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Erik Språng 51e60305e6 Update RateStatistics to handle too-little-data case. 
To avoid the case where a single data point or too short window is used,
causing bad behavior due to bad stats, update RateStatistics to return
an Optional rather than a plain rate.

There was also a strange off by one bug where the rate was slightly
overestimated (N + 1 buckets, N ms time window).

These changes requires updates to a number of places, and may very well
cause seeming perf regressions (but the stats were probablty more wrong
previously).

BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2029593002 .

Cr-Commit-Position: refs/heads/master@{#13103}
2016-06-10 20:13:33 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
RTCPeerConnectionInterface.mm createNativeConfiguration and other clean-up.
2016-06-09 10:18:35 +00:00
audio
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
base
Update RateStatistics to handle too-little-data case.
2016-06-10 20:13:33 +00:00
build
iOS: Disable Goma for iOS GN bots.
2016-06-06 16:09:55 +00:00
call
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
common_audio
GN: Add common_audio_unittests and common_video_unittests
2016-06-08 08:28:15 +00:00
common_video
Update RateStatistics to handle too-little-data case.
2016-06-10 20:13:33 +00:00
examples
Use scoped_refptr for On(Add|Remove)Stream and OnDataChannel.
2016-05-31 20:02:30 +00:00
libjingle
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-03 10:09:40 +00:00
media
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
modules
Update RateStatistics to handle too-little-data case.
2016-06-10 20:13:33 +00:00
p2p
Before validating a STUN packet, check it's big enough for a header.
2016-06-09 15:45:52 +00:00
pc
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-03 10:09:40 +00:00
sdk
RTCPeerConnectionInterface.mm createNativeConfiguration and other clean-up.
2016-06-09 10:18:35 +00:00
system_wrappers
GN: Fix errors in rtc_include_tests conditions
2016-06-06 21:04:16 +00:00
test
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
tools
Delete some unused header files.
2016-06-10 11:41:13 +00:00
video
Update RateStatistics to handle too-little-data case.
2016-06-10 20:13:33 +00:00
voice_engine
AudioDecoderIsacT: Require caller to always specify sample rate
2016-06-02 09:59:02 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
GN: Enable api,media,pc and p2p for the 'webrtc' target.
2016-06-10 08:57:05 +00:00
call.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
common_types.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
config.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add root owners to webrtc/OWNERS
2016-05-11 12:47:43 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Roll chromium_revision 086802955f..7fa6701bc5 (396351:398458)
2016-06-08 11:05:27 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Add sender controlled playout delay limits
2016-06-08 07:24:30 +00:00
video_receive_stream.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Remove metrics_default from rtc_media dependencies.
2016-06-01 12:28:57 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
GN: Add SDK tests to rtc_unittests.
2016-06-08 08:14:22 +00:00
webrtc.gyp
GN: Add SDK tests to rtc_unittests.
2016-06-08 08:14:22 +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.