webrtc_m130

History

ilnik 2a8c2f589a Added Vp9 simulcast tests.

For them implemeted upscaling in libyuv metrics calculation.
Updated maximum number of SL in vp9 encoder to 3.
Refactored names of some fields in Video_quality_check analyzer.

BUG=webrtc:7095

Review-Url: https://codereview.webrtc.org/2681683003
Cr-Commit-Position: refs/heads/master@{#16625}

2017-02-15 10:23:28 +00:00

api

Changing some PeerConnection-related comments.

2017-02-13 17:49:58 +00:00

audio

Remove VoEVideoSync interface.

2017-02-15 08:42:31 +00:00

base

Delete unused files macconversion.h and .cc.

2017-02-15 09:07:57 +00:00

call

Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.

2017-02-13 13:59:46 +00:00

common_audio

Fix for left shift of potentially negative value.

2017-02-09 11:05:59 +00:00

common_video

Added Vp9 simulcast tests.

2017-02-15 10:23:28 +00:00

examples

Revert of Add the url attribute to the IceCandidate (Java Wrapper) (patchset #4 id:120001 of https://codereview.webrtc.org/2690593002/ )

2017-02-14 22:13:56 +00:00

logging

Move AudioDecoder and related stuff to the api/ directory

2017-02-10 16:15:44 +00:00

media

Handle InitDecode and reset in fallback decoder.

2017-02-13 22:11:08 +00:00

modules

Added Vp9 simulcast tests.

2017-02-15 10:23:28 +00:00

p2p

Fix the build break by keeping the old Port::AddAddress method since the

2017-02-14 01:33:27 +00:00

Changing some PeerConnection-related comments.

2017-02-13 17:49:58 +00:00

sdk

Revert of Add the url attribute to the IceCandidate (Java Wrapper) (patchset #4 id:120001 of https://codereview.webrtc.org/2690593002/ )

2017-02-14 22:13:56 +00:00

stats

RTCRTPStreamStats.mediaTrackId renamed to trackId.

2017-01-27 14:35:16 +00:00

system_wrappers

Change NtpTime representation to single uint64_t

2017-02-15 09:18:15 +00:00

test

Remove VoEVideoSync interface.

2017-02-15 08:42:31 +00:00

tools

Move AudioDecoder and related stuff to the api/ directory

2017-02-10 16:15:44 +00:00

video

Added Vp9 simulcast tests.

2017-02-15 10:23:28 +00:00

voice_engine

Remove VoEVideoSync interface.

2017-02-15 08:42:31 +00:00

.gitignore

…

BUILD.gn

Added gn target for rtc_event_log2rtp_dump.

2017-02-08 08:28:09 +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

Remove VoEExternalMedia interface.

2017-02-14 12:55:00 +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

no_op_function.cc

Moving no_op_function.cc out of webrtc/build

2017-01-24 13:25:12 +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

…

README.chromium

…

typedefs.h

Only define NO_RETURN if undefined

2017-01-30 08:54:19 +00:00

video_decoder.h

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +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

Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )

2017-02-09 12:53:45 +00:00

video_send_stream.h

Remove FlexfecConfig and replace with specific struct in VideoSendStream.

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

webrtc.gni

Move iOS tests to XCTest from gtest.

2017-02-14 12:58:56 +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.