minyue d01ed1fe8f Fix an error in Audio Network Adaptor: time constant passed wrong. ...

BUG=wbrtc:6303

Review-Url: https://codereview.webrtc.org/2595283003
Cr-Commit-Position: refs/heads/master@{#15767}

2016-12-23 09:49:37 +00:00

..

audio_coding

Fix an error in Audio Network Adaptor: time constant passed wrong.

2016-12-23 09:49:37 +00:00

audio_conference_mixer

Refactor webrtc/modules/audio_conference_mixer for GN check

2016-12-22 14:56:21 +00:00

audio_device

Disables AudioDeviceTest.StartStopPlayout on iOS

2016-12-22 09:36:49 +00:00

audio_mixer

Delete voice_engine_configurations.h

2016-12-12 13:03:08 +00:00

audio_processing

To verify the upcoming code changes it is required

2016-12-20 21:45:58 +00:00

bitrate_controller

Add OWNERS to BWE modules.

2016-12-12 15:53:39 +00:00

congestion_controller

Revert "Revert of Parse FlexFEC RTP headers in Call and add integration with BWE. (patchset #17 id:460001 of https://codereview.webrtc.org/2553863003/ )"

2016-12-21 14:37:18 +00:00

desktop_capture

Revert of Refactor webrtc/modules/desktop_capture for GN check (patchset #1 id:1 of https://codereview.webrtc.org/2593713002/ )

2016-12-22 07:19:59 +00:00

include

Revert of Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame. (patchset #1 id:1 of https://codereview.webrtc.org/2574943003/ )

2016-12-19 17:36:33 +00:00

media_file

Refactor webrtc/modules/media_file for GN check

2016-12-21 08:20:52 +00:00

pacing

Refactor webrtc/modules/pacing for GN check

2016-12-21 08:26:58 +00:00

remote_bitrate_estimator

Remove redundant local variable from OveruseDetector::Detect.

2016-12-16 14:25:53 +00:00

rtp_rtcp

Make class of static functions in rtp_to_ntp.h:

2016-12-22 15:53:51 +00:00

utility

Delete voice_engine_configurations.h

2016-12-12 13:03:08 +00:00

video_capture

Replace VideoCaptureDataCallback by VideoSinkInterface.

2016-12-12 08:23:05 +00:00

video_coding

Do not call OnDecoderTiming before timing values are set.

2016-12-22 09:26:18 +00:00

video_processing

Revert of Refactor webrtc/modules/video_processing for GN check (patchset #3 id:40001 of https://codereview.webrtc.org/2595543002/ )

2016-12-21 10:18:54 +00:00

BUILD.gn

Refactor webrtc/modules/rtp_rtcp for GN check

2016-12-22 15:35:39 +00:00

module_common_types_unittest.cc

Move functionality out from AudioFrame and into AudioFrameOperations.

2016-12-05 09:46:20 +00:00

OWNERS

Remove all references to GYP

2016-11-16 19:11:38 +00:00