Hello,
We think about implementating an Audio line up (codecs, echo canceler, noise supressor, etc...) on an OMAP3530 platform for VOIP purpose.
I understand that for video the DSP (IVA2+) should be used because of the amount of data and algorithm processing.
For Audio I am not sure how DSP can provide a strong help to the HOST (ARM) in terms of algorithm computation since the ARM is a quite strong processor (600Mhz).
Actually we would like not to use the DSP bridge (DSPLink) which can simplify the SW architecture (no ARM - DSP communication is needed).
So if we only use the ARM to do the job the architecture will be:
Microphone -> McBsp -> Audio Driver-> Audio lineup -> Wifi - for uplink
Wifi -> Audio lineup -> Audio driver ->McBsp-> Speakerphone - for downlink
Questions:
Our only doubt is that how can we be sure that the ARM will be able to support that process (here a Windows Mobile will be used as an OS)?
Do you know any product that is using this type of architecture?
What does the IVA2+ include in terms of audio processing optimization that is susceptible to help us?
Thanks for the help.
Fabrice.