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aluebs@webrtc.org 59a1b1b928 Fix the different samples per channel in aecdump 
BUG=webrtc:3359
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6996 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 10:43:09 +00:00
..
base
Move end of namespace inside #ifdef
2014-08-27 10:17:22 +00:00
build
Create a copy of talk/sound under webrtc/sound.
2014-08-26 22:04:04 +00:00
common_audio
Refactoring common_audio/signal_processing: Remove macro WEBRTC_SPL_UDIV
2014-08-26 10:25:10 +00:00
common_video
Remove Android.mk build files.
2014-08-26 08:48:51 +00:00
examples
Remove former team members from OWNERS and WATCHLISTS
2014-08-26 06:12:08 +00:00
modules
Fix the different samples per channel in aecdump
2014-08-28 10:43:09 +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
sound
Create a copy of talk/sound under webrtc/sound.
2014-08-26 22:04:04 +00:00
system_wrappers
Remove Android.mk build files.
2014-08-26 08:48:51 +00:00
test
Remove Android.mk build files.
2014-08-26 08:48:51 +00:00
tools
Pass GYP DEPTH variable to isolate.
2014-06-13 09:02:15 +00:00
video
Disable EndToEndTest.RestartingSendStreamPreservesRtpState in video_engine_tests because it is flaky
2014-08-26 14:22:51 +00:00
video_engine
Remove Android.mk build files.
2014-08-26 08:48:51 +00:00
voice_engine
Remove Android.mk build files.
2014-08-26 08:48:51 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
GN: Disable Chromium clang plugins for standalone build.
2014-08-25 14:15:35 +00:00
call.h
Remove GetDefaultConfigs() from Call.
2014-07-07 04:45:15 +00:00
common_types.h
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +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
Check before send/receive rtp header extensions.
2014-07-20 15:27:35 +00:00
engine_configurations.h
Add boilerplate code for H.264.
2014-07-04 12:42:07 +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
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
Fix WEBRTC_AEC_DEBUG_DUMP (broken by int16->float conversion)
2014-08-25 06:26:04 +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
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
2014-07-11 13:44:02 +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
RTP video playback tool using Call APIs.
2014-08-06 16:26:56 +00:00
webrtc.gyp
Create a copy of talk/sound under webrtc/sound.
2014-08-26 22:04:04 +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.