[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Mirko Bonadei f0d9cda950 Revert "AEC3: Lockless transfer of render data to the capture thread"
This reverts commit 74ba99062c.

Reason for revert: Breaks downstream project.

Original change's description:
> AEC3: Lockless transfer of render data to the capture thread
> 
> This CL implements a lockless queue that replaces SwapQueue
> in the RenderWriter. This avoid stalls when the render and
> capture threads are accessing the queue at the same time.
> 
> Bug: webrtc:10205
> Change-Id: Ie7d6fcf9c80fad957e2a90537658fb730ca2ed72
> Reviewed-on: https://webrtc-review.googlesource.com/c/117643
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26298}

TBR=gustaf@webrtc.org,peah@webrtc.org

Change-Id: Ie76ee8835da4e44982d181a152c9ffa19ff33e23
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10205
Reviewed-on: https://webrtc-review.googlesource.com/c/118142
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26304}
2019-01-17 20:44:06 +00:00
api Add ability for VideoEncoder to signal frame rate allocation. 2019-01-17 15:40:53 +00:00
audio Fix chromium roll into WebRTC. 2019-01-16 15:19:57 +00:00
build_overrides Disable hermetic toolchain when building on macOS 10.14. 2018-12-07 13:57:28 +00:00
call Expose jitterBufferEmittedCount in addition to the existing jitterBufferDelay for getStats(). 2019-01-16 11:44:10 +00:00
common_audio (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
common_video (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Remove legacy video codec factories. 2019-01-14 14:56:40 +00:00
infra Re-add MSVC debug bots to CQ 2018-12-05 12:20:56 +00:00
logging Add DTLSTransport info into sender/receiver state. 2019-01-17 10:21:32 +00:00
media Add ability for VideoEncoder to signal frame rate allocation. 2019-01-17 15:40:53 +00:00
modules Revert "AEC3: Lockless transfer of render data to the capture thread" 2019-01-17 20:44:06 +00:00
p2p DTLS transport needs to forward network route change. 2019-01-17 19:25:25 +00:00
pc Add DTLSTransport info into sender/receiver state. 2019-01-17 10:21:32 +00:00
resources Opus multistream. 2019-01-17 12:23:23 +00:00
rtc_base Introduce webrtc::TaskQueue and TaskQueueFactory interfaces 2019-01-17 13:10:14 +00:00
rtc_tools (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries 2019-01-11 17:11:39 +00:00
sdk Move some video codec constants to separate file. 2019-01-17 15:29:53 +00:00
stats Expose jitterBufferEmittedCount in addition to the existing jitterBufferDelay for getStats(). 2019-01-16 11:44:10 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Remove definition of macro WEBRTC_THREAD_RR (it's unused). 2019-01-14 15:01:38 +00:00
test Introduce ExampleVideoQualityAnalyzer. 2019-01-17 16:50:38 +00:00
tools_webrtc Increase timeout for internal iOS perf, like for public android and desktop 2019-01-17 13:46:18 +00:00
video Move some video codec constants to separate file. 2019-01-17 15:29:53 +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 Reland "Compile frame analyzer for the host machine on perf tests." 2018-09-18 09:51:19 +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 Rebase std::is_trivially_* with absl::is_trivially_* 2018-11-26 19:20:27 +00:00
AUTHORS Add ',' between elements in RTCStatsReport::ToJson 2019-01-08 16:05:23 +00:00
BUILD.gn Remove legacy video codec factories. 2019-01-14 14:56:40 +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 some video codec constants to separate file. 2019-01-17 15:29:53 +00:00
DEPS Roll chromium_revision 3a9216d2da..e291b99915 (623639:623778) 2019-01-17 19:55:56 +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 Clean up root OWNERS. 2018-11-09 14:23:59 +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 Delete WebRtcVideoCapturer and related classes. 2019-01-03 15:15:48 +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 style rule about not using const optional<T>& arguments 2018-11-08 11:57:35 +00:00
WATCHLISTS Remove likely obsolete entries from WATCHLISTS 2018-10-30 07:46:29 +00:00
webrtc.gni Roll //third_party/ffmpeg and disable H264 when MSVC is used. 2019-01-15 10:25:01 +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