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Anton Sukhanov 292ce4ef25 Move datagram transport to JsepTransport
- This makes it consistent with ICE and MediaTransport ownership.
- Removes unnecessary datagram_transport() getter in DtlsTransportInternal

As a side effect this fixes bug in JsepTransportController, which moved datagram_transport to Dtls after creating it, then checked if (datagram_transport) to decide which RTP transport to create. As a result of this bug we were creating Sded instead of Unencrypted RTP with datagram transport.

Bug: webrtc:9719
Change-Id: Ic5b13a450ce6ac5b2a20d388657e3949aabef079
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139620
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28146}
2019-06-03 22:24:12 +00:00
api Change the gating of surfacing candidates on ICE transport type change 2019-06-03 18:41:13 +00:00
audio Reland "Avoid encrypting empty audio packet." 2019-05-28 12:30:07 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call Add support for early loss detection using transport feedback. 2019-05-29 13:21:10 +00:00
common_audio Add ability to play audio in circle for TestAudioDevice wav file capturer 2019-04-16 15:33:03 +00:00
common_video Move H.264 SPS VUI rewriting to FrameEncodeMetadataWriter. 2019-05-29 10:37:22 +00:00
crypto Adding new top-level directory crypto/ 2019-03-08 00:35:05 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Target SDK level 29 in AppRTCMobile. 2019-06-03 12:40:42 +00:00
logging Fix for crash in event log when using scenario tests. 2019-05-22 15:22:49 +00:00
media Check H264 packetization mode when using IsSameCodec 2019-05-31 21:18:21 +00:00
modules Allowing buffering a LNTF (loss notification) feedback message in RTCPSender 2019-06-03 16:28:34 +00:00
p2p Move datagram transport to JsepTransport 2019-06-03 22:24:12 +00:00
pc Move datagram transport to JsepTransport 2019-06-03 22:24:12 +00:00
resources RNN VAD: clean-up unit tests 2019-04-29 12:55:02 +00:00
rtc_base Lockless SwapQueue 2019-06-03 11:49:49 +00:00
rtc_tools Fix BWE simulation graph in event log visualization 2019-05-28 10:40:29 +00:00
sdk Change the gating of surfacing candidates on ICE transport type change 2019-06-03 18:41:13 +00:00
stats Implement QualityLimitationReasonTracker and expose "reason". 2019-05-28 16:23:55 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Ensure CpuInfo::DetectNumberOfCores is > 0 and thread safe. 2019-05-24 12:59:14 +00:00
test Allowing buffering a LNTF (loss notification) feedback message in RTCPSender 2019-06-03 16:28:34 +00:00
tools_webrtc mb: Implement 'quiet' flag in mb lookup 2019-05-31 11:51:15 +00:00
video Allowing buffering a LNTF (loss notification) feedback message in RTCPSender 2019-06-03 16:28:34 +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 Add Visual Studio Code project folder to gitignore file. 2019-01-21 18:42:33 +00:00
.gn Remove last mention of ortc from the codebase. 2019-05-25 07:28:05 +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 Allowing buffering a LNTF (loss notification) feedback message in RTCPSender 2019-06-03 16:28:34 +00:00
AUTHORS Import proto_library.gni when rtc_enable_protobuf is true 2019-02-27 09:56:42 +00:00
BUILD.gn Delete rtc_base/unittest_main.cc 2019-05-21 14:44:11 +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 Move kRtpCsrcSize from common_types.h to rtp_headers.h 2019-05-10 09:43:39 +00:00
DEPS Roll chromium_revision a3e71ebfa3..584b49b1a7 (665525:665633) 2019-06-03 20:56:26 +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 Update template to follow chromium copyright style 2013-04-24 01:01:28 +00:00
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 Do not run CheckNoStreamUsageIsAdded on tests. 2019-05-07 15:52:35 +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 Remove rule that discourages passing optional by const reference 2019-02-05 11:58:05 +00:00
WATCHLISTS Add fhernqvist to watchlist. 2019-06-03 10:13:40 +00:00
webrtc.gni Testing no /DUNICODE assumptions with Win more configs bots. 2019-03-28 08:46:37 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +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 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