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webrtc_m130/webrtc
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danilchap 34877eeec9 Revert of Added validation between RTP and RTCP timestamps (patchset #7 id:120001 of https://codereview.webrtc.org/1633843003/ ) 
Reason for revert:
May be the reason for mac_asan timeout

Original issue's description:
> Changed test to validate rtp timstamps not just in RTP packets but also in RTCP Sender Reports.
> Altered it to accept negative value since it is normal for RTCP packet coming before RTP packet to have slightly later time.
>
> BUG=webrtc:5433
>
> Committed: https://crrev.com/f4b9c775122b463db7eb2c4101603759a0d00caf
> Cr-Commit-Position: refs/heads/master@{#11417}

TBR=pbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5433

Review URL: https://codereview.webrtc.org/1652973002

Cr-Commit-Position: refs/heads/master@{#11446}
2016-02-01 16:25:08 +00:00
..
api
Make ECDSA default for RTCPeerConnection
2016-01-28 01:50:15 +00:00
audio
Use separate rtp module lists for send and receive in PacketRouter.
2016-02-01 12:40:04 +00:00
base
Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015).
2016-01-30 22:40:52 +00:00
build
H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding.
2016-01-27 09:36:07 +00:00
call
Use separate rtp module lists for send and receive in PacketRouter.
2016-02-01 12:40:04 +00:00
common_audio
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
common_video
H.264 video codec support using OpenH264 (http://www.openh264.org/) for encoding and FFmpeg (https://www.ffmpeg.org/) for decoding.
2016-01-27 09:36:07 +00:00
examples
Revert of Adding "first packet received" notification to PeerConnectionObserver. (patchset #5 id:80001 of https://codereview.webrtc.org/1581693006/ )
2016-01-28 13:06:16 +00:00
libjingle
Remove DISABLED_ON_ macros.
2016-01-04 21:44:16 +00:00
media
New rtc::VideoSinkInterface.
2016-01-28 12:47:13 +00:00
modules
Prevent zero division in VCMJitterEstimator.
2016-02-01 15:31:17 +00:00
p2p
Stop processing any incoming packets when turn port is disconnected.
2016-01-29 21:22:36 +00:00
sound
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
system_wrappers
Use rtc::time for all your timing needs!
2016-01-27 11:55:44 +00:00
test
Use separate rtp module lists for send and receive in PacketRouter.
2016-02-01 12:40:04 +00:00
tools
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
video
Revert of Added validation between RTP and RTCP timestamps (patchset #7 id:120001 of https://codereview.webrtc.org/1633843003/ )
2016-02-01 16:25:08 +00:00
voice_engine
Use separate rtp module lists for send and receive in PacketRouter.
2016-02-01 12:40:04 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Storing raw audio sink for default audio track.
2016-01-15 17:20:08 +00:00
audio_send_stream.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
call.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
codereview.settings
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
common.gyp
…
common.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
2015-11-25 21:20:11 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Added EncodedImage::GetBufferPaddingBytes.
2016-01-21 13:43:18 +00:00
video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_renderer.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
video_send_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
webrtc_examples.gyp
Disable warnings failing when using Clang on Windows.
2015-12-16 22:05:36 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Make ECDSA default for RTCPeerConnection
2016-01-28 01:50:15 +00:00
webrtc.gyp
Make ECDSA default for RTCPeerConnection
2016-01-28 01:50:15 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.