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Jonas Oreland 6545516a14 RtpSenderInterface::SetEncoderSelector
This cl/ adds a way of setting an EncoderSelector on a specific
RtpSenderInterface. This makes it possible to easily use different
EncoderSelector on different streams within the same or different PeerConnections.

The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor.

Iff a EncoderSelector is set on the RtpSender, it will take precedence
over the VideoEncoderFactory::GetEncoderSelector.

Bug: webrtc:14122
Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37150}
2022-06-08 11:06:52 +00:00
api RtpSenderInterface::SetEncoderSelector 2022-06-08 11:06:52 +00:00
audio Migrate some scripts to python3 2022-06-01 10:09:36 +00:00
build_overrides Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions 2021-05-27 16:16:01 +00:00
call RtpSenderInterface::SetEncoderSelector 2022-06-08 11:06:52 +00:00
common_audio Adopt absl::string_view in common_audio/ 2022-05-13 15:00:14 +00:00
common_video Remove dependency on rtc_base_approved from most targets 2022-04-25 12:15:30 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Add contributing.md 2021-11-03 14:59:46 +00:00
examples Update android example gradle and target sdk version. 2022-05-31 07:14:24 +00:00
g3doc Remove mention of MSVC support from docs 2022-05-02 09:26:32 +00:00
infra Rename 'iOS64 Sim Debug (iOS 14.0)' bot. 2022-06-03 07:17:27 +00:00
logging Improve RTC event log parser. 2022-05-18 09:14:40 +00:00
media RtpSenderInterface::SetEncoderSelector 2022-06-08 11:06:52 +00:00
modules Refactor rtcp::TransportFeedback to use Timestamp and TimeDelta internally 2022-06-08 09:33:22 +00:00
net/dcsctp dcsctp: Refactor Send Queue 2022-06-07 15:01:32 +00:00
p2p Reland "Don't create PacketSocketFactory inside BasicPortAllocatorSession" 2022-06-07 11:24:16 +00:00
pc RtpSenderInterface::SetEncoderSelector 2022-06-08 11:06:52 +00:00
resources AEC3: Changing the default for the use_conservative_tail_frequency_response flag. 2021-12-21 17:35:26 +00:00
rtc_base Reland "Don't create PacketSocketFactory inside BasicPortAllocatorSession" 2022-06-07 11:24:16 +00:00
rtc_tools Disable tests depending on libyuv upscale filtering 2022-06-02 13:24:46 +00:00
sdk Make the libaom AV1 encoder optional in RTCDefaultVideoEncoderFactory. 2022-06-01 10:54:55 +00:00
stats stats: expose minPlayoutDelay as nonstandard stat 2022-05-31 10:05:35 +00:00
system_wrappers Adopt absl::string_view in system_wrappers/ 2022-05-17 08:23:24 +00:00
test Reland "Don't create PacketSocketFactory inside BasicPortAllocatorSession" 2022-06-07 11:24:16 +00:00
tools_webrtc Rename 'iOS64 Sim Debug (iOS 14.0)' bot. 2022-06-03 07:17:27 +00:00
video RtpSenderInterface::SetEncoderSelector 2022-06-08 11:06:52 +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 Prevent jsoncpp from hiding deprecated declarations in WebRTC 2022-04-11 12:33:47 +00:00
.mailmap Add .mailmap for git. 2022-02-20 14:22:13 +00:00
.style.yapf Fix mb.py presubmit issues. 2021-12-08 08:53:00 +00:00
.vpython Remove unused script webrtc_dashboard_upload.py 2022-03-21 12:54:42 +00:00
.vpython3 Remove unused script webrtc_dashboard_upload.py 2022-03-21 12:54:42 +00:00
AUTHORS GCC: Avoid symbol clash in RenderBuffer 2022-05-24 10:47:56 +00:00
BUILD.gn Add support for stand-alone Fuchsia build. 2022-05-30 09:24:43 +00:00
CODE_OF_CONDUCT.md Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +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
DEPS Roll chromium_revision 42fc0665a4..3bf62ec766 (1011812:1011829) 2022-06-08 08:46:52 +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 phoglund from ENG_REVIEW_OWNERS 2021-10-08 08:29:42 +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 Reference "main" branches instead of "master" branches. 2021-07-15 11:07:44 +00:00
OWNERS Fix add some eng prod owners to PRESUBMIT.py. 2022-03-18 13:19:07 +00:00
PATENTS
PRESUBMIT.py Update portaudio to the latest 2022-05-13 09:01:34 +00:00
presubmit_test.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
presubmit_test_mocks.py tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
pylintrc tools_webrtc dir converted to py3 + top level PRESUBMIT script 2022-02-08 14:42:26 +00:00
README.chromium Add CPEPrefix. 2020-07-13 11:42:07 +00:00
README.md doc: add g3doc sitemap to toplevel readme 2021-07-23 07:55:17 +00:00
WATCHLISTS Update WATCHLISTS 2021-08-23 13:37:55 +00:00
webrtc.gni Fork test/ios/google_test_runner.mm from Chromium's //base 2022-05-04 11:26:07 +00:00
webrtc_lib_link_test.cc Deprecate PeerConnectionFactory::CreatePeerConnection 2021-05-10 08:47:48 +00:00
whitespace.txt Trigger CI bots 2021-12-16 17:45:31 +00:00

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