admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
danilchap e441bdb744 Cleanup RtpSender hiding RtpHeaderExtensionLength function. 
This function has no public use,
removed tests calling it: effect of registering extension is better
tested in AllocatePacket and SendPacket tests.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2530363002
Cr-Commit-Position: refs/heads/master@{#15258}
2016-11-28 10:55:01 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Change rtc::TimeNanos and rtc::TimeMicros return value from uint64_t to int64_t.
2016-11-28 09:55:05 +00:00
audio
Replace AudioConferenceMixer with AudioMixer.
2016-11-22 14:42:59 +00:00
base
Roll chromium_revision 5e821a778b..5c22c2afac (432715:434448)
2016-11-28 10:04:45 +00:00
build
Make Valgrind memcheck work in swarming.
2016-11-25 13:49:00 +00:00
call
Start probes only after network is connected.
2016-11-23 00:08:37 +00:00
common_audio
Smooth BWE and pass it to Audio Network Adaptor.
2016-11-28 10:34:23 +00:00
common_video
Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:1 of https://codereview.webrtc.org/2529143002/ )
2016-11-25 18:06:35 +00:00
examples
Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ )
2016-11-23 15:12:28 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
modules
Cleanup RtpSender hiding RtpHeaderExtensionLength function.
2016-11-28 10:55:01 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
Let MediaSession generate a FlexFEC SSRC when FlexFEC is active.
2016-11-22 11:38:04 +00:00
sdk
iOS HW encoder: Enable H264 High profile support
2016-11-25 15:32:37 +00:00
stats
Change rtc::TimeNanos and rtc::TimeMicros return value from uint64_t to int64_t.
2016-11-28 09:55:05 +00:00
system_wrappers
Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.
2016-11-17 10:27:20 +00:00
test
Roll chromium_revision 5e821a778b..5c22c2afac (432715:434448)
2016-11-28 10:04:45 +00:00
tools
Rename "Audio playout level" to "Audio level" on the Y-axis of the event log graph.
2016-11-24 16:30:42 +00:00
video
Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:1 of https://codereview.webrtc.org/2529143002/ )
2016-11-25 18:06:35 +00:00
voice_engine
Smooth BWE and pass it to Audio Network Adaptor.
2016-11-28 10:34:23 +00:00
.gitignore
…
BUILD.gn
Get rid of webrtc/base/latebindingsymboltable*
2016-11-25 14:47:15 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
Unify VideoCodecType to/from string functionality
2016-11-22 18:17:04 +00:00
common_types.h
Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:1 of https://codereview.webrtc.org/2529143002/ )
2016-11-25 18:06:35 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
Remove all references to GYP
2016-11-16 19:11:38 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
…
typedefs.h
…
video_decoder.h
Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
2016-11-22 09:43:06 +00:00
video_encoder.h
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
2016-11-18 09:34:14 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
video_send_stream.h
Sent bitrate stats are incorrect if FlexFEC is configured:
2016-11-25 11:52:55 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +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.