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Cesar Magalhaes bf40b42af5 Modified Simulation Framework Jitter Model. 
Using a right-sided (absolute value), truncated gaussian distribution originally with zero mean.

Currently truncated at x = 3 * std_dev.

Added expected value computation.

Modified jitter unittests accordingly.

BUG=webrtc:4848
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1237303002 .

Cr-Commit-Position: refs/heads/master@{#9587}
2015-07-15 17:47:22 +00:00
..
base
AppRTCDemo file logging.
2015-07-14 19:55:56 +00:00
build
Make .gni less sensitive to type of arm_use_neon flags
2015-07-09 16:45:09 +00:00
common_audio
Remove a cast again, after it was shown to worsen Windows perf.
2015-06-29 22:16:48 +00:00
common_video
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
examples/android/media_demo
VoE: Remove unused interfaces
2015-05-26 08:25:00 +00:00
libjingle
Remove webrtc/libjingle/{examples,session}.
2015-06-12 11:13:46 +00:00
modules
Modified Simulation Framework Jitter Model.
2015-07-15 17:47:22 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
1. Override and virtual has to be consistent.
2015-07-14 00:08:11 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Fix some clang warnings with -Wmissing-braces in WebRTC.
2015-07-02 00:52:18 +00:00
test
Fix follow-up in webrtc/test/field_trial.cc.
2015-07-10 00:21:10 +00:00
tools
Remove empty-string comparisons.
2015-07-09 14:48:27 +00:00
video
Base A/V synchronization on sync_labels.
2015-07-15 15:03:04 +00:00
video_engine
Base A/V synchronization on sync_labels.
2015-07-15 15:03:04 +00:00
voice_engine
Remove ResetStatistics from RTP feedback.
2015-07-07 15:32:56 +00:00
.gitignore
…
audio_receive_stream.h
Base A/V synchronization on sync_labels.
2015-07-15 15:03:04 +00:00
audio_send_stream.h
VoE2 API draft
2015-06-08 11:05:07 +00:00
BUILD.gn
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
call.h
C++11 in-class member initialization in Call configs.
2015-06-11 10:38:46 +00:00
codereview.settings
Update deeper codereview.settings files to match the root.
2015-06-24 19:40:39 +00:00
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Revert the process noise co-variance of the bitrate over-use estimator to its value prior to r9545.
2015-07-10 09:28:46 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Make Config::default_value leak instead of having an exit-time destructor.
2015-05-23 00:50:33 +00:00
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Add options for NetEq fast accelerate mode through libjingle
2015-06-01 08:29:55 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
experiments.h
…
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
…
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
Allow for framerate reduction for HW encoder.
2015-07-13 23:26:40 +00:00
video_engine_tests.isolate
…
video_frame.h
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
video_receive_stream.h
Base A/V synchronization on sync_labels.
2015-07-15 15:03:04 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Move frame input (ViECapturer) to webrtc/video/.
2015-06-26 04:58:23 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Move frame input (ViECapturer) to webrtc/video/.
2015-06-26 04:58:23 +00:00
webrtc.gyp
Merge video_engine_core into webrtc target.
2015-06-22 05:57:26 +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.