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Alexander Cooper a7d10811cd Revert "pipewire capturer: Reduce the amount of copying"
This reverts commit 8856410b6d.

Reason for revert: chromium:1447540

Original change's description:
> pipewire capturer: Reduce the amount of copying
>
> Improves the capture latency by reducing the amount of
> copying needed from the frame. We keep track of the
> damaged region of previous frame and union it with
> the damaged region of this frame and only copy this
> union of the frame over. X11 capturer already has
> such synchronization in place.
>
> The change is beneficial especially when there are
> small changes on the screen (e.g. clock ticking).
> For a 4k screen with 128 cores, I observed the
> capture latencies drop from 5 - 8 ms to 0 ms when the
> system is left idle. This is in line with the X11
> capturer.
>
> Bug: chromium:1291247
> Change-Id: Iffb441f9e1902d2658031f5f35b5372ee8e94073
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299720
> Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> Commit-Queue: Salman Malik <salmanmalik@chromium.org>
> Cr-Commit-Position: refs/heads/main@{#39968}

Bug: chromium:1291247
Change-Id: Id1bfd3fc39fea2bb1f232cad5218f90e144920e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306263
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40123}
2023-05-23 17:58:08 +00:00
api RequestedResolution - Bug fix 2023-05-22 13:58:50 +00:00
audio Pass rtcp message to RtpTransportController through newer interface 2023-05-17 17:19:23 +00:00
build_overrides Define enable_safe_libcxx in build_overrides/build.gni. 2023-05-03 08:18:25 +00:00
call Update WebRTC code version (2023-05-23T04:03:48). 2023-05-23 05:36:31 +00:00
common_audio Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
common_video webrtc_libyuv: Add support for more video types for consistency 2023-04-24 19:06:25 +00:00
data
docs fix some more minor typos 2023-05-11 12:26:25 +00:00
examples Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
experiments Add tool for generating field trial registry header 2022-10-18 07:25:43 +00:00
g3doc Add docs about adding a new test binary. 2023-03-07 11:12:33 +00:00
infra Add video_codec_perf_tests to desktop and android perf test suites 2023-05-23 12:13:29 +00:00
logging Use DD encoder/decoder in RTC event log encoder/parser. 2023-04-24 10:35:22 +00:00
media Unit tests for MediaChannel creation API 2023-05-23 13:24:46 +00:00
modules Revert "pipewire capturer: Reduce the amount of copying" 2023-05-23 17:58:08 +00:00
net/dcsctp Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
p2p Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ 2023-05-03 11:09:26 +00:00
pc Add test for split-mode SSRC callback 2023-05-23 07:56:57 +00:00
resources Clarify and extend test support for certain sample rates in audio processing 2022-08-03 14:26:36 +00:00
rtc_base Adds WebRTC-DisableRtxRateLimiter for enable/disable RTX rate limiter. 2023-05-23 09:59:11 +00:00
rtc_tools Format the rest 2023-05-03 12:56:39 +00:00
sdk Add EglThread class wrapping EglConnection and handler. 2023-05-23 14:02:21 +00:00
stats stats: remove media_type which was an alias for kind 2023-05-09 11:46:52 +00:00
system_wrappers Format the rest 2023-05-03 12:56:39 +00:00
test Format the rest 2023-05-03 12:56:39 +00:00
tools_webrtc Rewrite 'generate_sslroots' w/o OpenSSL. 2023-05-10 12:57:37 +00:00
video OveruseFrameDetector: complete removal of mac rules kill switch. 2023-05-23 09:05:53 +00:00
.clang-format Add IncludeBlocks to clang-format. 2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs Add formatting CLs to .git-blame-ignore-revs 2023-05-07 09:27:47 +00:00
.gitignore Add .cache to .gitignore. 2021-01-20 15:01:07 +00:00
.gn Set use_cxx to true. 2023-05-17 06:30:04 +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 Rewrite 'generate_sslroots' w/o OpenSSL. 2023-05-10 12:57:37 +00:00
AUTHORS Expose setCodecPreferences/getCapabilities for android 2023-05-15 19:24:15 +00:00
BUILD.gn Add video_codec_perf_tests to desktop and android perf test suites 2023-05-23 12:13:29 +00:00
CODE_OF_CONDUCT.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
codereview.settings
DEPS Roll chromium_revision 692840e030..c025c4ac7b (1147747:1147854) 2023-05-23 14:58:53 +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
LICENSE
license_template.txt
native-api.md Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
OWNERS Add infra owners file 2022-12-02 09:21:47 +00:00
OWNERS_INFRA Add infra owners file 2022-12-02 09:21:47 +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
README.md doc: add g3doc sitemap to toplevel readme 2021-07-23 07:55:17 +00:00
WATCHLISTS Remove xooglers from WATCHLISTS and OWNERS 2022-11-30 15:33:25 +00:00
webrtc.gni Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
webrtc_lib_link_test.cc Replace BuiltinVideo{Encoder,Decoder}Factory with Video{Encoder,Decoder}FactoryTemplate. 2023-03-31 19:31:50 +00:00
whitespace.txt Trigger bots 2023-05-16 08:24:54 +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