Karl Wiberg 88de4792d0 AudioEncoderIsac: Print error code if CHECK for successful encoding fails ...

This will hopefully make the crash in bug 4577 easier to understand if
it happens again.

BUG=4577
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52389004

Cr-Commit-Position: refs/heads/master@{#9100}

2015-04-28 13:43:43 +00:00

..

audio_coding

AudioEncoderIsac: Print error code if CHECK for successful encoding fails

2015-04-28 13:43:43 +00:00

audio_conference_mixer

Remove henrike@ from OWNERS

2015-04-01 15:08:49 +00:00

audio_device

Removes use of AudioManager.setSpeakerphoneOn in audio manager

2015-04-14 07:19:49 +00:00

audio_processing

audio_processing: Fixed incorrect usage of SetExtraOptions() in offline tool

2015-04-28 11:52:30 +00:00

bitrate_controller

Moving the pacer and the pacer thread to ChannelGroup.

2015-03-26 10:11:22 +00:00

desktop_capture

Do not use Magnifier if there are multiple screens since it sometimes crashes.

2015-04-24 15:46:15 +00:00

interface

Record H264 NALU type in the h264 header.

2015-04-23 18:14:46 +00:00

media_file

Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro

2015-03-04 13:04:54 +00:00

pacing

Moving the pacer and the pacer thread to ChannelGroup.

2015-03-26 10:11:22 +00:00

remote_bitrate_estimator

Improve TCP implementation by adding ssthresh and make it possible to start it with an offset.

2015-04-28 12:38:31 +00:00

rtp_rtcp

Clean up RTCPSender.

2015-04-27 11:32:31 +00:00

utility

Cleanup: Remove i420_video_frame.h header.

2015-04-10 10:52:15 +00:00

video_capture

GYP variables for building expat, icu, libsrtp, usrsctp

2015-04-27 12:38:37 +00:00

video_coding

Set correct encoder-specific settings for vpx in the new API.

2015-04-28 08:01:14 +00:00

video_processing

Cleanup: Remove i420_video_frame.h header.

2015-04-10 10:52:15 +00:00

video_render

Cleanup inside IncomingVideoStream.

2015-04-24 11:54:05 +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

Adds C++/JNI/Java unit test for audio device module on Android.

2015-03-09 12:40:43 +00:00

modules.gyp

Add h.264 AVC SPS parsing for resolution (re-land)

2015-04-24 11:56:44 +00:00

OWNERS

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

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