webrtc_m130

History

Anders Carlsson bd2220a9c4 Support more formats in RTCVideoFrame

Implement Obj-C version of webrtc::VideoFrameBuffer and use that in
RTCVideoFrame.

Bug: webrtc:7785
Change-Id: I49f42bcf451dd6769b3a79a65fe7b400dce22677
Reviewed-on: https://chromium-review.googlesource.com/536773
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18691}

2017-06-21 06:30:23 +00:00

api

Implement timing frames.

2017-06-19 14:18:55 +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

Enable SNI in ssl adapter.

2017-06-16 23:30:48 +00:00

call

Support building WebRTC without audio and video.

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

common_audio

Moving compile_assert_c.h to webrtc/base

2017-05-15 15:06:47 +00:00

common_video

Use uint8 pointer instead of std::vector in NV12Scale.

2017-06-19 15:08:48 +00:00

examples

Use information about blacklisted devices in video_quality_loopback_test

2017-06-19 08:16:45 +00:00

logging

Support building WebRTC without audio and video.

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

media

Implement timing frames.

2017-06-19 14:18:55 +00:00

modules

Use constexpr to avoid a static initializer

2017-06-20 17:57:09 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

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

p2p

Enable SNI in ssl adapter.

2017-06-16 23:30:48 +00:00

Fix uploading of available send bitrate statistics.

2017-06-19 15:58:15 +00:00

sdk

Support more formats in RTCVideoFrame

2017-06-21 06:30:23 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Prevent data race in GetStaticInstance

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

test

Added new AudioProcessing fuzzer

2017-06-20 12:26:55 +00:00

tools

Add NetEq delay plotting to event_log_visualizer

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

video

Minor updates to VideoReceiveStream:

2017-06-20 09:44:38 +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

…

BUILD.gn

Templated AudioEncoderFactory

2017-06-16 17:42:05 +00:00

codereview.settings

…

common_types.cc

Implement timing frames.

2017-06-19 14:18:55 +00:00

common_types.h

Implement timing frames.

2017-06-19 14:18:55 +00:00

config.cc

Implement timing frames.

2017-06-19 14:18:55 +00:00

config.h

Implement timing frames.

2017-06-19 14:18:55 +00:00

DEPS

Delete old include file webrtc/video_frame.h.

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

LICENSE

…

LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

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

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

…

PRESUBMIT.py

…

README.chromium

…

typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_receive_stream.h

Delete VieRemb class, move functionality to PacketRouter.

2017-04-19 06:38:35 +00:00

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

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.