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Åsa Persson 45b176fc22 Downgrade fps in same step as resolution in initial drop due to size. 
Bug: none
Change-Id: If0943ee291a029fa81035c72607873995ba8ab8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154742
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29342}
2019-09-30 12:28:26 +00:00
api
Refactor legacy FrameBuffer to use EncodedImageBuffer::Realloc
2019-09-30 09:48:26 +00:00
audio
Propagate task queue to create test::DirectTransport by TaskQueueBase interface
2019-09-30 03:23:07 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Propagate task queue to create test::DirectTransport by TaskQueueBase interface
2019-09-30 03:23:07 +00:00
common_audio
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
common_video
Always pass arguments to INSTANTIATE_TEST_SUITE_P.
2019-09-24 08:56:24 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Fixing some typos.
2019-09-10 10:03:50 +00:00
examples
New build target api:media_interface
2019-09-19 09:32:27 +00:00
logging
Adds abs-send-time and size field outputs to event log parser.
2019-09-24 16:31:22 +00:00
media
Remove obsolete todo comment in simulcast.h
2019-09-20 14:17:20 +00:00
modules
Revert "Prepares RtpSenderVideo for batch forwarding of generated packets"
2019-09-30 11:20:04 +00:00
p2p
Delete the BasicPortAllocator constructor that enables gturn
2019-09-24 07:46:28 +00:00
pc
Allow receive-only use of datagram transport for data channels.
2019-09-26 20:01:06 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
Add Slice method to CopyOnWriteBuffer and use it in FEC code.
2019-09-26 09:48:07 +00:00
rtc_tools
Use std::make_unique instead of absl::make_unique.
2019-09-17 15:47:29 +00:00
sdk
Avoids update of WebRTC.Audio.SourceMatchesRecordingSession for Android < N
2019-09-26 14:59:12 +00:00
stats
Add qualityLimitationResolutionChanges stat
2019-09-09 15:22:57 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Move code related to VideoCodingModule to its own build target
2019-09-10 12:34:38 +00:00
test
Propagate task queue to create test::DirectTransport by TaskQueueBase interface
2019-09-30 03:23:07 +00:00
tools_webrtc
Stop explicitly setting use_prebuilt_instrumented_libraries on msan bots.
2019-09-12 18:21:38 +00:00
video
Downgrade fps in same step as resolution in initial drop due to size.
2019-09-30 12:28:26 +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
Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH.
2019-09-04 18:49:28 +00:00
.gn
Switch to compiling WebRTC -std=c++14 by default
2019-09-09 19:24:16 +00:00
.vpython
Add vpython dependencies needed to run presubmit tests on LUCI
2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md
Update style guide for absl::make_unique.
2019-09-18 06:10:58 +00:00
AUTHORS
pipewire: handle deleting the capturer while a D-Bus call is in progress
2019-09-26 18:58:56 +00:00
BUILD.gn
Introduce api/crypto/BUILD.gn.
2019-09-13 17:21:47 +00:00
CODE_OF_CONDUCT.md
Add code of conduct to WebRTC repo
2017-05-16 12:09:13 +00:00
codereview.settings
Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit
2018-10-25 08:19:53 +00:00
common_types.h
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision 929ce7dfed..bd70e4cf18 (700900:701000)
2019-09-29 16:29:56 +00:00
ENG_REVIEW_OWNERS
Enforce LGTM from owners of depends-on paths in DEPS via presubmit.
2018-09-28 12:49:54 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
…
native-api.md
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Add juberti@ to webrtc root owners
2019-05-17 18:11:58 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +00:00
presubmit_test.py
Fixing py lint errors
2018-07-23 15:28:48 +00:00
PRESUBMIT.py
absl::make_unique presubmit check.
2019-09-17 17:47:31 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +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
Update WebRTC's C++ style guide to reflect the switch to C++14.
2019-09-16 11:45:35 +00:00
WATCHLISTS
Add saza to audio watchlists
2019-09-03 14:55:43 +00:00
webrtc.gni
Remove rtc_use_lto GN arg.
2019-08-20 14:00:49 +00:00
whitespace.txt
Whitespace change
2018-02-23 10:34:16 +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

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%