admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
Peter Boström 59b2d3ebe7 Remove zero-divide in VCMContentMetricsProcessing. 
Protects against zero-framerate in UpdateFrameRate.

BUG=webrtc:5124
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1636943003 .

Cr-Commit-Position: refs/heads/master@{#11382}
2016-01-26 15:18:56 +00:00
..
api
Update API for Objective-C RTCDataChannel.
2016-01-22 22:04:33 +00:00
audio
Remove mutable from rtc::CriticalSection members.
2016-01-25 11:52:53 +00:00
base
Remove non-monotonic clock support
2016-01-26 09:53:24 +00:00
build
rtc_use_h264 flag (replacing use_third_party_h264 flag) for building OpenH264/FFmpeg, false by default but can be overridden in supplement.gypi and build_overrides/webrtc.gni.
2016-01-21 11:34:47 +00:00
call
Switch to use new implementation in metrics.h.
2016-01-25 13:58:27 +00:00
common_audio
…
common_video
Remove mutable from rtc::CriticalSection members.
2016-01-25 11:52:53 +00:00
examples
Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set.
2016-01-15 22:49:15 +00:00
libjingle
…
modules
Remove zero-divide in VCMContentMetricsProcessing.
2016-01-26 15:18:56 +00:00
p2p
Remove two dead 'using' instances.
2016-01-18 23:24:36 +00:00
sound
…
system_wrappers
Remove non-monotonic clock support
2016-01-26 09:53:24 +00:00
test
Remove mutable from rtc::CriticalSection members.
2016-01-25 11:52:53 +00:00
tools
…
video
Switch to use new implementation in metrics.h.
2016-01-25 13:58:27 +00:00
voice_engine
Swap use of CriticalSectionWrapper with rtc::CriticalSection in voice_engine/
2016-01-21 18:37:44 +00:00
.gitignore
…
audio_receive_stream.h
Storing raw audio sink for default audio track.
2016-01-15 17:20:08 +00:00
audio_send_stream.h
…
audio_state.h
…
BUILD.gn
…
call.h
…
codereview.settings
…
common_types.cc
…
common_types.h
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
common.gyp
…
common.h
…
config.cc
…
config.h
…
engine_configurations.h
…
frame_callback.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
…
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
…
video_encoder.h
…
video_engine_tests.isolate
…
video_frame.h
Added EncodedImage::GetBufferPaddingBytes.
2016-01-21 13:43:18 +00:00
video_receive_stream.h
…
video_renderer.h
…
video_send_stream.h
…
webrtc_examples.gyp
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
webrtc.gyp
…

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.