Linux Audio the way to go!?

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#824
Hi there.

If there are people around using EMU 0404 USB I'd like to share a hint.

As you know it just supports 16/44.1 by default.

However, you can set the samplerate to e.g 96k under Windows, reboot into
Linux. Don't reset the EMU in between. This way you can run 24/96 under Linux.

As said earlier. MPD supports 24bits with latest git release. Give a manual compilation
and install a try.

Now you can use libsample to 96 in best sinc within MPD to convert your other stuff to
have fun until you turn the EMU off.

Cheers
 
#826
The future of brutefir ?

Hi folks.

A quick poll in between:

I was thinking for how long Brutefir is gonna remain a top notch
convolver.
As you might know Brutefir is not what you'd called well maintained
since quite some years by now.
I think it falls short on certain aspects already today.

What I would like to put on my Christmas wish list would be:

1. APIs need to be updated
2. Up2Date Dither algorithm (e.g. http://www.apogeedigital.com/pdf/UV22osquick.pdf)
3. Cuda or GPU engine support (several times faster than processorbase processing)
4. >SSE2 support
5. Adaptation to current scheduling algorithms
6. Optional: OSS 4Front IO module

Let me know what you think of it and if you have more wishes to add or
comments on above.

The next step would be to think of how to get started on tackling the
issues.

Cheers
Klaus
 
#827
Hello Soundcheck,

because I've lost tracking your hardware configuration could you please write a short description of your set-up at the current date? I know it's USB based and I'm trying to do the same. Only if I could find an asynchronus mode USB chip...
 
#829
Thank you soundcheck!


Hope I am not thrashing the thread too much but do you have a link with a description of the ec-design usb interface? I can't find much info on his site and the thread here on diyaudio is enormous.

What are the particularities that made you choose this usb receiver? I don't think it's asynchronous like the wavelenght dac, so how does it deal with jitter?

At the moment the only fast solution I see for a quick try is pcm2707 -> ASRC chip -> DAC or an all in one solutin with the ess chip having integrated asrc (and a good one I've heard).
 
#830
@sunra

I'd recommend that you have look at the Sabre DAC. That's IMO the way forward.
Though - the interface question remains of course. I feed mine via I2S from a DDDAC USB receiver without reclocker (because Sabre needs 64fs on the I2S bus) for the time being.
This limits me also to 44.1/48/16 of course. The good thing though - I can fully enjoy my
Linux USB tweaks.

Cheers
 
#832
Hi folks.

One more for today.

I introduced a Wiki about format conversions and re-/upsampling.
It is not yet complete, but at least it is starting point. As usual WIP ( work in progress)

http://www.diyaudio.com/wiki/index.php?page=LINUX+Audio+Formats

I'd apprectiate some input/feedback of you guys!

While checking things out, I figured that SOX seems to be even better
performing then libsamplerate ( 175db noise rej. vs 145db) - more in the Wiki.

The latest SOX (14.2) is a real killer machine. It has to be compiled from sources
though.


BTW: Above mentioned flacs from HDtracks will sound even better if you run a flac -d <yourflac> first. Then you put the wav on /dev/shm and play it from there.
I also compared my original SOX-VS upsampled from 44.1->96 and the hires version.
You might guess what's been more then the obvious result. 😉

Cheers
 
#834
Telstar said:
Try upsampling to files instead of doing it on the fly and you will see 🙂
Click to expand...


Go approx. two years back in this thread and you'll find somebody pointing exactly that out! Probably two posts further down during that time you'll find somebody saying that flacs shouldn't be uncompressed on the fly. 😉


BTW I also compared the "offline" upsampled (44-96) Marta Gomez version earlier. Can
beat the original from HDtracks. Though I am not sure if the material from HDtracks is master stuff or also downsampled from 192 or even higher. I doubt that Chesky records 96k.
For sure this would be a nice busines case for the future, if they come up in two years time with 192k downloads. 😉
 
#835
soundcheck said:


Go approx. two years back in this thread and you'll find somebody pointing exactly that out! Probably two posts further down during that time you'll find somebody saying that flacs shouldn't be uncompressed on the fly. 😉

BTW I also compared the "offline" upsampled (44-96) Marta Gomez version earlier. Can
beat the original from HDtracks. Though I am not sure if the material from HDtracks is master stuff or also downsampled from 192 or even higher. I doubt that Chesky records 96k.
For sure this would be a nice busines case for the future, if they come up in two years time with 192k downloads. 😉
Click to expand...

