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Aaron Golden 154a262b61 Don't clear self.videoFrame when setting up OpenGL in the EAGL video view.
It makes sense to clean up self.videoFrame in -teardownGL, but if
we happen to have a frame available in -setupGL then it's OK to
keep using that frame. Clearing the frame here frequently causes
the screen view to go black for a moment when the app returns from
the background.

Bug: webrtc:10059
Change-Id: Ic62f872a0a582c807cee1e30ea9bb32f31ada341
Reviewed-on: https://webrtc-review.googlesource.com/c/112213
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25816}
2018-11-28 09:00:06 +00:00
api Add PeerConnection option to configure minimum audio jitter buffer delay. 2018-11-27 19:49:48 +00:00
audio Add PeerConnection option to configure minimum audio jitter buffer delay. 2018-11-27 19:49:48 +00:00
build_overrides nit: Missing space in build_overrides/build.gni 2018-11-15 14:17:12 +00:00
call Add PeerConnection option to configure minimum audio jitter buffer delay. 2018-11-27 19:49:48 +00:00
common_audio Delete unneeded includes of common_types.h and gn deps on webrtc_common. 2018-11-20 16:28:39 +00:00
common_video Delete deprecated class WrappedI420Buffer 2018-11-13 10:59:10 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Adding WinUWP compilation support to WebRTC. 2018-11-28 08:32:30 +00:00
infra Remove ios32_sim_ios9_dbg from CQ. 2018-10-15 06:59:19 +00:00
logging Remove RSID from stream configs in new event log format. 2018-11-22 17:54:06 +00:00
media Add PeerConnection option to configure minimum audio jitter buffer delay. 2018-11-27 19:49:48 +00:00
modules AEC3: Add metrics for API call jitter 2018-11-27 19:52:08 +00:00
p2p Reland "Delay call to Destroy until after SignalDone has finished firing." 2018-11-28 02:55:11 +00:00
pc Add PeerConnection option to configure minimum audio jitter buffer delay. 2018-11-27 19:49:48 +00:00
resources Removing ancient and unused test scripts and data files 2018-11-05 16:08:46 +00:00
rtc_base Adding WinUWP compilation support to WebRTC. 2018-11-28 08:32:30 +00:00
rtc_tools Add magjed as owner of rtc_tools. 2018-11-26 11:30:34 +00:00
sdk Don't clear self.videoFrame when setting up OpenGL in the EAGL video view. 2018-11-28 09:00:06 +00:00
stats Expose delayed packet outage as a cumulative metric of samples in the new getStats API. 2018-11-27 15:10:09 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers Adding WinUWP compilation support to WebRTC. 2018-11-28 08:32:30 +00:00
test Adding WinUWP compilation support to WebRTC. 2018-11-28 08:32:30 +00:00
tools_webrtc Fix webrtc-internal ios json config 2018-11-28 08:53:26 +00:00
video Move RtcpStatistics from common_types.h to a new header file 2018-11-27 13:46:42 +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 cstring include need for strncmp 2018-11-26 20:49:36 +00:00
BUILD.gn Decouple //rtc_base:rtc_base_tests_utils from gunit. 2018-11-23 12:52:46 +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 RtcpStatistics from common_types.h to a new header file 2018-11-27 13:46:42 +00:00
DEPS Roll chromium_revision b04e513f82..82a8b043ef (611432:611537) 2018-11-28 03:36:36 +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
native-api.md Add documentation about field_trial/metrics custom impl. 2018-09-18 11:27:59 +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.py Add a presubmit check for absl/memory/memory.h inclusion 2018-11-15 12:15:40 +00:00
presubmit_test.py Fixing py lint errors 2018-07-23 15:28:48 +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 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 Adding WinUWP compilation support to WebRTC. 2018-11-28 08:32:30 +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