philipel 83c97da593 Only append SPS/PPS to bitstream if supplied out of band. ...

BUG=chromium:721597

Review-Url: https://codereview.webrtc.org/2945853002
Cr-Commit-Position: refs/heads/master@{#18701}

2017-06-21 14:22:40 +00:00

..

audio_coding

Improve the simulation stats aggregation in neteq_rtpplay

2017-06-20 16:20:00 +00:00

audio_conference_mixer

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

audio_device

Reduces sensitivity in audio-glitch detector for iOS

2017-06-16 12:09:10 +00:00

audio_mixer

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

audio_processing

Use constexpr to avoid a static initializer

2017-06-20 17:57:09 +00:00

bitrate_controller

Tune loss-based BWE to be more compatible with the low frequency loss reports of audio streams.

2017-06-16 14:47:00 +00:00

congestion_controller

Test and fix for huge bwe drop after alr state.

2017-06-21 06:42:30 +00:00

desktop_capture

Add reference counter of DxgiDuplicatorController to unload DXGI components

2017-06-19 20:59:42 +00:00

include

Implement timing frames.

2017-06-19 14:18:55 +00:00

media_file

Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ )

2017-04-26 10:38:35 +00:00

pacing

Use RaceChecker instead of ThreadChecker in a few places.

2017-06-16 12:02:05 +00:00

remote_bitrate_estimator

Test and fix for huge bwe drop after alr state.

2017-06-21 06:42:30 +00:00

rtp_rtcp

Only append SPS/PPS to bitstream if supplied out of band.

2017-06-21 14:22:40 +00:00

utility

Remove deprecation warning from JVM::Initialize with the context parameter.

2017-06-01 11:59:52 +00:00

video_capture

Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface

2017-06-14 13:46:38 +00:00

video_coding

Only append SPS/PPS to bitstream if supplied out of band.

2017-06-21 14:22:40 +00:00

video_processing

Update webrtc/media and webrtc/modules to new VideoFrameBuffer interface

2017-06-14 13:46:38 +00:00

BUILD.gn

This is an initial cl, which contains small amount of implemented functions, and large amount of unimplemented ones.

2017-06-05 13:01:26 +00:00

module_common_types_unittest.cc

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

OWNERS

Remove all references to GYP

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