webrtc_m130

History

henrikg@webrtc.org 863b536100 Allow opening an AEC dump from an existing file handle.

This is necessary for Chromium to be able enable the dump with the sanbox enabled. It will open the file in the browser process and pass the handle to the render process.

This changes FileWrapper to deal with the case were the file handle is not managed by the wrapper.

BUG=2567
R=andrew@webrtc.org, henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5239 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-12-06 16:05:17 +00:00

build

Remove the long disabled WEBRTC_SVNREVISION define.

2013-12-04 17:00:44 +00:00

common_audio

Upgrade scoped_ptr to Chromium's latest version.

2013-10-22 12:50:00 +00:00

common_video

Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"

2013-10-31 23:41:04 +00:00

modules

Allow opening an AEC dump from an existing file handle.

2013-12-06 16:05:17 +00:00

system_wrappers

Allow opening an AEC dump from an existing file handle.

2013-12-06 16:05:17 +00:00

test

Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.

2013-12-05 16:15:11 +00:00

tools

Made video quality toolchain more configurable.

2013-11-25 14:10:20 +00:00

video

Stop video capturers in multi-stream test.

2013-12-06 15:48:17 +00:00

video_engine

Fraction lost statistics not being reported

2013-12-06 11:56:55 +00:00

voice_engine

Allow opening an AEC dump from an existing file handle.

2013-12-06 16:05:17 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

call.h

Adds support for sending redundant payloads over RTX.

2013-12-04 10:24:26 +00:00

common_types.h

Add callbacks for send channel rtp statistics

2013-12-05 14:29:02 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

config.h

Rename RTP-extension constants.

2013-11-20 11:48:56 +00:00

engine_configurations.h

Adds a new voice engine warning for the typing noise off state.

2013-09-17 18:09:20 +00:00

experiments.h

Adds support for sending redundant payloads over RTX.

2013-12-04 10:24:26 +00:00

frame_callback.h

Implement and test EncodedImageCallback in new ViE API.

2013-11-26 11:41:59 +00:00

LICENSE

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

LICENSE_THIRD_PARTY

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +00:00

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PRESUBMIT.py

Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.

2013-07-22 22:32:50 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

transport.h

Rename newapi::Transport::SendRTP()->SendRtp().

2013-11-20 12:17:04 +00:00

typedefs.h

Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined.

2013-10-23 19:11:32 +00:00

video_engine_tests.isolate

Remove CallTest dependency on voice_engine/test/.

2013-12-05 14:24:17 +00:00

video_receive_stream.h

Set local SSRC for VideoReceiveStream.

2013-12-03 10:13:04 +00:00

video_renderer.h

Separate Call API/build files from video_engine/.

2013-10-28 16:32:01 +00:00

video_send_stream.h

Implement and test EncodedImageCallback in new ViE API.

2013-11-26 11:41:59 +00:00

webrtc_tests.gypi

Remove CallTest dependency on voice_engine/test/.

2013-12-05 14:24:17 +00:00

webrtc.gyp

Move implementation files out of the webrtc/ root.

2013-12-05 12:11:47 +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.