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henrik.lundin@webrtc.org b6fab2b1cd Introduce rtc::CheckedDivExact 
Use the new method to replace local ones in AudioEncoder{Opus,Isac}.

COAUTHOR:kwiberg@webrtc.org

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8148 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 11:08:53 +00:00
..
base
Introduce rtc::CheckedDivExact
2015-01-26 11:08:53 +00:00
build
Roll chromium_revision c086b4e..4664fe0 (312108:312733)
2015-01-25 19:17:56 +00:00
common_audio
Add beamforming to audioproc_float utility.
2015-01-15 01:28:36 +00:00
common_video
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
examples
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
libjingle
Change GetStreamBySsrc to not copy StreamParams.
2015-01-22 23:00:41 +00:00
modules
Introduce rtc::CheckedDivExact
2015-01-26 11:08:53 +00:00
overrides
Remove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 10:51:54 +00:00
p2p
Reland r7980:
2015-01-23 17:33:34 +00:00
sound
…
system_wrappers
Change a GYP reference to cpufeatures.gypi
2015-01-23 16:35:17 +00:00
test
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
tools
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
video
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
2015-01-22 09:39:59 +00:00
video_engine
Revert 8136 "Remove frame copy in ViEExternalRendererImpl::Rende..."
2015-01-23 21:20:41 +00:00
voice_engine
Re-allowing RED in voice engine.
2015-01-23 11:58:42 +00:00
.gitignore
…
BUILD.gn
Roll chromium_revision c086b4e..4664fe0 (312108:312733)
2015-01-25 19:17:56 +00:00
call.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
codereview.settings
…
common_types.h
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct.
2015-01-22 09:39:59 +00:00
common.gyp
…
common.h
…
config.cc
…
config.h
…
engine_configurations.h
…
experiments.h
…
frame_callback.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
…
video_encoder.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
video_engine_tests.isolate
…
video_frame.h
…
video_receive_stream.h
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
video_renderer.h
…
video_send_stream.h
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
webrtc_examples.gyp
Move internal capture+render to build_with_chromium==0 condition
2015-01-20 11:40:45 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
webrtc.gyp
Remove unnecessary dependencies from webrtc_all target.
2015-01-21 10:06:55 +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.