bjornv@webrtc.org f71785cd3b audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >> ...

Replaced trivial shift macro with >>. The actual implementation of the macro is simply >>.

Affected codecs:
* ilbc
* isac/fix

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7396 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-10-08 15:36:30 +00:00

..

audio_coding

audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>

2014-10-08 15:36:30 +00:00

audio_conference_mixer

GN: Add common configs to all targets.

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

audio_device

Roll chromium_revision c264a05..fc668e2 (297113:298195)

2014-10-07 12:49:34 +00:00

audio_processing

audio_processing: Replaced WEBRTC_SPL_RSHIFT_U32 with >>

2014-10-02 08:07:05 +00:00

bitrate_controller

GN: Add common configs to all targets.

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

desktop_capture

Remove mouse cursor capturer from the ScreenCapturer interface

2014-10-02 01:47:10 +00:00

interface

Add H.264 packetization.

2014-07-31 14:59:24 +00:00

media_file

GN: Add common configs to all targets.

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

pacing

GN: Add common configs to all targets.

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

remote_bitrate_estimator

GN: Enable libvpx, add link target and convert some test targets

2014-09-30 18:05:02 +00:00

rtp_rtcp

Remove callback from RtpDepacketizer::Parse().

2014-09-29 08:00:22 +00:00

utility

GN: Add common configs to all targets.

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

video_capture

GN: Add common configs to all targets.

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

video_coding

GN: Enable libvpx, add link target and convert some test targets

2014-09-30 18:05:02 +00:00

video_processing

GN: Add common configs to all targets.

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

video_render

GN: Add common configs to all targets.

2014-09-28 17:37:22 +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

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules_unittests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules.gyp

Reduce jitter delay for low fps streams.

2014-09-24 14:06:56 +00:00

OWNERS

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

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