webrtc_m130

History

Sami Kalliomäki bc061b4280 Create AndroidVideoBuffer and allow renderers to consume it.

Bug: webrtc:7760
Change-Id: I3e3fddf48090ae27b226c65ddbb51f2c3d8dc544
Reviewed-on: https://chromium-review.googlesource.com/535638
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18624}

2017-06-16 08:31:37 +00:00

api

Support building WebRTC without audio and video.

2017-06-15 19:52:32 +00:00

audio

Also scan stderr for audio files to test, due to change in Android test_runner

2017-06-14 16:35:11 +00:00

base

Delete SignalSrtpError.

2017-06-16 07:57:21 +00:00

call

Support building WebRTC without audio and video.

2017-06-15 19:52:32 +00:00

common_audio

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common_video

Support building WebRTC without audio and video.

2017-06-15 19:52:32 +00:00

examples

Make sure UI methods get called on the main thread

2017-06-13 12:25:33 +00:00

logging

Support building WebRTC without audio and video.

2017-06-15 19:52:32 +00:00

media

Delete SignalSrtpError.

2017-06-16 07:57:21 +00:00

modules

Validate references of frames inserted into FrameBuffer2.

2017-06-15 16:06:21 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

2017-05-17 01:00:06 +00:00

p2p

Adding PortAllocator option to support cases where sockets can't be bound.

2017-06-13 22:49:45 +00:00

Delete SignalSrtpError.

2017-06-16 07:57:21 +00:00

sdk

Create AndroidVideoBuffer and allow renderers to consume it.

2017-06-16 08:31:37 +00:00

stats

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system_wrappers

Prevent data race in GetStaticInstance

2017-05-29 07:01:14 +00:00

test

Fix Chromium style checker warnings for MockAudioDecoder

2017-06-14 10:31:17 +00:00

tools

Add NetEq delay plotting to event_log_visualizer

2017-06-14 13:09:58 +00:00

video

Use the same QP max for tests as in production

2017-06-15 13:34:42 +00:00

voice_engine

Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:

2017-06-12 19:45:32 +00:00

.gitignore

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BUILD.gn

Delete old include file webrtc/video_frame.h.

2017-06-13 13:05:05 +00:00

codereview.settings

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common_types.cc

Add RSID-based demuxing to RtpDemuxer

2017-06-08 11:19:13 +00:00

common_types.h

Add RSID-based demuxing to RtpDemuxer

2017-06-08 11:19:13 +00:00

config.cc

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config.h

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DEPS

Delete old include file webrtc/video_frame.h.

2017-06-13 13:05:05 +00:00

LICENSE

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LICENSE_THIRD_PARTY

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OWNERS

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PATENTS

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PRESUBMIT.py

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README.chromium

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typedefs.h

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video_receive_stream.h

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video_send_stream.h

Add field trial for balanced degradation preference.

2017-06-14 12:52:21 +00:00

webrtc.gni

Adds detection of audio glitches for playout on iOS (reland)

2017-06-13 16:00:18 +00:00

whitespace.txt

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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.