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== Introduction == | |||
This efforts aims at building tools and frameworks for analyzing audio and video | |||
quality for the Firefox WebRTC implementation. WebRTC involves peer-to-peer rich multimedia | |||
communications across variety of end-user devices and network conditions. The | |||
framework must be able to provide necessary means to simulate/emulate these usage | |||
configurations in order to analyze the behavior of WebRTC implementation in the firefox. | |||
== Scope (WIP) == | |||
Following is wishlist of functionalities that the framework must be able to support | |||
eventually. | |||
# Audio and Video Quality Analysis | |||
## Quantitative measurements | |||
## Qualitative measurements | |||
#Latency Impact Analysis | |||
## End to End Latency | |||
## Latency impacts due to local processing | |||
## Latency impacts under simulated constrained network conditions | |||
#RTCP based analysis | |||
== Typical WebRTC Media Pipeline== | |||
[[File:Firefox_WebRTC_Pipeline.png|650px]] | |||
== Peer Connection Audio Quality == | |||
[[File:AudioPerf-Setup.png|650px]] | |||
Analysis of tools and techniques for measuring WebRTC Audio Performance. | Analysis of tools and techniques for measuring WebRTC Audio Performance. | ||
This page is under construction. please come back later for more complete information | This page is under construction. please come back later for more complete information | ||