTP header extension.
2021-03-25 17:25:18 +00:00
examples
Introduce CreateDataChannelOrError
2021-05-26 09:43:29 +00:00
g3doc
Add documentation for RTC event log
2021-06-03 09:03:18 +00:00
logging
Add documentation for RTC event log
2021-06-03 09:03:18 +00:00
media
fall back to payload types from lower range after exhausting [96,127]
2021-06-01 07:50:43 +00:00
modules
Reinstate killswitch for WebRTC-Bwe-ReceiverLimitCapsOnly.
2021-06-03 09:15:58 +00:00
net/dcsctp
dcsctp: Stay in stream if not producing fragment
2021-06-02 09:45:59 +00:00
p2p
Remove cricket::DtlsTransportState.
2021-05-21 21:45:29 +00:00
pc
SdpOfferAnswerHandler: Significantly reduce audio impairment.
2021-06-02 15:15:04 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Proxy: solve event tracing with compile time strings.
2021-05-28 06:53:29 +00:00
rtc_tools
Convert to NTP time using the real clock.
2021-05-21 19:29:32 +00:00
sdk
Add resolution alignment properties to RTCVideoEncoder protocol.
2021-06-02 09:30:06 +00:00
stats
Populate qualityLimitationDurations stats for outbound RTP streams
2021-05-31 21:39:37 +00:00
system_wrappers
Make Clock::ConvertTimestampToNtpTime pure virtual
2021-05-21 09:55:14 +00:00
test
RtpFrameReferenceFinder return frames directly instead of via callback.
2021-05-26 15:47:03 +00:00
tools_webrtc
Make generate_license.py compatible with Python 3.
2021-05-31 07:05:46 +00:00
video
Use pixels from single active stream if set for balanced degradation settings.
2021-06-03 10:20:56 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Switch from check_targets to no_check_targets in .gn
2021-05-20 10:42:21 +00:00
.vpython
Update six library version
2021-04-26 16:39:07 +00:00
AUTHORS
Fix incorrect SSRC in RtpPacketSendInfo for RTX packets.
2021-05-31 20:51:07 +00:00
BUILD.gn
Switch from check_targets to no_check_targets in .gn
2021-05-20 10:42:21 +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 c165693ba5..fee5f397ef (888523:888712)
2021-06-03 02:54:05 +00:00
DIR_METADATA
Move metadata in OWNERS files to DIR_METADATA files.
2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
g3doc.lua
Improve webrtc documentation infra. Preview at:
2021-03-30 10:29:30 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Make the remote_bitrate_estimator build target private
2020-11-26 12:21:22 +00:00
OWNERS
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
PATENTS
…
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
sctp: Rename SctpTransport to UsrSctpTransport
2021-04-12 10:40:34 +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
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
WATCHLISTS
Add hta@ to rtc_base/ and api/ WATCHLISTS.
2021-01-06 09:43:34 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Turn on the RTC_ENABLE_WIN_WGC build flag.
2021-05-10 20:16:52 +00:00
whitespace.txt
Reland "Triggering CI."
2021-03-22 11:57:23 +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
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%