signalapp/webrtc
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[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
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Sergey Silkin 1daa7e8729 Use RTP timestamp when checking for frame duplication. 
Value of render timestamp can be the same for consecutive frames (e.g.
when old frames got decoded and need to be rendered immediately). It
should not be used for frame duplication checking.

Bug: b/122636276
Change-Id: Ie00bdd3fa50a2eacd48cba228fa3c54a6b206864
Reviewed-on: https://webrtc-review.googlesource.com/c/116790
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26210}
2019-01-11 08:52:13 +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 AEC3: More efficient comfort noise generation 2019-01-11 08:46:05 +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 method DecoderDatabase::RegisterPayload(...NetEqDecoder...) 2019-01-11 07:39:45 +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 Delete method DecoderDatabase::RegisterPayload(...NetEqDecoder...) 2019-01-11 07:39:45 +00:00
tools_webrtc mb: Turn LUCI desktop perf bots into builder + tester pairs 2018-12-21 13:43:04 +00:00
video Use RTP timestamp when checking for frame duplication. 2019-01-11 08:52:13 +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 8c0cc38022..783044b798 (621736:621838) 2019-01-11 01:38:22 +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