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webrtc_m130/webrtc
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asapersson fdca66910a Potential division by zero in RtpToNtpMs() in rtp_to_ntp.cc. 
CalculateFrequency() results in zero frequency (floating point) if the RTP timestamps in the RTCP list are equal.
Added check in UpdateRtcpList to not insert RTCP SR with the same RTP timestamp.

BUG=webrtc:5780

Review URL: https://codereview.webrtc.org/1891703002

Cr-Commit-Position: refs/heads/master@{#12429}
2016-04-19 14:04:52 +00:00
..
api
Add rotation to EncodedImage and make sure it is passed through encoders.
2016-04-19 13:01:32 +00:00
audio
Reland "Add check_deps rules in DEPS files."
2016-03-17 11:05:50 +00:00
base
Roll chromium_revision 212f976fef..61ed337cfe (387882:388120)
2016-04-19 10:03:31 +00:00
build
Fix coverage build.
2016-04-19 06:08:15 +00:00
call
Disabling SwitchesToASTThenBackToTOFForVideo test for MSan bot.
2016-04-18 23:19:02 +00:00
common_audio
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
common_video
Delete unused methods SetStartImage and SetTimeoutImage.
2016-04-19 13:19:47 +00:00
examples
Use the new appr.tc URL
2016-04-15 14:16:19 +00:00
libjingle
Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ )
2016-03-31 14:23:52 +00:00
media
Use initial bitrates for software VP8.
2016-04-19 13:53:22 +00:00
modules
Use initial bitrates for software VP8.
2016-04-19 13:53:22 +00:00
p2p
Make ReliableQuicStream::Write use base::StringPiece, not std::string
2016-04-14 18:12:51 +00:00
pc
Revert of Accept all the media profiles required by JSEP. (patchset #5 id:80001 of https://codereview.webrtc.org/1880913002/ )
2016-04-13 17:48:32 +00:00
system_wrappers
Potential division by zero in RtpToNtpMs() in rtp_to_ntp.cc.
2016-04-19 14:04:52 +00:00
test
Add rotation to EncodedImage and make sure it is passed through encoders.
2016-04-19 13:01:32 +00:00
tools
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video
Add rotation to EncodedImage and make sure it is passed through encoders.
2016-04-19 13:01:32 +00:00
voice_engine
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Replace scoped_ptr with unique_ptr in webrtc/audio/
2016-02-23 18:46:39 +00:00
audio_send_stream.h
Relanding https://codereview.webrtc.org/1715883002/ in pieces.
2016-03-11 11:06:48 +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
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
call.h
Make the audio channel communicate network state changes to the call.
2016-03-22 22:32:31 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +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
Tune BWE to be a bit less sensitive to spurious delay events.
2016-03-10 13:13:26 +00:00
common.gyp
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
common.h
Added support in the AEC for refined filter adaptation.
2016-04-15 18:23:36 +00:00
config.cc
Change transport sequence number extension strings to specify what revision is implemented.
2016-03-01 15:27:27 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
DEPS
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Make pbos owner of additional video files.
2016-02-26 12:29:17 +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
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Remove use_openssl from webrtc
2016-04-18 18:13:17 +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
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +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 rotation to EncodedImage and make sure it is passed through encoders.
2016-04-19 13:01:32 +00:00
video_receive_stream.h
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
video_send_stream.h
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +00:00
webrtc_examples.gyp
Add test runner scripts for instrumentation tests
2016-04-08 13:54:32 +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
…
webrtc_tests.gypi
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc.gyp
Move frame_callback.h to common_video/include.
2016-04-19 04:12:57 +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.