admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
mflodman@webrtc.org 0a7d4eed98 Remove usage of BitrateController in VoiceEngine. 
Bitrate controller is used in VoiceEngine to smoothen the fraction loss
from RTCP report blocks. This CL removes the usage of the
BitrateController and calculates its own fraction loss average insted.
This introduces some duplicated code between BitrateController and
Channel, but removes processing overhead and the incorrect bandwidth
estimation numbers reported by the bitrate controller.

BUG=4310
TEST=voe_cmd_test with network simulator
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39999004

Cr-Commit-Position: refs/heads/master@{#8386}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8386 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 12:57:44 +00:00
..
base
Remove the incremental IP address behavior from virtualsocketserver
2015-02-14 00:44:20 +00:00
build
GN: Changes for vp9, opus and direct trace
2015-02-16 12:47:45 +00:00
common_audio
WebRtc_GetCPUFeaturesARM is only available on android
2015-02-11 17:03:24 +00:00
common_video
GN: Changes for vp9, opus and direct trace
2015-02-16 12:47:45 +00:00
examples
This is a code clean up. No functional change intended.
2015-02-13 14:32:13 +00:00
libjingle
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
modules
Removing CELT.
2015-02-17 12:37:14 +00:00
overrides
Add arraysize to overrides to avoid macros redefinitions in Chromium
2015-02-14 02:46:13 +00:00
p2p
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
sound
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
system_wrappers
GN: Changes for vp9, opus and direct trace
2015-02-16 12:47:45 +00:00
test
Remove EventWrapper::Reset().
2015-02-13 14:58:38 +00:00
tools
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
video
Remove EventWrapper::Reset().
2015-02-13 14:58:38 +00:00
video_engine
Remove TimeToSendPacket and TimeToSendPadding from the default module.
2015-02-17 10:15:47 +00:00
voice_engine
Remove usage of BitrateController in VoiceEngine.
2015-02-17 12:57:44 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.h
Add method for incrementing RtpPacketCounter. Removes duplicate code.
2015-02-04 08:35:21 +00:00
common.gyp
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.cc
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
config.h
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.
2014-05-23 17:27:18 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +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 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Enable Clang warning implicit-fallthrough and annotate the code.
2015-01-28 18:38:13 +00:00
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Remove default arguments in EncodedImageCallback.
2015-02-09 09:14:48 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
CVO capturer feature: allow unrotated frame flows through the capture pipeline.
2015-02-11 18:38:53 +00:00
video_receive_stream.h
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
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
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.