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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Qiang Chen 6d30631f01 Bug Fix: Multiplex Codec Crash 
This CL adds a lock to stashed_images_ in MultiplexEncoderAdapter.
Without lock, it is possible that different threads acts on
stashed_images_ simultaneously and leads to crash.

Bug: webrtc:8965
Change-Id: I887861092d185c3bd6047eb529d8c1cf57fa4648
Reviewed-on: https://webrtc-review.googlesource.com/59260
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22261}
2018-03-02 00:14:48 +00:00
api Audio codec implementations: Take optional codec ID argument 2018-03-01 19:28:38 +00:00
audio Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 1. 2018-02-28 15:38:15 +00:00
build_overrides Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
call Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." 2018-03-01 18:07:29 +00:00
common_audio Conditionally include real_fourier_openmax.h. 2018-02-23 09:02:36 +00:00
common_video Drop dependency of common_video on api:libjingle_peerconnection_api. 2018-02-19 13:20:24 +00:00
data WebRTC: Replace ProjectRootPath by ResourcePath 2016-11-22 18:43:05 +00:00
examples Update iOS AppRTC to use PeerConnection Unified Plan 2018-03-01 17:25:18 +00:00
infra Removing android_compile_mips_dbg from CQ. 2018-03-01 09:13:33 +00:00
logging Add extra RTC_CHECKs in RtcEventLog. 2018-02-27 13:23:49 +00:00
media Remove public sync_label from StreamParams 2018-03-01 18:25:03 +00:00
modules Bug Fix: Multiplex Codec Crash 2018-03-02 00:14:48 +00:00
ortc Removed new calls on RtpTransportControllerSend. 2018-02-22 12:54:43 +00:00
p2p Add method to modify magic cookie of a STUN message 2018-02-28 07:02:10 +00:00
pc Audio codec implementations: Take optional codec ID argument 2018-03-01 19:28:38 +00:00
resources Adding FourPeople_1280x720_30.yuv. 2018-02-12 15:55:00 +00:00
rtc_base Renamed rtc::RateLimiter to rtrc::DataRateLimiter. 2018-02-28 19:58:24 +00:00
rtc_tools Preparing for task queue in congenstion controller 2018-02-20 12:35:15 +00:00
sdk Updating android/README. 2018-03-01 20:22:48 +00:00
stats Removing useless dependencies on //testing/gmock. 2018-01-26 13:34:12 +00:00
system_wrappers Fixing issues that affect newest versions of NDK. 2018-02-27 10:30:43 +00:00
test Revert "Adding gtest-spi.h in webrtc/test/gtest.h" 2018-02-28 15:19:07 +00:00
tools_webrtc Delete unused sample project code. 2018-02-20 09:51:52 +00:00
video Reland "Replaced temporal_layer_thresholds_bps[] field with num_temporal_layers." 2018-03-01 18:07:29 +00:00
.clang-format Tune ObjC clang-format configuration 2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs Create .git-blame-ignore-revs and add Java format CL to it. 2016-10-20 09:20:39 +00:00
.gitignore Reland "Make it possible to run video_quality_loopback_test in swarming." 2018-01-23 13:03:17 +00:00
.gn Re-enabling 'gn check' on //examples/*. 2018-02-19 15:07:45 +00:00
.vpython vpython: Specify dependency on pywin32 2018-02-14 13:56:39 +00:00
AUTHORS Expose a link-local network interfaces enumeration option 2018-02-06 19:12:04 +00:00
BUILD.gn Removing definition of FEATURE_ENABLE_VOICEMAIL. 2018-02-19 15:51:24 +00:00
CODE_OF_CONDUCT.md Add code of conduct to WebRTC repo 2017-05-16 12:09:13 +00:00
codereview.settings Make Gerrit the default for WebRTC changes 2017-09-29 01:38:07 +00:00
common_types.cc Add flexfec payload name to string-type conversions 2018-01-31 08:58:39 +00:00
common_types.h Remove unused fields from VideoCodecVP8. 2018-02-09 15:55:59 +00:00
DEPS Roll chromium_revision a410274678..2eea119a99 (539759:540206) 2018-03-01 18:42:58 +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
LICENSE_THIRD_PARTY Remove custom MD5 / SHA-1 implementations. 2018-02-19 15:03:35 +00:00
native-api.md Remove legacy VoiceEngine. 2018-01-12 11:31:52 +00:00
OWNERS Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
PATENTS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
presubmit_test.py Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks. 2017-10-13 03:48:26 +00:00
PRESUBMIT.py Delete unused MediaFile module. 2018-01-29 11:18:18 +00:00
pylintrc Removing invalid-name from disabled pylint checks. 2017-10-11 08:06:49 +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 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" 2018-01-10 15:55:04 +00:00
typedefs.h Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix 2018-02-05 11:24:59 +00:00
WATCHLISTS Move remaining traces of VoiceEngine 2018-01-17 13:27:47 +00:00
webrtc.gni Removing obsolete defines. 2018-02-19 14:35:45 +00:00
whitespace.txt Whitespace change 2018-02-23 10:34:16 +00:00

README.md

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