stefan@webrtc.org 0b38478885 Add support for parsing header only RTP dumps with bwe_rtp_play. ...

Also adds support for printing the original_length in rtp_to_text.

R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7812 4adac7df-926f-26a2-2b94-8c16560cd09d

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

..

audio_coding

Relanding r7807.

2014-12-04 12:14:12 +00:00

audio_conference_mixer

GN: Add common configs to all targets.

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

audio_device

Update isolate files for Android APK tests.

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

audio_processing

Remove need for assembly offset generation in aecm and ns module.

2014-12-03 17:54:38 +00:00

bitrate_controller

Put send-side bwe probing under finch experiment.

2014-11-10 13:55:16 +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

Set simulcastIdx field to zero even if it has no meaning.

2014-12-01 15:29:29 +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

Adjust parameter in videoprocessor_integration_test for vp9.

2014-12-03 02:00:12 +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

Merge remote bitrate estimator changes.

2014-12-04 15:34:06 +00:00

OWNERS

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

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