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webrtc_m130/webrtc
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danilchap e565a04de3 Revert of Fix crash parsing malformed rtp packet (patchset #1 id:1 of https://codereview.webrtc.org/2067793003/ ) 
Reason for revert:
breaks Win64 bots compile

Original issue's description:
> Fix crash parsing malformed rtp packet
> where header extesnsion size mismatch expected.
>
> BUG=chromium:620242
> R=asapersson@webrtc.org
>
> Committed: https://crrev.com/5a45fe6fd7a509fb4c3a9b09cdbd2278055f1d4c
> Cr-Commit-Position: refs/heads/master@{#13170}

TBR=asapersson@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:620242

Review-Url: https://codereview.webrtc.org/2074763002
Cr-Commit-Position: refs/heads/master@{#13171}
2016-06-16 17:04:57 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
Delete unused code.
2016-06-16 13:51:40 +00:00
audio
Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test.
2016-06-14 19:13:07 +00:00
base
Performance fix for H264 RBSP parsing.
2016-06-16 13:54:58 +00:00
build
GN: Add modules_unittests
2016-06-13 07:19:53 +00:00
call
Changes synchronization offset perfomance tracking
2016-06-16 07:58:58 +00:00
common_audio
GN: Add common_audio_unittests and common_video_unittests
2016-06-08 08:28:15 +00:00
common_video
Performance fix for H264 RBSP parsing.
2016-06-16 13:54:58 +00:00
examples
Add RTCEventLog API to ObjC.
2016-06-14 22:03:19 +00:00
libjingle
Adding more detail to MessageQueue::Dispatch logging.
2016-06-10 21:17:33 +00:00
media
Delete GetExecutablePath and related unused code.
2016-06-16 10:44:44 +00:00
modules
Revert of Fix crash parsing malformed rtp packet (patchset #1 id:1 of https://codereview.webrtc.org/2067793003/ )
2016-06-16 17:04:57 +00:00
p2p
Change initial DTLS retransmission timer from 1 second to 50ms.
2016-06-16 00:15:35 +00:00
pc
Delete GetExecutablePath and related unused code.
2016-06-16 10:44:44 +00:00
sdk
Add RTCEventLog API to ObjC.
2016-06-14 22:03:19 +00:00
system_wrappers
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
test
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
tools
New misc scripts, header_usage.sh and author_line_count.sh.
2016-06-15 07:06:32 +00:00
video
This cl:
2016-06-15 07:47:58 +00:00
voice_engine
Delete unused code.
2016-06-16 13:51:40 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test.
2016-06-14 19:13:07 +00:00
audio_send_stream.h
Configure VoE NACK through AudioSendStream::Config, for send streams.
2016-06-14 17:02:46 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +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
GN: Add video_engine_tests
2016-06-14 10:53:09 +00:00
call.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Configure VoE NACK through AudioSendStream::Config, for send streams.
2016-06-14 17:02:46 +00:00
config.h
Configure VoE NACK through AudioSendStream::Config, for send streams.
2016-06-14 17:02:46 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
Add kwiberg@webrtc.org as root owner.
2016-06-14 09:39:40 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 086802955f..7fa6701bc5 (396351:398458)
2016-06-08 11:05:27 +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
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +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
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
video_receive_stream.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Remove metrics_default from rtc_media dependencies.
2016-06-01 12:28:57 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
GN: Add SDK tests to rtc_unittests.
2016-06-08 08:14:22 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.