henrik.lundin@webrtc.org fddeaf5daa Switch to using AudioEncoderG722 instead of ACMG722 ...

This change switches from the old codec wrapper ACMG722 to the new
AudioEncodeG722 wrapped in an ACMGenericCodecWrapper.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39879004

Cr-Commit-Position: refs/heads/master@{#8330}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8330 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-02-11 13:28:44 +00:00

..

audio_coding

Switch to using AudioEncoderG722 instead of ACMG722

2015-02-11 13:28:44 +00:00

audio_conference_mixer

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

audio_device

Refactoring WebRTC Java/JNI audio recording in C++ and Java.

2015-02-11 08:39:19 +00:00

audio_processing

Fix bug when there are no blocks in a chunk in Beamformer

2015-02-11 00:48:55 +00:00

bitrate_controller

Enable bitrate probing by default.

2015-02-03 12:34:17 +00:00

desktop_capture

GN: Prepare to remove webrtc_base target

2015-01-21 20:22:33 +00:00

interface

Revamp the ProcessThreadImpl implementation.

2015-02-06 09:44:45 +00:00

media_file

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

pacing

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

remote_bitrate_estimator

Introduce PacketReceiver and remove configuration of simulations via the BweTestConfig.

2015-02-10 09:49:00 +00:00

rtp_rtcp

Implementing a packet router class, used to route RTP packets to the

2015-02-06 13:10:39 +00:00

utility

Refactoring WebRTC Java/JNI audio recording in C++ and Java.

2015-02-11 08:39:19 +00:00

video_capture

Rename GYP and GN targets for video capture+render.

2015-02-11 07:47:47 +00:00

video_coding

Restructure GYP for vp9, opus and direct trace

2015-02-09 10:01:40 +00:00

video_processing

Remove ChangeUniqueID.

2015-01-29 12:14:13 +00:00

video_render

Rename GYP and GN targets for video capture+render.

2015-02-11 07:47:47 +00:00

module_common_types_unittest.cc

Fix issues with incorrect wrap checks when having big buffers and high bitrate.

2013-04-11 17:48:02 +00:00

modules_java_chromium.gyp

Fixes build breakage when building WebRTC in Chromium and having include_tests=1.

2013-07-08 18:14:58 +00:00

modules_java.gyp

Build all java files into jar for each module on Android

2013-07-01 17:52:39 +00:00

modules_tests.isolate

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

modules_unittests.isolate

Removes unused test files by audio_processing/transient

2014-12-15 16:13:05 +00:00

modules.gyp

Rename GYP and GN targets for video capture+render.

2015-02-11 07:47:47 +00:00

OWNERS

Add kjellander@webrtc.org as OWNER for *.isolate

2014-06-10 05:42:53 +00:00