platform-external-webrtc

Files

johan f53d7374cf H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.

- Changed method name to clarify that entire Nalus are expected.
- Added unit test code.
- Adjusted InsetSpsPpsNalus() implementation to above requirement.

BUG=webrtc:5948

Review-Url: https://codereview.webrtc.org/2638933002
Cr-Commit-Position: refs/heads/master@{#16221}

2017-01-23 17:29:33 +00:00

androidjunit

Format all Java in WebRTC.

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

api

Fixed public_deps for libjingle_peerconnection{,_api}

2017-01-23 15:37:43 +00:00

audio

Pass SdpAudioFormat through Channel, without converting to CodecInst

2017-01-19 15:03:59 +00:00

base

Delete or update left-over ASSERT use and comments.

2017-01-23 15:43:05 +00:00

build

Removing unused code from webrtc/build

2017-01-23 16:49:12 +00:00

call

Pass SdpAudioFormat through Channel, without converting to CodecInst

2017-01-19 15:03:59 +00:00

common_audio

Replace RTC_DCHECK(false) with RTC_NOTREACHED().

2017-01-12 10:24:27 +00:00

common_video

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

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

examples

Create //webrtc/api:libjingle_peerconnection_api + refactorings.

2017-01-23 12:56:25 +00:00

logging

GN: Refactor so that WebRTC compiles with rtc_enable_protobuf=false.

2017-01-17 10:37:34 +00:00

media

Removed double-special-casing of ISAC in libjingle and WebRtcVoE.

2017-01-23 16:55:48 +00:00

modules

H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.

2017-01-23 17:29:33 +00:00

p2p

Create //webrtc/api:libjingle_peerconnection_api + refactorings.

2017-01-23 12:56:25 +00:00

Removed double-special-casing of ISAC in libjingle and WebRtcVoE.

2017-01-23 16:55:48 +00:00

sdk

Minor style change suggested by internal static analysis tool.

2017-01-23 14:09:35 +00:00

stats

RTCMediaStreamTrackStats.kind added and collected.

2017-01-17 10:53:23 +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

Add TransportFeedbackPacketLossTracker.

2017-01-23 16:07:05 +00:00

tools

Script to start stubbed loopback video test with Espresso

2017-01-23 15:55:42 +00:00

video

H264SpsPpsTracker.InsertSpsPpsNalus() should accept Nalus with header.

2017-01-23 17:29:33 +00:00

voice_engine

Add TransportFeedbackPacketLossTracker.

2017-01-23 16:07:05 +00:00

.gitignore

…

BUILD.gn

Create //webrtc/api:libjingle_peerconnection_api + refactorings.

2017-01-23 12:56:25 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

codereview.settings

…

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

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

config.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +00:00

DEPS

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

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

LICENSE

…

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

RTCMediaStreamTrackStats.framesDropped collected by RTCStatsCollector.

2017-01-23 15:21:55 +00:00

video_send_stream.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

2017-01-16 14:59:19 +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.