kwiberg@webrtc.org b3ad8cf6ca Make an AudioEncoder subclass for iSAC ...

BUG=3926

Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675
and reverted: https://code.google.com/p/webrtc/source/detail?r=7676

R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-12-11 10:08:19 +00:00

..

audio_coding

Make an AudioEncoder subclass for iSAC

2014-12-11 10:08:19 +00:00

audio_conference_mixer

GN: Add common configs to all targets.

2014-09-28 17:37:22 +00:00

audio_device

Attempt to fix FYI bots.

2014-12-09 19:59:27 +00:00

audio_processing

Support 48kHz in Noise Suppression

2014-12-10 19:30:57 +00:00

bitrate_controller

Change all system clock types to int64_t in bitrate_controller.

2014-12-08 19:46:23 +00:00

desktop_capture

Fix a platform check to use WEBRTC_WIN instead of OS_WIN.

2014-11-12 20:53:00 +00:00

interface

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

media_file

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

pacing

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

remote_bitrate_estimator

Add support for parsing header only RTP dumps with bwe_rtp_play.

2014-12-04 15:43:49 +00:00

rtp_rtcp

Add video send bitrates to histogram stats:

2014-12-09 09:47:53 +00:00

utility

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

video_capture

OWNERS: Remove tomasl@ and mallinath@

2014-11-24 10:05:05 +00:00

video_coding

Remove jitter_estimate_test.h

2014-12-10 21:08:39 +00:00

video_processing

Use size_t more consistently for packet/payload lengths.

2014-11-20 22:28:14 +00:00

video_render

OWNERS: Remove tomasl@ and mallinath@

2014-11-24 10:05:05 +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

Update isolate files for Android APK tests.

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

modules.gyp

Make an AudioEncoder subclass for comfort noise

2014-12-10 07:29:08 +00:00

OWNERS

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

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