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pbos@webrtc.org caba2d2a37 Add DeliveryStatus enum to DeliverPacket(). 
Allows signalling why packet delivery failed. Especially enables
signaling that delivery fails because the incoming packet had an unknown
SSRC. This allows an application to react and create receivers for the
new streams.

R=mflodman@webrtc.org
BUG=3228

Review URL: https://webrtc-codereview.appspot.com/12289005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 13:57:12 +00:00
..
base
Adds a modified copy of talk/base to webrtc/base. It is the first step in
2014-05-13 18:00:26 +00:00
build
Adds a modified copy of talk/base to webrtc/base. It is the first step in
2014-05-13 18:00:26 +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
WebRTCDemo: correct set trace filter operation.
2014-05-05 04:24:47 +00:00
modules
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
overrides/webrtc/base
Adds a modified copy of talk/base to webrtc/base. It is the first step in
2014-05-13 18:00:26 +00:00
system_wrappers
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
test
Add DeliveryStatus enum to DeliverPacket().
2014-05-14 13:57:12 +00:00
tools
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
video
Add DeliveryStatus enum to DeliverPacket().
2014-05-14 13:57:12 +00:00
video_engine
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
voice_engine
Add webrtc field trials API.
2014-05-14 12:24:04 +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
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 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
Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.
2013-07-22 22:32:50 +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
Rename Start/Stop in Video{Send,Receive}Streams.
2014-04-24 11:13:21 +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
Adds a modified copy of talk/base to webrtc/base. It is the first step in
2014-05-13 18:00:26 +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.