webrtc_m130

History

Bjorn Mellem 6eb03b81bb Remove dependency on gunit headers in virtualsocketserver.

BUG=7810

Change-Id: I66d9aeaca2dd81c20f78052a15ea3680e23a1501
Reviewed-on: https://chromium-review.googlesource.com/534354
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18579}

2017-06-14 00:13:53 +00:00

api

Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP.

2017-06-02 21:37:37 +00:00

audio

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

base

Remove dependency on gunit headers in virtualsocketserver.

2017-06-14 00:13:53 +00:00

call

Log an error in RtpDemuxer::FindSsrcAssociations() if kMaxProcessedSsrcs exceeded

2017-06-13 14:57:31 +00:00

common_audio

Moving compile_assert_c.h to webrtc/base

2017-05-15 15:06:47 +00:00

common_video

Removing backward compatible header

2017-06-13 11:25:37 +00:00

examples

Make sure UI methods get called on the main thread

2017-06-13 12:25:33 +00:00

logging

Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format

2017-06-13 09:08:14 +00:00

media

Remove webrtcvideoengine2.h

2017-06-13 14:28:31 +00:00

modules

Tune vp9 quality scaler parameters

2017-06-13 18:34:49 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

2017-05-17 01:00:06 +00:00

p2p

Adding PortAllocator option to support cases where sockets can't be bound.

2017-06-13 22:49:45 +00:00

Remove DCHECK from PeerConnectionFactory::worker_thread.

2017-06-13 22:01:49 +00:00

sdk

Implement org.webrtc.VideoEncoder using the android MediaCodec.

2017-06-13 16:07:29 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Prevent data race in GetStaticInstance

2017-05-29 07:01:14 +00:00

test

Use rtp_header_extension_map.h instead of rtp_header_extension.h

2017-06-12 14:01:20 +00:00

tools

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

video

Only create H264 frames if there are no gaps in the packet sequence number.

2017-06-13 09:47:28 +00:00

voice_engine

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

.gitignore

…

BUILD.gn

Delete old include file webrtc/video_frame.h.

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

codereview.settings

Give a more specific URL for creating WebRTC checkout

2016-04-01 07:36:53 +00:00

common_types.cc

Add RSID-based demuxing to RtpDemuxer

2017-06-08 11:19:13 +00:00

common_types.h

Add RSID-based demuxing to RtpDemuxer

2017-06-08 11:19:13 +00:00

config.cc

Fix default extension id for video content type extension

2017-04-26 09:35:31 +00:00

config.h

Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )

2017-04-11 17:34:31 +00:00

DEPS

Delete old include file webrtc/video_frame.h.

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

LICENSE

…

LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

OWNERS

Don't add or rename files in webrtc/ and webrtc/api/ without a proper review

2017-05-08 13:01:54 +00:00

PATENTS

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PRESUBMIT.py

…

README.chromium

…

typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_receive_stream.h

Delete VieRemb class, move functionality to PacketRouter.

2017-04-19 06:38:35 +00:00

video_send_stream.h

Update adaptation stats to support degradations in both resolution and framerate.

2017-05-16 06:40:18 +00:00

webrtc.gni

Adds detection of audio glitches for playout on iOS (reland)

2017-06-13 16:00:18 +00:00

whitespace.txt

Moving whitespace file up by one folder

2017-01-24 13:19:42 +00:00

README.chromium

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.