webrtc_m130

History

stefan@webrtc.org de98478965 Update the remote bitrate estimator before passing the packet to the RTP module.

This solves the problem of reconstructed packets biasing the bandwidth estimate.

TEST=vie_auto_test --automated, trybots
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1594005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4171 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-06-04 12:15:40 +00:00

build

Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.

2013-05-29 00:38:02 +00:00

common_audio

Replace the old resampler with SincResampler in the voice engine signal path.

2013-06-03 19:00:29 +00:00

common_video

Include files from webrtc/.. paths in common_video/

2013-06-04 09:02:37 +00:00

modules

Update the remote bitrate estimator before passing the packet to the RTP module.

2013-06-04 12:15:40 +00:00

system_wrappers

Replace the old resampler with SincResampler in the voice engine signal path.

2013-06-03 19:00:29 +00:00

test

Remove #pragma once

2013-05-29 13:19:09 +00:00

tools

Include files from webrtc/.. paths in tools/

2013-06-04 08:14:10 +00:00

video_engine

Update the remote bitrate estimator before passing the packet to the RTP module.

2013-06-04 12:15:40 +00:00

voice_engine

Fix error in mixing test for supported sample rates.

2013-06-03 22:52:09 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

common_types.h

Replace ExtraCodecOptions with new Config class that supports multiple settings at once.

2013-05-14 08:02:25 +00:00

common_unittest.cc

Added a config class to ease passing a set of options across webrtc.

2013-05-13 08:06:36 +00:00

common.h

Added a config class to ease passing a set of options across webrtc.

2013-05-13 08:06:36 +00:00

engine_configurations.h

Remove WEBRTC_*_ENGINE_NETWORK_API use

2013-04-04 19:51:42 +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

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

typedefs.h

Disable clang C++11 warnings to permit OVERRIDE keyword.

2013-05-08 08:34:34 +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.