wu@webrtc.org ebdb0e3ad0 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. ...

- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.

This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.

BUG=
TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-03-06 23:49:08 +00:00

..

audio_coding

NetEq4: Changing the behavior of playout_timestamp_ update

2014-03-06 10:28:07 +00:00

audio_conference_mixer

Add Config struct for experimental AGC.

2014-01-29 04:57:25 +00:00

audio_device

ifdef the alsa code based on macro USE_X11

2014-02-20 03:05:05 +00:00

audio_processing

Switch to correct interpretation of int and float input data in audio_processing_unittest

2014-03-05 00:18:53 +00:00

bitrate_controller

Adding a new ramp-up-down-up test

2014-03-06 07:19:28 +00:00

desktop_capture

Call PrintWindow for the first time of capturing to capture the window frames correctly.

2014-02-27 16:43:12 +00:00

interface

Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.

2014-03-06 23:49:08 +00:00

media_file

Removes usage of ListWrapper from several files.

2014-01-13 15:21:30 +00:00

pacing

Adds APIs for reporting pacer queuing delay.

2014-02-27 22:32:40 +00:00

remote_bitrate_estimator

Fixes a bug in the simulation framework where the time offset is accumulating as the packet trace is repeated, causing increasingly large gaps with no packets being transmitted.

2014-03-06 15:46:46 +00:00

rtp_rtcp

Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.

2014-03-06 23:49:08 +00:00

utility

AviRecorder is missing a critical section.

2014-02-24 09:19:36 +00:00

video_capture

VideoCaptureAndroid: don't deliver frames after stopCapture().

2014-03-04 18:17:55 +00:00

video_coding

Remove upper check for number of cores in VCM, I didn't find any good reasons for checking this.

2014-03-05 15:18:45 +00:00

video_processing/main

Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error.

2013-11-25 09:06:33 +00:00

video_render

Removes usage of ListWrapper from several files.

2014-01-13 15:21:30 +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

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

modules_unittests.isolate

Add trace-based delivery filter to BWE test framework.

2014-01-24 10:00:27 +00:00

modules.gyp

Add RTCP packet class.

2014-02-21 08:14:45 +00:00