platform-external-webrtc

Files

Danil Chapovalov 8769e1797e Rename rtp::Packet to RtpPacket

TBR=

Bug: webrtc:5565
Change-Id: Ib66d5ddf2989188e4a058f9a674a66afb5f35c7a
Reviewed-on: https://webrtc-review.googlesource.com/1226
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19836}

2017-09-14 13:37:29 +00:00

api

Reject configs with ridiculously many channels instead of crashing

2017-09-14 13:27:39 +00:00

audio

- Removes voe_conference_test.

2017-09-14 12:46:50 +00:00

call

Reland of Use RtxReceiveStream. (patchset #1 id:1 of https://codereview.webrtc.org/3007303002/ )

2017-09-11 09:32:16 +00:00

common_audio

Revert of Remove typedefs.h from webrtc/ root (part 1) (patchset #3 id:40001 of https://codereview.webrtc.org/3007253002/ )

2017-09-08 18:00:21 +00:00

common_video

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

examples

Android AppRTCMobile: Transition local render to new VideoSink interface

2017-09-13 14:11:16 +00:00

logging

Remove no- prefix from command line flags in rtc_event_log2text and rtc_event_log2rtp_dump and negate their meaning.

2017-09-12 12:57:36 +00:00

media

Add new video codec factories

2017-09-14 08:24:56 +00:00

modules

Rename rtp::Packet to RtpPacket

2017-09-14 13:37:29 +00:00

ortc

Move optional.h to webrtc/api/

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

p2p

Add support for changing the TLS elliptic curve set.

2017-09-08 20:02:03 +00:00

- Removes voe_conference_test.

2017-09-14 12:46:50 +00:00

rtc_base

Remove definition of thread annotation macros without RTC_ prefix

2017-09-13 08:25:46 +00:00

rtc_tools

Add a check that sanitizers cause a fatal failure

2017-09-14 11:05:01 +00:00

sdk

Make toVideoFrame in I420Frame return the original frame.

2017-09-14 11:43:59 +00:00

stats

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

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

system_wrappers

Update thread annotiation macros to use RTC_ prefix

2017-09-09 11:17:22 +00:00

test

- Removes voe_conference_test.

2017-09-14 12:46:50 +00:00

video

Reland of Delete Rtx-related methods from RTPPayloadRegistry. (patchset #1 id:1 of https://codereview.webrtc.org/3011093002/ )

2017-09-13 14:13:57 +00:00

voice_engine

- Removes voe_conference_test.

2017-09-14 12:46:50 +00:00

.gitignore

…

BUILD.gn

Removing webrtc/config.h

2017-09-14 11:02:39 +00:00

codereview.settings

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common_types.cc

Reland of move deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3010553002/ )

2017-09-13 12:16:26 +00:00

common_types.h

Reland of move deprecated CodecType methods. (patchset #1 id:1 of https://codereview.webrtc.org/3010553002/ )

2017-09-13 12:16:26 +00:00

DEPS

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

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

LICENSE

…

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

Revert of Remove typedefs.h from webrtc/ root (part 1) (patchset #3 id:40001 of https://codereview.webrtc.org/3007253002/ )

2017-09-08 18:00:21 +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.