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peah 135259ac8f In order to be able to analyze the AGC behavior on 
aecdump recordings in an efficient manner, it is
important to be able to use a standardized analysis
script. For this to be feasible, data log points should
be present.

This CL adds those logpoints as well as the framework
needed to for those to work.

BUG=webrtc:6564

Review-Url: https://codereview.webrtc.org/2457783003
Cr-Commit-Position: refs/heads/master@{#14812}
2016-10-28 10:12:15 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Testing of FileVideoCapturer.
2016-10-27 14:26:44 +00:00
audio
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
base
Fix "IsLoopbackIp" to cover all loopback addresses; not just 127.0.0.1.
2016-10-28 01:30:28 +00:00
build
Restore symbol level for Android builds.
2016-10-27 07:00:20 +00:00
call
Remove use of VoECodec in video/call tests.
2016-10-27 07:23:14 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Reland of Move bitstream parser to more appropriate directory. (patchset #1 id:1 of https://codereview.webrtc.org/2430353004/ )
2016-10-26 09:48:24 +00:00
examples
Testing of FileVideoCapturer.
2016-10-27 14:26:44 +00:00
libjingle
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
logging
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
media
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
modules
In order to be able to analyze the AGC behavior on
2016-10-28 10:12:15 +00:00
p2p
Fix "IsLoopbackIp" to cover all loopback addresses; not just 127.0.0.1.
2016-10-28 01:30:28 +00:00
pc
Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit.
2016-10-25 17:50:41 +00:00
sdk
iOS: Optimize video scaling and cropping
2016-10-20 10:34:32 +00:00
stats
RTCStats equality operator added.
2016-10-25 11:31:27 +00:00
system_wrappers
Delete DataLog abstraction, which was almost unused.
2016-10-21 08:53:01 +00:00
test
Remove use of VoECodec in video/call tests.
2016-10-27 07:23:14 +00:00
tools
Use bayesian estimate of acked bitrate.
2016-10-27 15:19:34 +00:00
video
Remove use of VoECodec in video/call tests.
2016-10-27 07:23:14 +00:00
voice_engine
New statistics interface for APM
2016-10-28 08:32:24 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Remove remnants of libsrtp1
2016-10-25 16:47:31 +00:00
call.h
Integrate FlexfecReceiveStream with Call.
2016-10-24 06:37:24 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common_types.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
config.h
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +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
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_frame.h
Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ )
2016-10-24 07:44:17 +00:00
video_receive_stream.h
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
video_send_stream.h
Implement framesEncoded stat in video send ssrc stats.
2016-10-24 09:09:39 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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.