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Åsa Persson f5f7e8e806 Ensure that fps adaptation count can go back to zero when framerate is unrestricted. 
Bug: webrtc:12867
Change-Id: I1c11d1a1154ea3d802cdc01e260f72a7e9d17e99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221373
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#34265}
2021-06-10 16:00:39 +00:00
api
Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz.
2021-06-09 18:41:47 +00:00
audio
Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz.
2021-06-09 18:41:47 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
Ensure that fps adaptation count can go back to zero when framerate is unrestricted.
2021-06-10 16:00:39 +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
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Create a VideoFrameTrackingId RTP header extension.
2021-03-25 17:25:18 +00:00
examples
Added PeerConnectionObserverJni::OnRemoveTrack()
2021-06-03 19:24:55 +00:00
g3doc
doc: document rtp payload type mapping behaviour
2021-06-04 06:23:32 +00:00
logging
Add documentation for RTC event log
2021-06-03 09:03:18 +00:00
media
Add small cooldown to unsignalled ssrc stream creation.
2021-06-10 13:13:21 +00:00
modules
Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz.
2021-06-09 18:41:47 +00:00
net/dcsctp
dcsctp: Prevent overflow of missing parameters
2021-06-09 14:12:53 +00:00
p2p
Avoid generating a random id for candidate stats.
2021-06-08 08:00:01 +00:00
pc
Prepare for breakout of session_description.{h,cc}
2021-06-10 13:20:32 +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
Update the only 3 remaining kFilterBilinear to kFilterBox.
2021-06-08 13:19:23 +00:00
sdk
Move AV1X-AV1 mapping to VideoCodecTypeMime
2021-06-04 15:38:48 +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
Break out pc/session_description in its own build target (part 1)
2021-06-09 11:39:06 +00:00
tools_webrtc
Add MB configs for M1 bots
2021-06-09 19:02:28 +00:00
video
Ensure that fps adaptation count can go back to zero when framerate is unrestricted.
2021-06-10 16:00:39 +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
Added PeerConnectionObserverJni::OnRemoveTrack()
2021-06-03 19:24:55 +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 415567486a..d2f297f391 (890510:890623)
2021-06-09 06:36:14 +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
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Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%