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Re: [codec] #8: Sample rates?

"Benjamin M. Schwartz" <bmschwar@fas.harvard.edu> Wed, 14 April 2010 01:05 UTC

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Date: Tue, 13 Apr 2010 21:05:18 -0400
From: "Benjamin M. Schwartz" <bmschwar@fas.harvard.edu>
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To: stephen botzko <stephen.botzko@gmail.com>
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Cc: codec@ietf.org
Subject: Re: [codec] #8: Sample rates?
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stephen botzko wrote:
> I rather like the idea of negotiating maximum audio bandwidth.  For me that
> is different from dynamic complexity management, and is being signaled for a
> different purpose (wasting coded bits on unheard spectrum degrades the
> quality of the heard spectrum).

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.

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.

> 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.

--Ben

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