platform-external-webrtc

Files

henrik.lundin@webrtc.org 84eb0e952e Add clean test to NetEq perf test

Add another test to NetEqPerformanceTest with no packet losses or
clock drift. The purpose of this test would be to focus on the
"clean" code path, i.e., the path taken when there are no network
problems. The reason is that this code path is presumably much
lighter in complexity, and regressions could easily drown in the
heavier code involved when combating losses and drift.

BUG=2859
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5452 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-01-28 21:50:35 +00:00

build

Add new API (webrtc.gyp:webrtc) to merge_libs.gyp.

2014-01-24 10:43:47 +00:00

common_audio

Remove the requirement to call set_sample_rate_hz and friends.

2014-01-07 17:45:09 +00:00

common_video

Add SwapFrame() to VideoSendStreamInput.

2013-12-11 16:26:16 +00:00

examples

Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera.

2014-01-24 18:42:12 +00:00

modules

Add clean test to NetEq perf test

2014-01-28 21:50:35 +00:00

system_wrappers

Removes the remaining uses of the list wrapper class and the list wrapper class.

2014-01-13 22:41:34 +00:00

test

Wire up RTX in VideoReceiveStream.

2014-01-24 09:30:53 +00:00

tools

Now printing less output from compare_videos.py.

2014-01-07 17:59:30 +00:00

video

Drop early packets when not sending in TransportAdapter.

2014-01-27 13:03:02 +00:00

video_engine

Revert 5421 "Fix deadlock on register/unregister observer while ..."

2014-01-27 22:00:57 +00:00

voice_engine

Output logs to stderr from voe_cmd_test by default.

2014-01-21 22:17:43 +00:00

.gitignore

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

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

call.h

Remove outdated DestroyVideoSendStream comment.

2014-01-07 12:27:22 +00:00

common_types.h

Remove empty VideoCodecGeneric struct.

2014-01-14 12:55:59 +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 RTX in VideoReceiveStream.

2014-01-24 09:30:53 +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

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

Correctly define OVERRIDE when building with g++ 4.7 and C++11 support

2013-12-10 19:20:46 +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

Wire up RTX in VideoReceiveStream.

2014-01-24 09:30:53 +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

Wire up RTX in VideoReceiveStream.

2014-01-24 09:30:53 +00:00

webrtc_examples.gyp

Android example apps: fixes issue where useful failure information was suppressed.

2014-01-21 19:03:51 +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.