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pbos@webrtc.org f3085e43ab Remove VideoSendStreamInput::PutFrame. 
PutFrame just copies the frame before swapping it, if it's required that
can easily be done outside this API before swapping the frame.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14529006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6229 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-23 09:41:45 +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
Adds missing include of assert header.
2014-05-22 14:31:14 +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
Remove VideoSendStreamInput::PutFrame.
2014-05-23 09:41:45 +00:00
video_engine
Fix deadlock in RegisterPreDecodeImageCallback.
2014-05-23 09:41:07 +00:00
voice_engine
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
2014-05-21 21:18:46 +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
…
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
…
PRESUBMIT.py
Made the presubmit script accept license headers back to 2003
2014-05-15 18:21:17 +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
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
Remove VideoSendStreamInput::PutFrame.
2014-05-23 09:41:45 +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.