[linux-audio-user] Ardour and DAT

ISh ish at sarai.net
Thu Jul 21 02:33:01 EDT 2005


Thanks a lot Steve your advice is very helpful :) 

ISh

Steve Harris wrote:

>On Wed, Jul 20, 2005 at 10:37:48 +0530, ISh wrote:
>  
>
>>1. Now a lot of field recordings and Music in on Cd (16bit ,44Khz) 
>>format. And I am working with 48Khz ,16bit on the jackd(deamon) and on 
>>ardour. Now if I oversample/ ?resample at a higher rate? in ardour by 
>>importing it in the 'Auditions' region i.e. basically from 44Khz to 
>>48Khz. Will I loose out on quality. Is the Ardour re-sampler good enough 
>>to handle this conversion.
>>    
>>
>
>Will they definatly only accept 48k DAT? I was under the impression that
>DATs are now often 44.1k as well. OTOH digital radio (in the UK at
>least) is 48k, so it seems reasonable.
>
>The Ardour resmapler (SRC) is pretty good as long as you select a decent
>sinc interpolation. You will loose qlauilty, but I doubt if you will hear
>the difference.
> 
>  
>
>>2. Or, should I work with 16bit-44Khz setting (on jackd/Ardour) and 
>>export the final mixed session once finished to 16bit-48Khz(DAT) . So I 
>>can take the dump of the final session on DAT tapes. Will this way be 
>>better than the point(1) above in which all the audio is individually 
>>re-sampled at
>>    
>>
>
>Resampling can be quite time consuming, so forthat reason you might want
>to convert at the end, as its easier to batch process it, and you may end
>up converting less material. It wont make much difference to the quality.
> 
>  
>
>>3. I have a 'Tascam DA-P 1' DAT recorder on which I have to dump the 
>>final edit. So how do I go about this. Should I play the audio in Ardour 
>>only and the send it to the DAT recorder. Or should i play the exported 
>>file in a player like Alsaplayer/ Audacity(which plays audio rec at 
>>48Khz ???) and then take the dump through SPDIF out into the DAT 
>>recorder. Is this possible to do the above mentioned process(s) any 
>>other way, which might be better.
>>    
>>
>
>The choice is between sending it over SPDIF at 48k (involving a resampling
>stage), or over analogue (hopefully balanced) I guess? It depends on your
>feelings about analogue v's digital noise, personally I would go with
>SPDIF. The app that you use to send the audio out over shouldn't make any
>difference, but make sure your buffers are large.
> 
>  
>
>>Basically Im dealing with a lot of audio at 44KHz which has to mixed 
>>together and the final output will be on 48Khz(DAT Tape). What could be 
>>the best way to deal with it so that there is no or very little loss in 
>>quality. Any help will be greatly appreciated.
>>    
>>
>
>I would say that a high quality sinc resampler will be the best quality.
>You shouldn't be able to distinguish th resampled audio from the original,
>but you sould experiemnt, of course.
>
>- Steve
>  
>




More information about the Linux-audio-user mailing list