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henrik.lundin@webrtc.org 8315d7de85 Remove dual stream functionality in VoiceEngine 
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.

BUG=3520
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 16:07:26 +00:00
..
base
Remove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 10:51:54 +00:00
build
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
common_audio
Remove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 10:51:54 +00:00
common_video
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
examples
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
libjingle
Added support for an Origin header in STUN messages.
2015-01-10 00:47:02 +00:00
modules
Remove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 10:51:54 +00:00
overrides
Remove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 10:51:54 +00:00
p2p
Add support for trying alternate server (STUN 300 error message) on TCP
2015-01-10 02:41:32 +00:00
sound
rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.
2014-10-01 16:33:03 +00:00
system_wrappers
Remove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 10:51:54 +00:00
test
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
tools
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
video
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
video_engine
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
voice_engine
Remove dual stream functionality in VoiceEngine
2015-01-14 16:07:26 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Enabling building with NEON on ARM64
2014-11-26 17:01:40 +00:00
call.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.h
Refactor some receive-side stats.
2014-12-18 13:50:16 +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
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
config.h
Revert 8028 "Support associated payload type when registering Rt..."
2015-01-09 20:22:46 +00:00
engine_configurations.h
Add VP9 codec to VCM and vie_auto_test.
2014-11-01 06:10:48 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +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
GN: Add BUILD.gn files + kjellander to OWNERS
2014-06-23 19:21:07 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
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
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 289723:291647
2014-08-25 14:16:32 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Annotate COMPILE_ASSERT with __attribute__((unused)).
2014-11-17 13:47:38 +00:00
video_decoder.h
Add support for VP9 in webrtc::Call and video_loopback.
2014-11-04 19:41:15 +00:00
video_encoder.h
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Report encoded frame size in VideoSendStream.
2014-12-01 15:23:21 +00:00
video_receive_stream.h
Wire up bandwidth stats to the new API and webrtcvideoengine2.
2014-11-05 14:05:29 +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 payload-based padding by default and remove the API.
2014-12-19 15:33:17 +00:00
webrtc_examples.gyp
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
webrtc_perf_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
webrtc_tests.gypi
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
webrtc.gyp
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +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.