Re: [codec] draft test and processing plan for the IETF Codec

[Jean-Marc Valin <[e-mail]>]

On 11-04-18 04:56 PM, David Virette wrote:
> Hi Jean-Marc,
> I agree that a well-designed VoIP application should ideally use the
> available bandwidth and I think that in the current test plan, the maximum
> available bandwidth will always be taken into account. For super-wideband
> mode, the input will be 32 kHz, then low pass filtered at 14 kHz and then
> probably resampled at 24 kHz for OPUS allowing it to use its maximum
> bandwidth for those modes.

Well, if you low-pass filter at 14 kHz then by definition you're not using 
all the available bandwidth. If what you have is a 32 kHz input signal, 
then the best thing you can do is resample it to *48* kHz (without any 
low-pass filtering) and then feed it to Opus at that rate. You can consider 
Opus as a 48 kHz codec that can also do lower bandwidths. In fact, the best 
way to use Opus is to always run it at 48 kHz and to let it scale bandwidth 
internally based on the bit-rate. A while ago I posted this sample of Opus 
scaling from 8 kb/s narrowband to 60 kb/s fullband: 
http://jmvalin.ca/misc_stuff/opus_sweep.wav

Cheers,

	Jean-Marc