andresp@webrtc.org 817a034cf2 Fix TimeToSendPadding return to be 0 if no padding bytes are sent. ...

BUG=3694
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15149005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6900 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-08-14 08:24:47 +00:00

..

audio_coding

Removing TODOs related to AcmReceiverBitExactness checksums

2014-08-13 13:02:00 +00:00

audio_conference_mixer

This is related to an earlier CL of enabling Opus 48 kHz.

2014-07-18 12:28:28 +00:00

audio_device

Fixing uninitialized variable in file_audio_device.cc.

2014-08-12 11:09:12 +00:00

audio_processing

modules/audio_processing: Moves declaration of kDelayDiffOffsetSamples

2014-08-11 15:39:00 +00:00

bitrate_controller

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

desktop_capture

Fix an invalid memory access due to typo in win/cursor.cc.

2014-07-15 20:32:03 +00:00

interface

Add H.264 packetization.

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

media_file

Add unit test for MediaFile WAV file writing

2014-07-17 08:11:32 +00:00

pacing

Improvements to the pacer where it lost some budget due to truncation errors.

2014-07-15 16:40:38 +00:00

remote_bitrate_estimator

RTP video playback tool using Call APIs.

2014-08-06 16:26:56 +00:00

rtp_rtcp

Fix TimeToSendPadding return to be 0 if no padding bytes are sent.

2014-08-14 08:24:47 +00:00

utility

Some refactoring inside rtp_rtcp/.

2014-07-08 12:10:51 +00:00

video_capture

Runtime guard for iOS7 property.

2014-07-18 17:17:59 +00:00

video_coding

Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798

2014-08-11 21:06:30 +00:00

video_processing

Remove the VPM denoiser.

2014-07-15 09:50:40 +00:00

video_render

Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798

2014-08-11 21:06:30 +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

Since the packet loss rate cannot be estimated accurately, there is always a mismatch between the estimated packet loss rate and the true one. Such a mismatch will make Opus FEC suboptimal.

2014-08-07 12:31:36 +00:00

OWNERS

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

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