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phoglund@webrtc.org 22292df53b Adding explicit check for using dummy file devices. 
Calling into the file device factory without being compiled with file
devices makes no sense and would cause hard-to-debug errors. Therefore
I'm adding an explicit check so this isn't allowed.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 11:39:19 +00:00
..
base
webrtc/base: Fixes miss in base.gyp for windows. See https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle.gyp?r=6503#764 for the corresponding condition.
2014-07-07 17:42:45 +00:00
build
Tight data race suppressions around thread_posix.
2014-07-08 10:36:39 +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
Adding explicit check for using dummy file devices.
2014-07-08 11:39:19 +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
Preserve RTP states for restarted VideoSendStreams.
2014-07-07 13:06:48 +00:00
tools
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video
Extract RTP-header SSRC inline in Call.
2014-07-08 07:38:12 +00:00
video_engine
Preserve RTP states for restarted VideoSendStreams.
2014-07-07 13:06:48 +00:00
voice_engine
Add ExperimentalNs support in Config
2014-06-30 17:39:53 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
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
Add boilerplate code for H.264.
2014-07-04 12:42:07 +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
Add boilerplate code for H.264.
2014-07-04 12:42:07 +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
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
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
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 Call-based tests.
2014-06-27 08:47:52 +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.