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Johannes Kron 133bf2bd28 Reland "Distinguish between send and receive codecs"
This reverts commit e57b266a20.

Reason for revert: Fixed negotiation of send-only clients.

Original change's description:
> Revert "Distinguish between send and receive codecs"
>
> This reverts commit c0f25cf762.
>
> Reason for revert: breaks negotiation with send-only clients
>
> (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
>
> Original change's description:
> > Distinguish between send and receive codecs
> >
> > Even though send and receive codecs may be the same, they might have
> > different support in HW. Distinguish between send and receive codecs
> > to be able to keep track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30284}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
> Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30292}

TBR=steveanton@webrtc.org,kron@webrtc.org


Bug: chromium:1029737
Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30348}
2020-01-22 13:55:41 +00:00
api Reland "Distinguish between send and receive codecs" 2020-01-22 13:55:41 +00:00
audio Reformat GN files. 2020-01-21 12:13:11 +00:00
build_overrides Remove crbug.com/904400 workaround. 2019-03-15 18:36:23 +00:00
call [Overuse] Setting the target bitrate through the interface. 2020-01-22 13:38:38 +00:00
common_audio Reformat GN files. 2020-01-21 12:13:11 +00:00
common_video Reformat GN files. 2020-01-21 12:13:11 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
docs Update Linux documentation links 2020-01-10 07:51:46 +00:00
examples Reformat GN files. 2020-01-21 12:13:11 +00:00
logging Reformat GN files. 2020-01-21 12:13:11 +00:00
media Reland "Distinguish between send and receive codecs" 2020-01-22 13:55:41 +00:00
modules Add absolute capture time to video sender path. 2020-01-22 13:09:28 +00:00
p2p Reformat GN files. 2020-01-21 12:13:11 +00:00
pc Reland "Distinguish between send and receive codecs" 2020-01-22 13:55:41 +00:00
resources Reformat GN files. 2020-01-21 12:13:11 +00:00
rtc_base Moves ownership of time controller into NetworkEmulationManager. 2020-01-22 11:12:27 +00:00
rtc_tools Reformat GN files. 2020-01-21 12:13:11 +00:00
sdk Reformat GN files. 2020-01-21 12:13:11 +00:00
stats Reformat GN files. 2020-01-21 12:13:11 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Reformat GN files. 2020-01-21 12:13:11 +00:00
test Propagate multicodec support to other places of PC level framework 2020-01-22 13:34:18 +00:00
tools_webrtc Reformat GN files. 2020-01-21 12:13:11 +00:00
video [Overuse] Setting the target bitrate through the interface. 2020-01-22 13:38:38 +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 new format cleanup. 2019-07-11 16:18:51 +00:00
.gitignore Add .clangd to .gitignore 2019-10-28 12:27:50 +00:00
.gn Switch to compiling WebRTC -std=c++14 by default 2019-09-09 19:24:16 +00:00
.vpython Add source-side perf upload script for WebRTC. 2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md Fix typo in abseil-in-webrtc.md. 2019-12-18 14:27:34 +00:00
AUTHORS Update Android camera switch API to allow specifying a name 2020-01-09 16:04:09 +00:00
BUILD.gn Fix video_replay to build and actually work 2020-01-22 13:16:28 +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 3f2a66dfa6..c565cfe6eb (733758:733868) 2020-01-22 04:52:34 +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 #COMPONENT to WebRTC. 2019-10-08 12:20:39 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
PRESUBMIT.py Extract an interface from the perf results logger. 2020-01-14 06:05:02 +00:00
presubmit_test.py Use source_sets in component builds and static_library in release builds. 2019-10-17 21:17:18 +00:00
presubmit_test_mocks.py Reland: Add presubmit check for changes in 3pp 2018-05-22 13:11:18 +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 guidance to style guide how to reference a bug in a TODO 2019-12-11 11:55:52 +00:00
WATCHLISTS Add saza to audio watchlists 2019-09-03 14:55:43 +00:00
webrtc.gni Reformat GN files. 2020-01-21 12:13:11 +00:00
webrtc_lib_link_test.cc Rewrite the lib link test to just be a binary. 2019-10-18 07:42:20 +00:00
whitespace.txt Revert "Whitespace change" 2019-11-11 14:58:20 +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