I believe most masters are at 24/96, not higher. 192k should come the next year i believe.

If you want to do a good test, use SOX with the following parameters (assuming you have a 16/44.1 redbook source):

sox input.wav -b 24 output.wav rate -v -s 192k

(176.4 would be better, if your dac does not resample it again)
The ratio behind this is not really to improve the sound quality, but to avoid the DAC to do its crappy oversampling.

I did the above test compared with the secret rabbit code and otachan, which are both considered the best realtime resamplers, and the offline wins.

Of course i'm not using flac, but plain simple pcm wav.
SOX is open source and works under linuz.
 
#836
@sunra I'd recommend that you have look at the Sabre DAC. That's IMO the way forward. Though - the interface question remains of course. I feed mine via I2S from a DDDAC USB receiver without reclocker (because Sabre needs 64fs on the I2S bus) for the time being. This limits me also to 44.1/48/16 of course. The good thing though - I can fully enjoy my Linux USB tweaks. Cheers
Click to expand...

Thank you for the reply. I've seen in your system description at AA that you are using EC-design usb-to I2S. Are you still using that module or you are preferring the DDDAC USB receiver? Something wrong with ec-designs re-clocker?
 
#837
One of my favorite albums : Marta Gomez - Entre cada palada - - now at 24/96 -- sounds awesome I can't listen anymore to the realtime upsampled (mpd) version ( 44.1->96).
Click to expand...

Me again,

probably I've arrived late to this discussion but how does actually influence re-sampling (or playing files sampled to more than 44.1K, 16bit) when dddac can't stream more than 44.1k? In fact it's a problem with any usb interface except the one from wavelength audio.

Do you get such a significant change in audio quality?

edit: that's one of my favourite albums too 🙂
 
#838
Since sox offers probably the best quality now, why not using it online, instead of offline processing? It is rather simple:

.asoundrc:

Code: 
pcm.upsample44 {
        type plug
        slave {
                pcm raw44
                format S32_LE
                channels 2
                rate 44100
        }
}
pcm.upsample48 {
        type plug
        slave {
                pcm raw48
                format S32_LE
                channels 2
                rate 48000
        }
}
pcm.raw44 {
         type file
         slave {
                 pcm null
         }
         format "raw"
         file /home/pavel/fifo44.raw
        
}
pcm.raw48 {
         type file
         slave {
                 pcm null
         }
         format "raw"
         file /home/pavel/fifo48.raw
}

eternal sox for 44100Hz audio:
Code: 
while true; sox -V -t raw  -r 44100 -c 2 -s -4 fifo44.raw  -4 -t alsa hw:0 rate -v 176400; done

eternal sox for 48000Hz audio:
Code: 
while true; sox -V -t raw  -r 48000 -c 2 -s -4 fifo48.raw  -4 -t alsa hw:0 rate -v 192000; done

Fortunately SoX does not open the alsa output device until it starts receiving input data, which allows for several instances running idle concurrently.

Now you can play CD audio via upsample44 device upsampled to 176400Hz by best sox algorithm, and 48kHz audio via upsample48 device upsampled to 192000Hz. Just make the fifos and start the eternal soxes in advance.

Actually adding libsox to the alsa rate plugin types would be a much nicer solution 🙂
 
#839
@ phofman

Just got back and saw you post. cool stuff! 😉

Where do you put your loops? Did you check out processor usage, when using -v -s for online sampling?


@ sunra

I am enjoying old fashioned 44.1 😉 sound on my main gear. At my desk I am playing with 96k
and EMU.
The 1543 DACS I'am feeding via ec-designs USB and Sabre via DDDAC USB receiver, since I2S was better. SPDIF was taken from EMU USB 0404.

Next action: I need to find a good hires I2S input to the Sabre. Of course minimum 6 channels - and all in sync.
 
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