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Media/WebRTC Audio Perf: Difference between revisions
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## Latency impacts under simulated constrained network conditions | ## Latency impacts under simulated constrained network conditions | ||
#RTCP based analysis | #RTCP based analysis | ||
#Codec Configuration variability analysis | |||
#Hardware and Platform variability analysis | |||
== Typical WebRTC Media Pipeline== | == 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]] | [[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 == | == Peer Connection Audio Quality == | ||
Having looked at some background information, let' | |||
[[File:AudioPerf-Setup.png|650px]] | [[File:AudioPerf-Setup.png|650px]] | ||