webrtc_m130

History

nisse 5b4b522641 Delete Rtx-related methods from RTPPayloadRegistry.

Delete methods IsRtx, IsEncapsulated and RestoreOriginalPacket.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/3006993002
Cr-Commit-Position: refs/heads/master@{#19739}

2017-09-08 09:11:54 +00:00

api

Change RtpSender to have multiple stream_ids

2017-09-07 17:03:24 +00:00

audio

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

call

Add reporting of googContentType via GetStats on send side

2017-09-06 19:32:35 +00:00

common_audio

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

common_video

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

examples

Make PeerConnectionClient non-singleton.

2017-08-31 15:03:46 +00:00

logging

Remove RtcEventLogImpl forward declaration from rtc_event_log.h

2017-09-07 15:02:35 +00:00

media

Add reporting of googContentType via GetStats on send side

2017-09-06 19:32:35 +00:00

modules

Delete Rtx-related methods from RTPPayloadRegistry.

2017-09-08 09:11:54 +00:00

ortc

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

p2p

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

In PC integration tests, create tracks/streams with random IDs.

2017-09-07 21:12:05 +00:00

rtc_base

Reland of "Decoupling rtc_base from Obj-C code"

2017-09-07 14:36:28 +00:00

rtc_tools

Add visualization of late feedback packets, and avoid plotting incorrect delays computed based on those.

2017-09-06 09:06:32 +00:00

sdk

Improve the style of the objc wrapper for PeerConnectionInterface::SetBitrate.

2017-09-05 21:43:03 +00:00

stats

Recently we moved webrtc/base to webrtc/rtc_base, so these

2017-08-30 11:45:58 +00:00

system_wrappers

Revert of Rename thread annotation macros to have RTC prefix for syncrhonization primitives. (patchset #1 id:1 of https://codereview.webrtc.org/3004393002/ )

2017-09-06 08:38:35 +00:00

test

Keep track of the capacity delay error in the FakeNetworkPipe.

2017-09-07 16:08:50 +00:00

video

Delete Rtx-related methods from RTPPayloadRegistry.

2017-09-08 09:11:54 +00:00

voice_engine

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

.gitignore

…

BUILD.gn

Add VideoProcessorIntegrationTest for MediaCodec implementations.

2017-09-07 14:50:06 +00:00

codereview.settings

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common_types.cc

Revert of Remove deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3009583002/ )

2017-08-28 14:16:43 +00:00

common_types.h

Move optional.h to webrtc/api/

2017-09-05 15:43:13 +00:00

config.h

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +00:00

DEPS

Move RtpExtension to api/ directory and config.h/.cc to call/.

2017-09-01 13:29:30 +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

webrtc.gni

Revert of Add //build/android/pylib/results/ to Android isolate packaging (patchset #2 id:40001 of https://codereview.webrtc.org/3011023002/ )

2017-09-06 19:11:27 +00:00

whitespace.txt

…

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.