andresp@webrtc.org b08a12d6e8 Isolate debug recording from video sender into a thread safe small class. ...

R=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5353 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-01-08 12:38:22 +00:00

..

audio_coding

Remove the requirement to call set_sample_rate_hz and friends.

2014-01-07 17:45:09 +00:00

audio_conference_mixer

Remove the requirement to call set_sample_rate_hz and friends.

2014-01-07 17:45:09 +00:00

audio_device

Android build: make it quiet on success and not overly noisy on failure.

2013-12-20 22:49:35 +00:00

audio_processing

Delay Estimator: Converts a constant into a configurable parameter.

2014-01-08 08:18:15 +00:00

bitrate_controller

Change the low-bitrate handling in BitrateControllerImpl

2013-10-24 09:24:06 +00:00

desktop_capture

Make MouseCursor mutable

2013-12-18 02:18:01 +00:00

interface

Revert r5294 to re-roll r5293.

2013-12-16 12:24:44 +00:00

media_file

Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0.

2013-12-09 16:23:00 +00:00

pacing

Add thread_annotations for clang targets.

2013-12-20 20:20:50 +00:00

remote_bitrate_estimator

Revert r5294 to re-roll r5293.

2013-12-16 12:24:44 +00:00

rtp_rtcp

Add include guards to forward_error_correction_internal.h

2013-12-24 05:06:12 +00:00

utility

Ensure WEBRTC_MODULE_UTILITY_VIDEO is undefined for enable_video==0.

2013-12-09 16:23:00 +00:00

video_capture

Roll chromium_revision 232627:238260

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

video_coding

Isolate debug recording from video sender into a thread safe small class.

2014-01-08 12:38:22 +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

Modify video_render/ to allow a single old frame.

2013-12-16 18:24:37 +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

Roll chromium_revision 232627:238260

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

modules.gyp

Remove the requirement to call set_sample_rate_hz and friends.

2014-01-07 17:45:09 +00:00