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henrik.lundin@webrtc.org 4aecd008dd Add support for 40 and 60 ms frames to AudioEncoderIlbc 
BUG=3926
COAUTHOR:kwiberg@webrtc.org

R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37789004

Cr-Commit-Position: refs/heads/master@{#8182}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8182 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 13:16:44 +00:00
..
base
GN: Remove webrtc_base target in favor for rtc_base.
2015-01-28 07:55:45 +00:00
build
Roll chromium_revision c086b4e..4664fe0 (312108:312733)
2015-01-25 19:17:56 +00:00
common_audio
Use non-zero data in AudioRingBufferTest.
2015-01-27 21:59:44 +00:00
common_video
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
examples
Use int64_t more consistently for times, in particular for RTT values.
2015-01-12 21:51:21 +00:00
libjingle
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
modules
Add support for 40 and 60 ms frames to AudioEncoderIlbc
2015-01-28 13:16:44 +00:00
overrides
Remove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 10:51:54 +00:00
p2p
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +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
Change a GYP reference to cpufeatures.gypi
2015-01-23 16:35:17 +00:00
test
Move system_wrappers.gyp files to the proper directory.
2015-01-14 09:30:52 +00:00
tools
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +00:00
video
Make sure ByteReader<T>::Read* is properly constified.
2015-01-28 12:38:16 +00:00
video_engine
Update StreamDataCounter with FEC bytes.
2015-01-27 12:17:29 +00:00
voice_engine
Fix bug in GetREDStatus(): it doesn't actually return the current status.
2015-01-26 22:35:29 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
BUILD.gn
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +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
Update StreamDataCounter with FEC bytes.
2015-01-27 12:17:29 +00:00
common.gyp
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +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
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +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
Log configs when creating video streams in Call.
2015-01-15 10:09:39 +00:00
webrtc_examples.gyp
Move internal capture+render to build_with_chromium==0 condition
2015-01-20 11:40:45 +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
Make it easier to use external libyuv + cleanup GYP files.
2015-01-26 19:17:26 +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.