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kwiberg@webrtc.org be96bfb179 Re-land "Switch to using AudioEncoderIsac instead of ACMISAC" 
It should work now, after the fix in r8431.

Previously committed in r8342, reverted in r8372, committed in r8378,
and reverted in r8412.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4228
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34279004

Cr-Commit-Position: refs/heads/master@{#8433}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8433 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 15:10:49 +00:00
..
base
Fix WebRTC IP leaks.
2015-02-18 20:27:45 +00:00
build
Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737)
2015-02-18 10:38:11 +00:00
common_audio
WebRtc_GetCPUFeaturesARM is only available on android
2015-02-11 17:03:24 +00:00
common_video
Fix I420VideoFrame unittests
2015-02-18 15:54:10 +00:00
examples
This is a code clean up. No functional change intended.
2015-02-13 14:32:13 +00:00
libjingle
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
modules
Re-land "Switch to using AudioEncoderIsac instead of ACMISAC"
2015-02-19 15:10:49 +00:00
overrides
Add arraysize to overrides to avoid macros redefinitions in Chromium
2015-02-14 02:46:13 +00:00
p2p
Fix WebRTC IP leaks.
2015-02-18 19:10:22 +00:00
sound
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
system_wrappers
GN: Changes for vp9, opus and direct trace
2015-02-16 12:47:45 +00:00
test
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
tools
Re-land "Remove <(webrtc_root) from source file entries."
2015-01-29 14:30:41 +00:00
video
Add RtcpPacketTypeCounter stats to new API.
2015-02-19 12:47:45 +00:00
video_engine
Add RtcpPacketTypeCounter stats to new API.
2015-02-19 12:47:45 +00:00
voice_engine
Adding two new stats to VoiceReceiverInfo
2015-02-18 15:24:39 +00:00
.gitignore
…
BUILD.gn
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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
Add RtcpPacketTypeCounter stats to new API.
2015-02-19 12:47:45 +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
Add RtcpPacketTypeCounter stats to new API.
2015-02-19 12:47:45 +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
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +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
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737)
2015-02-18 10:38:11 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Enable Clang warning implicit-fallthrough and annotate the code.
2015-01-28 18:38:13 +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
Remove default arguments in EncodedImageCallback.
2015-02-09 09:14:48 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
CVO capturer feature: allow unrotated frame flows through the capture pipeline.
2015-02-11 18:38:53 +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
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +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
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.