webrtc_m130

History

eladalon d0244c21cd Add RSID-based demuxing to RtpDemuxer

Make RtpDemuxer able to demux RTP packets according to RSID (RTP Stream ID), as well as the (pre-existing) ability to demux according to SSRC.

BUG=None

Review-Url: https://codereview.webrtc.org/2920993002
Cr-Commit-Position: refs/heads/master@{#18495}

2017-06-08 11:19:13 +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

New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

2017-06-01 11:02:35 +00:00

base

Deleting unused build target.

2017-06-07 11:50:13 +00:00

call

Add RSID-based demuxing to RtpDemuxer

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

common_audio

Moving compile_assert_c.h to webrtc/base

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

common_video

Update I420Buffer to new VideoFrameBuffer interface

2017-06-01 17:02:26 +00:00

examples

Remove webrtc deps from AppRTCMobile.

2017-06-07 11:13:48 +00:00

logging

Delete rtc_event_log/ringbuffer.h

2017-06-07 08:53:30 +00:00

media

Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ )

2017-06-07 13:17:49 +00:00

modules

Make error resilience configurable through VideoCodecVP9 resilience setting (removes hard coded value in vp9_impl.cc).

2017-06-08 09:55:08 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

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

p2p

Fix issue with send-side bandwidth estimation over TURN TCP connections.

2017-05-27 01:40:05 +00:00

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

sdk

ObjC: Marshal all VideoTrackSource methods to the signaling thread

2017-06-07 14:53:41 +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

Revert of Periodically update codec bit/frame rate settings. (patchset #8 id:140001 of https://codereview.webrtc.org/2883963002/ )

2017-06-07 13:17:49 +00:00

tools

Revert of Refactored incoming bitrate estimator. (patchset #8 id:140001 of https://codereview.webrtc.org/2917873002/ )

2017-06-08 07:10:31 +00:00

video

Add new screenshare full stack test with limited queue.

2017-06-08 08:29:15 +00:00

voice_engine

New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

2017-06-01 11:02:35 +00:00

.gitignore

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BUILD.gn

Delete webrtc/call.h (replaced with webrtc/call/call.h).

2017-06-08 09:21:17 +00:00

codereview.settings

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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 webrtc/call.h (replaced with webrtc/call/call.h).

2017-06-08 09:21:17 +00:00

LICENSE

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LICENSE_THIRD_PARTY

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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

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README.chromium

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typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

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

video_frame.h

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +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

Reland of Removes usage of native base::android::GetApplicationContext()

2017-05-26 08:51:53 +00:00

whitespace.txt

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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.