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henrik.lundin@webrtc.org 12396aba42 Update PacketSource and RtpFileSource 
The NextPacket method should now return NULL when the end of the
source was reached. In the RtpFileSource, this means that when
the end of file is reached, NULL is returned. Also, when an RTCP
packet is encountered, the next packet will be read from file
immediately.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6479 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 12:20:31 +00:00
..
base
base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/
2014-06-17 14:48:44 +00:00
build
base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/
2014-06-17 14:48:44 +00:00
common_audio
common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16
2014-06-16 10:30:14 +00:00
common_video
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
examples
AppRTCDemo(android): support app (UI) & capture rotation.
2014-06-06 18:40:44 +00:00
modules
Update PacketSource and RtpFileSource
2014-06-18 12:20:31 +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
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
test
Updated W3C getusermedia tests to the latest version of the spec.
2014-06-17 08:46:58 +00:00
tools
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video
Implements start bitrate for new video API.
2014-06-16 08:57:39 +00:00
video_engine
Add max limit of number for overuses. When limit is reached always apply the rampup delay.
2014-06-16 14:27:19 +00:00
voice_engine
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Initial GN work for WebRTC
2014-06-17 08:54:03 +00:00
call.h
Implements start bitrate for new video API.
2014-06-16 08:57:39 +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
…
LICENSE_THIRD_PARTY
Consolidate all third party licenses in LICENSE_THIRD_PARTY.
2013-05-05 18:54:10 +00:00
OWNERS
Add kjellander@webrtc.org as OWNER for *.isolate
2014-06-10 05:42:53 +00:00
PATENTS
…
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
…
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
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +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
Enable pacing by default and remove the option to disable it from the new API.
2014-06-12 15:12:25 +00:00
webrtc_examples.gyp
Add webrtc field trials API.
2014-05-14 12:24:04 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
Rename neteq4 folder to neteq
2014-06-09 08:10:28 +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.