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henrik.lundin 3c938fc5ea Add NetEq delay plotting to event_log_visualizer 
This CL adds the capability to analyze and plot how NetEq behaves in
response to a network trace.

BUG=webrtc:7467

Review-Url: https://codereview.webrtc.org/2876423002
Cr-Commit-Position: refs/heads/master@{#18590}
2017-06-14 13:09:58 +00:00
..
api
Implement operator<< for AudioCodecInfo and AudioCodecSpec
2017-06-14 10:27:40 +00:00
audio
Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:
2017-06-12 19:45:32 +00:00
base
Remove dependency on gunit headers in virtualsocketserver.
2017-06-14 00:13:53 +00:00
call
Log an error in RtpDemuxer::FindSsrcAssociations() if kMaxProcessedSsrcs exceeded
2017-06-13 14:57:31 +00:00
common_audio
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common_video
Removing backward compatible header
2017-06-13 11:25:37 +00:00
examples
Make sure UI methods get called on the main thread
2017-06-13 12:25:33 +00:00
logging
Change rtp header extension AbsoluteSendTime::Write to take time in 24bit format
2017-06-13 09:08:14 +00:00
media
Add field trial for balanced degradation preference.
2017-06-14 12:52:21 +00:00
modules
Add NetEq delay plotting to event_log_visualizer
2017-06-14 13:09:58 +00:00
ortc
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p2p
Adding PortAllocator option to support cases where sockets can't be bound.
2017-06-13 22:49:45 +00:00
pc
Fix Chromium style checker warnings for MockAudioDecoder
2017-06-14 10:31:17 +00:00
sdk
Implement org.webrtc.VideoEncoder using the android MediaCodec.
2017-06-13 16:07:29 +00:00
stats
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system_wrappers
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test
Fix Chromium style checker warnings for MockAudioDecoder
2017-06-14 10:31:17 +00:00
tools
Add NetEq delay plotting to event_log_visualizer
2017-06-14 13:09:58 +00:00
video
Implement kBalanced degradation preference.
2017-06-14 06:25:22 +00:00
voice_engine
Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide:
2017-06-12 19:45:32 +00:00
.gitignore
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BUILD.gn
Delete old include file webrtc/video_frame.h.
2017-06-13 13:05:05 +00:00
codereview.settings
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common_types.cc
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common_types.h
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config.cc
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config.h
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DEPS
Delete old include file webrtc/video_frame.h.
2017-06-13 13:05:05 +00:00
LICENSE
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LICENSE_THIRD_PARTY
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OWNERS
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PATENTS
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PRESUBMIT.py
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README.chromium
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typedefs.h
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video_receive_stream.h
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video_send_stream.h
Add field trial for balanced degradation preference.
2017-06-14 12:52:21 +00:00
webrtc.gni
Adds detection of audio glitches for playout on iOS (reland)
2017-06-13 16:00:18 +00:00
whitespace.txt
…

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.