webrtc_m130

History

VladimirTechMan bf5a801097 Definitions of video-codec name constants don't match their declarations

Constants kRTCVp8CodecName, kRTCVp9CodecName, kRTCH264CodecName declared
in RTCRtpCodecParameters.h were defined without the "RTC" part in their
names, thus using them in the app code resulted in linking errors. This
patch fixes the naming mismatch.

BUG=webrtc:7721

Review-Url: https://codereview.webrtc.org/2910863002
Cr-Commit-Position: refs/heads/master@{#18363}

2017-06-01 03:09:12 +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

Support epoll in PhysicalSocketServer.

2017-05-31 20:09:18 +00:00

call

Create unit tests for RtpDemuxer

2017-05-31 16:12:25 +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

Use oneof in event log proto to reduce memory consumption.

2017-05-31 12:45:56 +00:00

media

Recreate FlexfecReceiveStream separately from VideoReceiveStream.

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

modules

Small cleanup of rtp_rtcp testAPI tests.

2017-05-31 14:35:16 +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

Definitions of video-codec name constants don't match their declarations

2017-06-01 03:09:12 +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

Propagate input capture time in ViEEncoder::OnFrame.

2017-05-31 13:33:21 +00:00

tools

Ensure the RtcEventLog parser is backwards compatible

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

video

Disable flaky test EndToEndTest.TestFlexfecRtpStatePreservation on linux for now.

2017-05-31 14:46:56 +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

…

BUILD.gn

Create unit tests for RtpDemuxer

2017-05-31 16:12:25 +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

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.