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Erik Språng 7d687b13ed SimulcastEncoderAdapter, don't start streams without enough bitrate 
Currently a bug in InitEncode() sets all stream initially to active.
This CL actually bases the active-flag on available start bitrate.

Bug: webrtc:9747
Change-Id: If197b0c69376d96c717f2a391fba8108895018f3
Reviewed-on: https://webrtc-review.googlesource.com/99960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24711}
2018-09-12 16:34:45 +00:00
api
AEC3: Reduce filter divergence during low-echo double-talk
2018-09-12 14:09:00 +00:00
audio
Delete always true member voe::Channel::pacing_enabled_
2018-09-11 11:40:11 +00:00
build_overrides
Add phoglund@ to various OWNERS and remove kjellander@
2017-10-19 09:21:12 +00:00
call
Use SdpVideoFormat in VideoReceiveStream::Decoder
2018-09-11 15:03:04 +00:00
common_audio
Remove simple stringstream usages.
2018-09-06 12:53:19 +00:00
common_video
Make the rtp timestamp member of EncodedImage private
2018-09-12 13:44:36 +00:00
data
…
examples
Remove redundant initializers from WebRTC Java code.
2018-09-11 09:58:10 +00:00
infra
Remove MSVC debug bots from CQ.
2018-09-01 09:20:03 +00:00
logging
Avoid wrong parsing of padding length and its use in NetEq simulation.
2018-09-12 11:23:03 +00:00
media
SimulcastEncoderAdapter, don't start streams without enough bitrate
2018-09-12 16:34:45 +00:00
modules
Make the rtp timestamp member of EncodedImage private
2018-09-12 13:44:36 +00:00
ortc
Fix no_exit_time_destructors in ortc.
2018-09-07 20:47:15 +00:00
p2p
Revert "Add SSLConfig object to IceServer."
2018-09-12 10:46:04 +00:00
pc
Revert "Add SSLConfig object to IceServer."
2018-09-12 10:46:04 +00:00
resources
Reland "Add Y4mFileReader"
2018-09-05 09:30:08 +00:00
rtc_base
Fix typo in bitbuffer.h
2018-09-12 15:23:44 +00:00
rtc_tools
Allow printing graphs as protobuf in event_log_visualizer.
2018-09-11 09:53:12 +00:00
sdk
Place static objects into a container that gets leaked.
2018-09-12 13:36:45 +00:00
stats
getStats: add kind alias for mediaType
2018-09-03 11:49:30 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Refactor initialization of GetRealTimeClock singleton.
2018-09-10 18:00:44 +00:00
test
in test::FrameGeneratorCapturer try to keep up with fps when overloaded
2018-09-12 14:55:09 +00:00
tools_webrtc
Excluded tests and test utils from code coverage by generate_coverage_command.py
2018-09-11 15:06:50 +00:00
video
Make the rtp timestamp member of EncodedImage private
2018-09-12 13:44:36 +00:00
.clang-format
Tune ObjC clang-format configuration
2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore format commit.
2018-06-20 09:26:44 +00:00
.gitignore
Roll back checking in the third_party directory
2018-06-27 13:04:08 +00:00
.gn
Re-enable gtest absl pretty printers.
2018-08-13 13:54:05 +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
Add a list of allowed and disallowed Abseil things
2018-08-27 08:37:20 +00:00
AUTHORS
Remove unused var in device info bits from video capture module for Linux
2018-09-12 10:36:33 +00:00
BUILD.gn
Enable -Wexit-time-destructors and -Wglobal-constructors.
2018-09-06 12:43:20 +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.h
Remove double declaration of 2 conversion functions.
2018-09-11 06:00:05 +00:00
DEPS
Roll chromium_revision bd99b7d4a8..c07e991426 (590554:590658)
2018-09-12 14:25:36 +00:00
LICENSE
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
…
native-api.md
Add documentation for WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS.
2018-08-21 13:29:36 +00:00
OWNERS
Add a list of allowed and disallowed Abseil things
2018-08-27 08:37:20 +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
Upload WebRTC CLs from Chromium.
2018-09-04 11:14:07 +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
Add a list of allowed and disallowed Abseil things
2018-08-27 08:37:20 +00:00
WATCHLISTS
Reland "Remove steveanton from api/ watchlist"
2018-09-05 11:45:33 +00:00
webrtc.gni
Fix paths for macOS framework’s copy actions.
2018-09-06 14:13:11 +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
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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