henrik.lundin@webrtc.org f9471807a2 Add Opus support to neteq_rtpplay ...

BUG=2416
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7595 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-11-03 15:19:58 +00:00

..

audio_coding

Add Opus support to neteq_rtpplay

2014-11-03 15:19:58 +00:00

audio_conference_mixer

GN: Add common configs to all targets.

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

audio_device

Add more sanity checks to workaround the unidentified problem that CaptureThread is still running while related resouces are destroyed already.

2014-11-03 09:58:40 +00:00

audio_processing

replace inline assembly WebRtcNsx_SynthesisUpdateNeon by intrinsics.

2014-10-31 22:07:35 +00:00

bitrate_controller

Add UMA metrics for the initial (after two seconds) packet loss, round-trip time and bandwidth estimate of a WebRTC call.

2014-11-03 14:42:43 +00:00

desktop_capture

Remove mouse cursor capturer from the ScreenCapturer interface

2014-10-02 01:47:10 +00:00

interface

Add H.264 packetization.

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

media_file

GN: Add common configs to all targets.

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

pacing

Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate.

2014-10-23 11:57:05 +00:00

remote_bitrate_estimator

Simplify bwe tests.

2014-10-31 10:47:12 +00:00

rtp_rtcp

Add stats for duplicate sent and received NACK requests.

2014-10-29 12:42:30 +00:00

utility

GN: Add common configs to all targets.

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

video_capture

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

video_coding

Add stats for video:

2014-11-03 14:40:38 +00:00

video_processing

GN: Add common configs to all targets.

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

video_render

Update all .isolate files for the new format.

2014-10-31 18:08:09 +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 all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

modules_unittests.isolate

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

modules.gyp

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +00:00

OWNERS

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

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