webrtc_m130

History

sprang@webrtc.org 0e93257cee Add callbacks for receive channel RTP statistics

This allows a listener to receive new statistics (byte/packet counts, etc) as it
is generated - avoiding the need to poll. This also makes handling stats from
multiple RTP streams more tractable. The change is primarily targeted at the new
video engine API.

TEST=Unit test in ReceiveStatisticsTest.
Integration tests to follow as call tests when fully wired up.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-01-23 10:00:39 +00:00

build

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

2014-01-21 15:54:56 +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, fixes crash on devices with only front cameras.

2014-01-22 21:31:24 +00:00

modules

Add callbacks for receive channel RTP statistics

2014-01-23 10:00:39 +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

Removes usage of ListWrapper from several files.

2014-01-13 15:21:30 +00:00

tools

Now printing less output from compare_videos.py.

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

video

Add configuration and test for extended RTCP reference time reports to new video api.

2014-01-20 08:34:49 +00:00

video_engine

Add callbacks for receive channel RTP statistics

2014-01-23 10:00:39 +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Remove outdated DestroyVideoSendStream comment.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

Set up receiver RTX config using a std::map.

2014-01-20 14:43:55 +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

Set up receiver RTX config using a std::map.

2014-01-20 14:43:55 +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 statistics in video send stream of new video engine api

2014-01-07 09:54:34 +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.