kwiberg@webrtc.org 663fdd02fd Make an AudioEncoder subclass for Opus ...

BUG=3926
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7552 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-10-29 07:28:36 +00:00

..

audio_coding

Make an AudioEncoder subclass for Opus

2014-10-29 07:28:36 +00:00

audio_conference_mixer

GN: Add common configs to all targets.

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

audio_device

Roll chromium_revision c264a05..fc668e2 (297113:298195)

2014-10-07 12:49:34 +00:00

audio_processing

Make NSinst_t* const and rename to self in ns_core

2014-10-28 22:52:09 +00:00

bitrate_controller

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

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

Add macros and APIs for webrtc histograms.

2014-10-23 12:57:56 +00:00

rtp_rtcp

For FIR packet, payload length is zero, so SendToNetwork function is failing.

2014-10-22 09:47:14 +00:00

utility

GN: Add common configs to all targets.

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

video_capture

Query Android device orientation on every camera frame received.

2014-10-17 16:25:06 +00:00

video_coding

Add macros and APIs for webrtc histograms.

2014-10-23 12:57:56 +00:00

video_processing

GN: Add common configs to all targets.

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

video_render

Fix double backslashes in incoming_video_stream.cc

2014-10-28 08:47:16 +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

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules_unittests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

modules.gyp

Move (test) RtpFileReader to a lightweight target.

2014-10-27 18:01:03 +00:00

OWNERS

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

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