signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Java 2.8%
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Benjamin Wright e9bece37ae Minor change to the Json Config format for the replay file. 
See: test/fuzzers/configs/replay_packet_fuzzer for example configurations.

Bug: webrtc:10117
Change-Id: Ife2bf7d053bc4feb4d7e6e38ff31280236c962b6
Reviewed-on: https://webrtc-review.googlesource.com/c/116764
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26204}
2019-01-10 18:38:08 +00:00
api Add PeerConnection option to enable RTX handling in the audio jitter buffer. 2019-01-10 16:28:43 +00:00
audio Add PeerConnection option to enable RTX handling in the audio jitter buffer. 2019-01-10 16:28:43 +00:00
build_overrides Disable hermetic toolchain when building on macOS 10.14. 2018-12-07 13:57:28 +00:00
call Add PeerConnection option to enable RTX handling in the audio jitter buffer. 2019-01-10 16:28:43 +00:00
common_audio [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 2018-11-28 18:25:07 +00:00
common_video Introduce VideoFrame::id to keep track of frames inside application. 2019-01-04 08:59:26 +00:00
data Remove old data files. 2018-10-05 14:40:21 +00:00
examples Fix macOS demo privacy crash. 2019-01-10 10:50:49 +00:00
infra Re-add MSVC debug bots to CQ 2018-12-05 12:20:56 +00:00
logging Parse logs without RTX SSRC even if there is an RTX payload type. 2019-01-10 14:43:39 +00:00
media Add PeerConnection option to enable RTX handling in the audio jitter buffer. 2019-01-10 16:28:43 +00:00
modules Refactor FrameBuffer to store decoded frames history separately 2019-01-10 15:11:15 +00:00
p2p Include absl/memory/memory.h if absl::make_unique is used 2019-01-08 20:08:32 +00:00
pc Add PeerConnection option to enable RTX handling in the audio jitter buffer. 2019-01-10 16:28:43 +00:00
resources Base ApmTest.Process on AudioProcessingStats.output_rms_dbfs 2018-12-18 16:45:03 +00:00
rtc_base Remove TaskQueue::PostAndReply as unused 2019-01-10 16:06:57 +00:00
rtc_tools Delete NetEqTest::ExtDecoderMap 2019-01-08 16:25:05 +00:00
sdk Include absl/memory/memory.h if absl::make_unique is used 2019-01-08 20:08:32 +00:00
stats Add ',' between elements in RTCStatsReport::ToJson 2019-01-08 16:05:23 +00:00
style-guide Add style guide rule about paired .h and .cc files 2018-03-14 13:02:35 +00:00
system_wrappers [Fuchsia] Implement detection of available cores. 2018-12-17 19:37:46 +00:00
test Revert "Add a high bitrate full stack test with fake codec" 2019-01-10 11:49:05 +00:00
tools_webrtc mb: Turn LUCI desktop perf bots into builder + tester pairs 2018-12-21 13:43:04 +00:00
video Minor change to the Json Config format for the replay file. 2019-01-10 18:38:08 +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 Reland "Default to dlopening the PipeWire." 2019-01-08 08:09:42 +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 Remove CodecInst pt.3 2018-12-18 07:42:21 +00:00
DEPS Roll chromium_revision 4a7f224d18..6e83997c8b (621417:621525) 2019-01-10 10:52:19 +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 Reland "Default to dlopening the PipeWire." 2019-01-08 08:09:42 +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