platform-external-webrtc

Files

nisse 284542b882 Make OverheadObserver::OnOverheadChanged count RTP headers only

This lets the RTP code be unaware of lower layers, and the
SetTransportOverhead method is deleted from RTPSender and RtpRtcp.

Instead, that method is added to CongestionController and
TransportFeedbackAdapter, where it is more appropriate.

BUG=wertc:6847

Review-Url: https://codereview.webrtc.org/2589743002
Cr-Commit-Position: refs/heads/master@{#15995}

2017-01-10 16:58:32 +00:00

androidjunit

Format all Java in WebRTC.

2016-09-29 11:12:51 +00:00

api

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

audio

GN: Refactor webrtc_nonparallel_tests and audio_tests to avoid crossing package boundaries.

2017-01-05 14:03:24 +00:00

base

Drop unneeded includes of base/stream.h.

2017-01-10 11:44:41 +00:00

build

Disable some filesystem tests that don't make sense on iOS.

2017-01-10 11:00:41 +00:00

call

Clean up ramp-up tests and make sure they all pass.

2017-01-09 12:19:24 +00:00

common_audio

Fix for left shift of potentially negative value.

2017-01-10 11:37:20 +00:00

common_video

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

examples

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

logging

Convert rtc_event_log from webrtc::Clock to rtc::TimeMicros.

2016-12-20 13:03:58 +00:00

media

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

modules

Make OverheadObserver::OnOverheadChanged count RTP headers only

2017-01-10 16:58:32 +00:00

p2p

Add disabled certificate check support to IceServer PeerConnection API.

2017-01-09 16:35:45 +00:00

Reland of: Separating SCTP code from BaseChannel/MediaChannel.

2017-01-09 22:53:41 +00:00

sdk

Android: Wrap webrtc::field_trial::FindFullName in Java

2017-01-10 16:02:03 +00:00

stats

RTCIceCandidateStats.transportId added and collected.

2017-01-02 17:59:31 +00:00

system_wrappers

Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )

2017-01-09 09:22:16 +00:00

test

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

tools

Add method needed to extract frame capture and arrival timestamps from rtc event logs.

2016-12-20 16:51:52 +00:00

video

Make OverheadObserver::OnOverheadChanged count RTP headers only

2017-01-10 16:58:32 +00:00

voice_engine

Make OverheadObserver::OnOverheadChanged count RTP headers only

2017-01-10 16:58:32 +00:00

.gitignore

…

BUILD.gn

Reland of "Remove webrtc/libjingle/{xmllite,xmpp}"

2017-01-09 07:02:01 +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

Reject XR TargetBitrate items with unsupported layer indices

2016-12-06 14:09:00 +00:00

common_types.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

config.cc

Keep all codec parameters in VideoReceiveStream::Decoder

2016-11-25 11:56:41 +00:00

config.h

Keep all codec parameters in VideoReceiveStream::Decoder

2016-11-25 11:56:41 +00:00

DEPS

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

LICENSE

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LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

OWNERS

Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it

2016-11-29 10:52:22 +00:00

PATENTS

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PRESUBMIT.py

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README.chromium

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typedefs.h

Remove extra uses of basictypes.h.

2016-12-09 14:54:08 +00:00

video_decoder.h

Move VideoDecoder::Create() logic to separate internal video decoder factory

2016-12-01 08:27:35 +00:00

video_encoder.h

Move usage of QualityScaler to ViEEncoder.

2016-11-29 09:44:22 +00:00

video_frame.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

video_receive_stream.h

Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.

2016-11-28 15:02:19 +00:00

video_send_stream.h

Implement periodic bandwidth probing in application-limited region.

2016-11-28 21:11:24 +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.