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Media/WebRTC Audio Perf

977 bytes added, 20:12, 9 December 2013
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## Latency impacts under simulated constrained network conditions
#RTCP based analysis
#Codec Configuration variability analysis
#Hardware and Platform variability analysis
 
== Typical WebRTC Media Pipeline==
Below picture captures various components involved in the flow of
media captured from mic/camera till it gets transported. The reverse direction
follows a similar path back till the RTP packets gets delivered as raw media for
rendering.
[[File:Firefox_WebRTC_Pipeline.png|650px]]
With several moving components in the pipeline, it becomes necessary to analyze
the impact these might have on the overall quality of the media being transmitted
or rendered. For instance, the parts of the pipeline highlighted ( marked star)
has potential to induce latency and impact quality of the encoded media. Thus,
having possibilities to measure, analyze and account these impact has potential
to improve the performance of the Firefox WebRTC implementation.
 
Not to forget, the pipeline doesn't capture impacts of latency induced due to
network bandwidth, latency and congestion scenarios.
== Peer Connection Audio Quality ==
Having looked at some background information, let'
[[File:AudioPerf-Setup.png|650px]]
Confirm
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