Re: [codec] #8: Sample rates?

Koen Vos <koen.vos@skype.net> Wed, 14 April 2010 01:36 UTC

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Message-ID: <20100413183602.86565rmv5hve5d6q@mail.skype.net>
Date: Tue, 13 Apr 2010 18:36:02 -0700
From: Koen Vos <koen.vos@skype.net>
To: bens@alum.mit.edu, "Benjamin M. Schwartz" <bmschwar@fas.harvard.edu>
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Cc: codec@ietf.org, stephen botzko <stephen.botzko@gmail.com>
Subject: Re: [codec] #8: Sample rates?
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Quoting "Benjamin M. Schwartz":
> 1. Why would high frequencies be unheard?  Cheap speakers and microphones
> have difficulties with low frequencies, but not high frequencies, and
> routinely go all the way up past the limit of hearing.

Not all hardware supports arbitrary/high sampling rates.  PSTN  
gateways don't go above 8 kHz.  Same for some mobile devices.


> 2. Why would it need to be negotiated?  For a suitably designed format,
> the encoder could choose not to waste bits on high frequencies without any
> negotiation or extra signalling.

Without signaling, how would the encoder know that the farend decoder  
will not take advantage of frequencies above a certain threshold?


>
>> Signaling the bandwidth, and defining the
>> internal codec rate as fullband should let us lock down the RTP timestamp
>> rate at 48 kHz (which I think is desirable).
>
> I do agree that having "only one mode" would be ideal, to maximize
> interoperability.  I wonder whether we can achieve high enough
> computational efficiency for this to be viable.

Changing the RTP timestamp sampling rate causes no computational  
complexity, does it?  Perhaps an extra multiplication for each packet  
or so?  The point was that RTP timestamp sampling rate should  
disconnected from the actual audio signals.

koen.