webrtc_m130

History

terelius 8fbc765421 Ensure the RtcEventLog parser is backwards compatible

The old video send stream configs could contain multiple SSRCs and RTX SSRCs (in case of simulcast). To ensure that the RtcEventLog parser is backwards compatible, we have to return one config for every SSRC.

Also update the parsing functions for the other config types to return the config instead of passing in an output parameter.

BUG=webrtc:7731

Review-Url: https://codereview.webrtc.org/2912113002
Cr-Commit-Position: refs/heads/master@{#18343}

2017-05-31 09:03:16 +00:00

api

Add separate base classes for I420 and I444 buffers

2017-05-30 08:21:59 +00:00

audio

Rename elad.alon to eladalon, to avoid confusion between repositories.

2017-05-25 07:15:35 +00:00

base

Disable SIGPIPE for sockets created on iOS.

2017-05-26 23:27:09 +00:00

call

This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.

2017-05-30 10:52:10 +00:00

common_audio

Moving compile_assert_c.h to webrtc/base

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

common_video

Add separate base classes for I420 and I444 buffers

2017-05-30 08:21:59 +00:00

examples

Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2911053002/ )

2017-05-30 08:48:47 +00:00

logging

Ensure the RtcEventLog parser is backwards compatible

2017-05-31 09:03:16 +00:00

media

Recreate FlexfecReceiveStream separately from VideoReceiveStream.

2017-05-30 08:31:37 +00:00

modules

Drop the rtp_rtcp module's dependency on call.

2017-05-31 06:38:14 +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

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

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

sdk

Re-add removed sdk headers to the framework target.

2017-05-31 08:44:41 +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

Fixing absolute path that was breaking chromium

2017-05-30 13:23:41 +00:00

tools

Ensure the RtcEventLog parser is backwards compatible

2017-05-31 09:03:16 +00:00

video

Persist RTP state for FlexFEC.

2017-05-30 09:32:12 +00:00

voice_engine

This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport.

2017-05-30 10:52:10 +00:00

.gitignore

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BUILD.gn

Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2911053002/ )

2017-05-30 08:48:47 +00:00

call.h

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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

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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.