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stefan@webrtc.org 24bd364d3e Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels. 
This fixes an issue where the user doesn't know which channels are "active" and therefore can't properly sum the estimates for all channels.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6041 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-02 12:35:37 +00:00
..
build
Remove neteq_unittests from Android builds
2014-04-29 09:49:51 +00:00
common_audio
Fix constness of AudioBuffer accessors.
2014-04-30 16:44:13 +00:00
common_video
Added include of assert.h for files calling assert but missing the include.
2014-04-29 17:54:17 +00:00
examples
Replace scoped_array<T> with scoped_ptr<T[]>.
2014-04-25 23:10:28 +00:00
modules
Only clamp to 16 kHz when AECM is enabled.
2014-04-30 18:58:23 +00:00
system_wrappers
Replace scoped_array<T> with scoped_ptr<T[]>.
2014-04-25 23:10:28 +00:00
test
Added include of assert.h for files calling assert but missing the include.
2014-04-29 17:54:17 +00:00
tools
Replace scoped_array<T> with scoped_ptr<T[]>.
2014-04-25 23:10:28 +00:00
video
Add thread annotations to Call API.
2014-04-28 13:00:21 +00:00
video_engine
Change GetEstimatedSend/RecvBandwidth to return the total bandwidth of a channel group instead of splitting it up among channels.
2014-05-02 12:35:37 +00:00
voice_engine
Only clamp to 16 kHz when AECM is enabled.
2014-04-30 18:58:23 +00:00
.gitignore
…
call.h
Remove TraceCallback use from Call.
2014-04-24 11:35:33 +00:00
common_types.h
Made common_types.h PacketTime declaration match https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65
2014-04-29 17:50:47 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
engine_configurations.h
Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.
2014-04-14 14:12:50 +00:00
experiments.h
Adding API for setting bandwidth estimation configurations.
2014-03-25 10:37:31 +00:00
frame_callback.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
2014-04-14 20:08:03 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Roll chromium_revision 260462:266514
2014-04-29 09:36:40 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Fix ARM64 detection.
2014-04-01 01:19:08 +00:00
video_engine_tests.isolate
Merge metrics_unittests into video_engine_tests.
2013-12-13 14:31:47 +00:00
video_receive_stream.h
Rename Start/Stop in Video{Send,Receive}Streams.
2014-04-24 11:13:21 +00:00
video_renderer.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
video_send_stream.h
Rename Start/Stop in Video{Send,Receive}Streams.
2014-04-24 11:13:21 +00:00
webrtc_examples.gyp
Make WebRTC Android examples build without sourcing envsetup.sh
2014-04-15 08:35:00 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Fix the Android compilation (better structure for NetEq test libs)
2014-04-24 13:19:04 +00:00
webrtc.gyp
Integrate fake_network_pipe into direct_transport.
2013-12-18 20:28:25 +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.