platform-external-webrtc

Files

sprang dceb42da3e Update screen simulcast config and fix periodic encoder param update

Lower then bitrate so that the delta between the highest layer of the
lower stream and lowest layer of the higher stream is not too large.

Also fix a bug in vie_encoder where the codec was not perioducally
updated unless a new bitrate estimate was triggered.

BUG=webrtc:4172

Review-Url: https://codereview.webrtc.org/2791273002
Cr-Commit-Position: refs/heads/master@{#18232}

2017-05-23 13:45:08 +00:00

api

Renaming probing_interval to bwe_period globally.

2017-05-18 21:32:41 +00:00

audio

Store/restore RTP state for audio streams with same SSRC within a call

2017-05-23 13:07:11 +00:00

base

Fixing potential AsyncInvoker deadlock that occurs for "reentrant" invocations.

2017-05-22 22:32:51 +00:00

call

Store/restore RTP state for audio streams with same SSRC within a call

2017-05-23 13:07:11 +00:00

common_audio

Moving compile_assert_c.h to webrtc/base

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

common_video

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +00:00

examples

AppRTCMobile support to turn off the WebRTC AGC and HPF

2017-05-18 22:09:33 +00:00

logging

Replace AudioSendStream::Config with rtclog::StreamConfig.

2017-05-22 17:12:26 +00:00

media

Update screen simulcast config and fix periodic encoder param update

2017-05-23 13:45:08 +00:00

modules

Preserve level controller output when no other effects are active

2017-05-23 12:33:56 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

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

p2p

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

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

2017-05-19 20:09:47 +00:00

sdk

Field trial support to whenever possible turn off the AGC and HPF

2017-05-22 22:48:47 +00:00

stats

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

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

system_wrappers

Adding target to track asm_defines.h

2017-05-15 21:59:17 +00:00

test

Store/restore RTP state for audio streams with same SSRC within a call

2017-05-23 13:07:11 +00:00

tools

Replace AudioSendStream::Config with rtclog::StreamConfig.

2017-05-22 17:12:26 +00:00

video

Reland of use allocated encoders in SimulcastEncoderAdapter. (patchset #1 id:1 of https://codereview.webrtc.org/2893003002/ )

2017-05-23 10:32:16 +00:00

voice_engine

Replace AudioSendStream::Config with rtclog::StreamConfig.

2017-05-22 17:12:26 +00:00

.gitignore

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BUILD.gn

Revert of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2890733003/ )

2017-05-17 13:52:58 +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

Add read support of RtpStreamId/RepairedRtpStreamId header extensions.

2017-04-19 09:59:48 +00:00

common_types.h

Delete RawVideoType enum, use the VideoType enum instead.

2017-04-28 14:18:05 +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

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +00:00

LICENSE

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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

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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

Replace sanitizers:deps with exe_and_shlib_deps (webrtc)

2017-05-19 06:51:46 +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.