platform-external-webrtc

Files

philipel 0f0763d86d Make the new jitter buffer the default jitter buffer.

This CL contains only the changes necessary to make the switch to the new jitter
buffer, clean up will be done in follow up CLs.

In this CL:
 - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
   new video jitter buffer the default one.
 - Moved WebRTC.Video.KeyFramesReceivedInPermille and
   WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.

BUG=webrtc:5514

Review-Url: https://codereview.webrtc.org/2627463004
Cr-Commit-Position: refs/heads/master@{#16114}

2017-01-17 11:31:15 +00:00

androidjunit

Format all Java in WebRTC.

2016-09-29 11:12:51 +00:00

api

RTCMediaStreamTrackStats.kind added and collected.

2017-01-17 10:53:23 +00:00

audio

Fix for bwe with overhead on audio only calls.

2017-01-17 07:55:07 +00:00

base

Delete unused file faketaskrunner.h.

2017-01-16 08:30:33 +00:00

build

Replace default locale with US locale on Android.

2017-01-16 12:57:32 +00:00

call

Let FlexfecReceiveStreamImpl send RTCP RRs.

2017-01-17 09:33:54 +00:00

common_audio

Replace RTC_DCHECK(false) with RTC_NOTREACHED().

2017-01-12 10:24:27 +00:00

common_video

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

examples

Espresso test case to control loopback call

2017-01-16 10:23:09 +00:00

logging

GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false.

2017-01-17 10:37:34 +00:00

media

Let FlexfecReceiveStreamImpl send RTCP RRs.

2017-01-17 09:33:54 +00:00

modules

Make the new jitter buffer the default jitter buffer.

2017-01-17 11:31:15 +00:00

p2p

RTCTransportStats.dtlsState replaces .activeConnection

2017-01-16 15:38:02 +00:00

Provide better message for when RTCP mux "require" policy is triggered.

2017-01-13 19:34:39 +00:00

sdk

Android GlTextureFrameBuffer: Re-attach texture in setSize

2017-01-17 10:51:06 +00:00

stats

RTCMediaStreamTrackStats.kind added and collected.

2017-01-17 10:53:23 +00:00

system_wrappers

Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )

2017-01-09 09:22:16 +00:00

test

Let FlexfecReceiveStreamImpl send RTCP RRs.

2017-01-17 09:33:54 +00:00

tools

Comparison of videos with reference frame not starting from zero

2017-01-17 11:24:57 +00:00

video

Make the new jitter buffer the default jitter buffer.

2017-01-17 11:31:15 +00:00

voice_engine

UdpTransport:IsIpAddressValid: Added extra :: check for ipv6

2017-01-17 10:31:37 +00:00

.gitignore

…

BUILD.gn

Rename base/analytics/ to base/numerics/

2017-01-13 17:14:33 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

codereview.settings

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common_types.cc

Reject XR TargetBitrate items with unsupported layer indices

2016-12-06 14:09:00 +00:00

common_types.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

config.cc

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

config.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

DEPS

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

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

OWNERS

Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it

2016-11-29 10:52:22 +00:00

PATENTS

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PRESUBMIT.py

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README.chromium

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typedefs.h

Remove extra uses of basictypes.h.

2016-12-09 14:54:08 +00:00

video_decoder.h

Move VideoDecoder::Create() logic to separate internal video decoder factory

2016-12-01 08:27:35 +00:00

video_encoder.h

Move usage of QualityScaler to ViEEncoder.

2016-11-29 09:44:22 +00:00

video_frame.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

video_receive_stream.h

Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.

2016-11-28 15:02:19 +00:00

video_send_stream.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +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.