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stefan@webrtc.org 4ef438e2de Remove the send-side cname getter APIs from voice and video engine. 
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 09:55:30 +00:00
..
base
Rebase webrtc/base with r6655 version of talk/base:
2014-07-10 22:47:02 +00:00
build
Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams.
2014-07-10 10:35:12 +00:00
common_audio
common_audio: Removes macro WEBRTC_SPL_SHIFT_W16
2014-07-03 13:38:53 +00:00
common_video
GN: Implement BUILD.gn for common_video.
2014-07-03 17:04:12 +00:00
examples
WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering.
2014-07-03 05:59:22 +00:00
modules
Remove the send-side cname getter APIs from voice and video engine.
2014-07-11 09:55:30 +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
Clean data races from system_wrappers_unittests.
2014-07-07 20:37:39 +00:00
test
Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams.
2014-07-10 10:35:12 +00:00
tools
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video
Remove the send-side cname getter APIs from voice and video engine.
2014-07-11 09:55:30 +00:00
video_engine
Remove the send-side cname getter APIs from voice and video engine.
2014-07-11 09:55:30 +00:00
voice_engine
Remove the send-side cname getter APIs from voice and video engine.
2014-07-11 09:55:30 +00:00
.gitignore
…
BUILD.gn
GN: Refactor base/BUILD.gn and fix dbus-glib error.
2014-06-28 18:05:22 +00:00
call.h
Remove GetDefaultConfigs() from Call.
2014-07-07 04:45:15 +00:00
common_types.h
Count total bytes sent in RTPSender::Bytes().
2014-07-10 16:24:54 +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
Remove the send-side cname getter APIs from voice and video engine.
2014-07-11 09:55:30 +00:00
engine_configurations.h
Add boilerplate code for H.264.
2014-07-04 12:42:07 +00:00
experiments.h
Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable.
2014-07-08 13:59:46 +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
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +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
TSan: Move suppressions to source file.
2014-06-27 09:18:51 +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
Remove GetDefaultConfigs() from Call.
2014-07-07 04:45:15 +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 the send-side cname getter APIs from voice and video engine.
2014-07-11 09:55:30 +00:00
webrtc_examples.gyp
WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering.
2014-07-03 05:59:22 +00:00
webrtc_perf_tests.isolate
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
webrtc_tests.gypi
Refactor ramp-up tests to have separate help files for the test classes, to make things more reusable.
2014-07-08 13:59:46 +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.