webrtc_m130

History

henrik.lundin@webrtc.org 139706ec0b Stop using ACM factory in VoiceEngine

The factory injection was introduces in order to facilitate switching
between ACM1 and ACM2. Now, ACM1 is being deprecated, and this switching
mechanism is no longer needed.

BUG=2996
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5954 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-04-22 08:51:21 +00:00

build

Set include_internal_video_capture=1 for video_capture_tests

2014-04-16 12:59:49 +00:00

common_audio

common_audio: VADFree() now returns void

2014-04-22 04:45:35 +00:00

common_video

Propagate capture ntp timestamp from rtp to renderer.

2014-04-15 17:46:33 +00:00

examples

VideoCaptureAndroid: stop referencing ViERenderer

2014-04-08 22:55:07 +00:00

modules

Create ACM2 instance when calling AudioCodingModule::Create

2014-04-22 08:18:42 +00:00

system_wrappers

Add Chromium's ScopedVector.

2014-04-18 21:20:54 +00:00

test

Remove use of tmpnam.

2014-04-16 08:04:26 +00:00

tools

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

video

Let A/V sync test use default AudioCoding module

2014-04-22 08:36:33 +00:00

video_engine

Propagate capture ntp timestamp from rtp to renderer.

2014-04-15 17:46:33 +00:00

voice_engine

Stop using ACM factory in VoiceEngine

2014-04-22 08:51:21 +00:00

.gitignore

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

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

call.h

Remove internal codecs from VideoSendStream.

2014-03-19 08:43:57 +00:00

common_types.h

Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-04-17 10:45:01 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

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

config.h

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +00:00

engine_configurations.h

Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-04-14 14:12:50 +00:00

experiments.h

Adding API for setting bandwidth estimation configurations.

2014-03-25 10:37:31 +00:00

frame_callback.h

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +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

OWNERS

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +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

supplement.gypi

Removes script for generating supplement.gypi also adds git ignore for tools/gn.

2014-01-21 15:54:56 +00:00

transport.h

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

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

typedefs.h

Fix ARM64 detection.

2014-04-01 01:19:08 +00:00

video_engine_tests.isolate

Merge metrics_unittests into video_engine_tests.

2013-12-13 14:31:47 +00:00

video_receive_stream.h

Make VideoReceiveStream::GetStats() const.

2014-02-07 15:32:45 +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

Simplify pacer interface.

2014-03-19 10:59:52 +00:00

webrtc_examples.gyp

Make WebRTC Android examples build without sourcing envsetup.sh

2014-04-15 08:35:00 +00:00

webrtc_perf_tests.isolate

Move realtime tests to webrtc_perf_tests.

2013-12-13 12:48:05 +00:00

webrtc_tests.gypi

Adding NetEq performance test to webrtc_perf_tests

2014-01-10 08:24:04 +00:00

webrtc.gyp

Integrate fake_network_pipe into direct_transport.

2013-12-18 20:28:25 +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.