generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
aa5ea1c0f9
webrtc/webrtc
History
minyue@webrtc.org aa5ea1c0f9 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED 
2. Add two new APIs to configure codec internal FEC

3. Add a test and listened to results. This is based modifying EncodeDecodeTest and deriving a new class from it.

New ACM gives good result.
Old ACM does not use NetEq 4, so FEC won't be decoded.

BUG=
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6233 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 15:16:51 +00:00
..
base Rebase webrtc/base 6163:6216 (svn diff -r 6163:6216 http://webrtc.googlecode.com/svn/trunk/talk/base, apply diff manually) 2014-05-21 20:42:17 +00:00
build removed webrtc_base_tests_utils from merge libs as it was breaking some builds. 2014-05-15 21:45:09 +00:00
common_audio Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 2014-05-21 21:18:46 +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 WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker 2014-05-21 03:37:45 +00:00
modules 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED 2014-05-23 15:16:51 +00:00
overrides/webrtc/base Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h. 2014-05-21 16:52:14 +00:00
system_wrappers Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 2014-05-21 21:18:46 +00:00
test Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 2014-05-21 21:18:46 +00:00
tools Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. 2014-05-21 21:18:46 +00:00
video Revert "Revert "Remove VideoSendStreamInput::PutFrame."" 2014-05-23 13:03:45 +00:00
video_engine Revert "Revert "Remove VideoSendStreamInput::PutFrame."" 2014-05-23 13:03:45 +00:00
voice_engine 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED 2014-05-23 15:16:51 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
call.h Add DeliveryStatus enum to DeliverPacket(). 2014-05-14 13:57:12 +00:00
common_types.h Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
common.gyp Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
common.h Add a Config class interface to AudioProcessing for passing options. 2013-07-25 18:28:29 +00:00
config.cc Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
config.h Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
engine_configurations.h Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 2014-05-12 12:19:19 +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 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 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. 2014-04-14 20:08:03 +00:00
PATENTS Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
PRESUBMIT.py Made the presubmit script accept license headers back to 2003 2014-05-15 18:21:17 +00:00
README.chromium Move src/ -> webrtc/ 2012-10-22 18:19:23 +00:00
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 Remove ALLOW_UNUSED. 2014-05-05 18:18:02 +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 Revert "Revert "Remove VideoSendStreamInput::PutFrame."" 2014-05-23 13:03:45 +00:00
webrtc_examples.gyp Add webrtc field trials API. 2014-05-14 12:24:04 +00:00
webrtc_perf_tests.isolate Move realtime tests to webrtc_perf_tests. 2013-12-13 12:48:05 +00:00
webrtc_tests.gypi Add webrtc field trials API. 2014-05-14 12:24:04 +00:00
webrtc.gyp Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +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.