webrtc_m130

History

andrew@webrtc.org 27c6980239 Move the volume quantization workaround from VoE to AGC.

Voice engine shouldn't really have to manage this. Instead, have AGC
keep track of the last input volume, so that it can avoid getting stuck
under coarsely quantized conditions.

Add a test to verify the behavior.

TESTED=unittests, and observed that AGC didn't get stuck on a MacBook
where this problem can actually occur.

R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-02-18 20:24:56 +00:00

build

Rename merged webrtc lib to libwebrtc_merged.a.

2014-02-11 18:22:29 +00:00

common_audio

Refactoring common_audio/signal_processing: Removed two macros used by isac only.

2014-02-14 23:12:34 +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

Move the volume quantization workaround from VoE to AGC.

2014-02-18 20:24:56 +00:00

system_wrappers

Use system's cpu_features library

2014-02-05 18:43:46 +00:00

test

Remove external encryption API for VoE.

2014-02-18 11:27:22 +00:00

tools

Now printing less output from compare_videos.py.

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

video

Incorrect overhead calculation when using FEC + RTP extension headers.

2014-02-18 08:40:33 +00:00

video_engine

Remove external encryption API for VoE.

2014-02-18 11:27:22 +00:00

voice_engine

Move the volume quantization workaround from VoE to AGC.

2014-02-18 20:24:56 +00:00

.gitignore

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

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

call.h

Add configuration for cpu overuse detection to video send stream.

2014-01-31 10:05:07 +00:00

common_types.h

Remove external encryption API for VoE.

2014-02-18 11:27:22 +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

Remove external encryption API for VoE.

2014-02-18 11:27:22 +00:00

experiments.h

Add experiment: SkipEncodingUnusedStreams

2014-02-10 09:20:51 +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

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

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

Wire up statistics in video receive stream of new API

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