webrtc_m130

History

Sami Kalliomäki 58c742ce7d Call VideoCapturer.initialize directly from Java.

Passing the call through JNI is unnecessary.

Bug: webrtc:7730
Change-Id: Icf1ecd7e2ea54033342120311c70d47b4a4f7c9a
Reviewed-on: https://chromium-review.googlesource.com/521050
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18408}

2017-06-02 11:27:36 +00:00

api

Update I420Buffer to new VideoFrameBuffer interface

2017-06-01 17:02:26 +00:00

audio

New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

2017-06-01 11:02:35 +00:00

base

SafeMin/SafeMax: Fix wrong return type when given two enum arguments

2017-06-02 11:24:11 +00:00

call

New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

2017-06-01 11:02:35 +00:00

common_audio

Moving compile_assert_c.h to webrtc/base

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

common_video

Update I420Buffer to new VideoFrameBuffer interface

2017-06-01 17:02:26 +00:00

examples

Update I420Buffer to new VideoFrameBuffer interface

2017-06-01 17:02:26 +00:00

logging

Print configured header extensions and codecs in rtc_event_log2text.

2017-06-01 12:23:03 +00:00

media

Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.

2017-06-01 20:22:42 +00:00

modules

Rename rtp_header_extension.h to rtp_header_extension_map.h

2017-06-02 09:11:27 +00:00

ortc

Remove VirtualSocketServer's dependency on PhysicalSocketServer.

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

p2p

Fix issue with send-side bandwidth estimation over TURN TCP connections.

2017-05-27 01:40:05 +00:00

Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport.

2017-06-01 20:22:42 +00:00

sdk

Call VideoCapturer.initialize directly from Java.

2017-06-02 11:27:36 +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

Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device.

2017-06-01 21:10:29 +00:00

tools

Change all numerical string inputs to int and remove unused stderr

2017-06-02 09:16:27 +00:00

video

Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device.

2017-06-01 21:10:29 +00:00

voice_engine

New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender.

2017-06-01 11:02:35 +00:00

.gitignore

…

BUILD.gn

enabling gn check on the whole WebRTC repo

2017-06-01 20:01:48 +00:00

call.h

…

codereview.settings

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common_types.cc

Prevent memory corruption by StreamId::Set

2017-06-02 10:37:48 +00:00

common_types.h

Delete RawVideoType enum, use the VideoType enum instead.

2017-04-28 14:18:05 +00:00

config.cc

Fix default extension id for video content type extension

2017-04-26 09:35:31 +00:00

config.h

Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2809653004/ )

2017-04-11 17:34:31 +00:00

DEPS

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

…

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_frame.h

Move webrtc/video_frame to common_video/include.

2017-05-15 09:42:11 +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

Update adaptation stats to support degradations in both resolution and framerate.

2017-05-16 06:40:18 +00:00

webrtc.gni

Reland of Removes usage of native base::android::GetApplicationContext()

2017-05-26 08:51:53 +00:00

whitespace.txt

…

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.