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Old 20th July 2008, 09:09 AM   #581
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Hi folks.

I am playing around with a M-Audio Transit. I am using the Toslink interface towards a TAS5706 based full-digital amplifier, which I am currently testing.

I have also written a small how-to to get the Transit working under Linux. You'll find it in the Wiki. (I btw already put some effort into updating all Wiki pages. It's not finished yet. But it is getting better every day.)

Now: The M-Audio driver is working fine so far at 44.1/16 .

But how do I control the card? E.g. to set latencies, buffers sample-rates etc. I need some kind of host application or command-
set to set resp. tweak the parameters.

Does somebody have a clue how to tackle this issue?

Cheers
\Klaus
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Old 22nd July 2008, 08:24 AM   #582
phofman is offline phofman  Czech Republic
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I do not think there are special controls for M-Audio USB cards in Alsa. Concerning the buffers, I would assume M-Audio cards use the same facilities as other regular USB cards - your field of specialization

Take a loot at the source files for m-audio usb cards:

http://git.alsa-project.org/?p=alsa-...357e8e;hb=HEAD

http://git.alsa-project.org/?p=alsa-...934e40;hb=HEAD
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Old 22nd July 2008, 08:58 AM   #583
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I looked them up. I've been in there before. The quirks seem to be the entry into the config.
(I've implemented a very basic quirk for the Benchmark DAC1 before and it even worked).

However. I am not really aware how these work in general and how they can be used.
I would need the internal address and parameter structures of
the M-Audio Transit first - right? Then I'd need an application to set these? So it is not that easy. It shouldn't be too difficult to send a plain Ascii string over the interface ( I'd need to figure that out, too)


I need to read Takashi Iwai's paper on driver devlopment.
Now I am entering the depth of driver development.

Perhaps that's what I should do anyhow.

Cheers
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Old 14th August 2008, 12:35 PM   #584
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does anyone know if this can be implemented to Linux platform?


http://www.audioasylum.com/forums/pc...s/3/32581.html
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Old 14th August 2008, 01:27 PM   #585
phofman is offline phofman  Czech Republic
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Quote:
Originally posted by ackcheng
does anyone know if this can be implemented to Linux platform?
http://www.audioasylum.com/forums/pc...s/3/32581.html
I do not see anything special. ASIO = alsa, the secret rabbit upsampling library is multiplatform, can be used as plugin in the built-in alsa resampler (e.g. dmix).

http://64.233.183.104/search?q=cache...s&ct=clnk&cd=5

http://blog.flameeyes.eu/articles/20...alsa-hda-users

Individual parameters of libsamplerate could be tweaked in code or perhaps in the config files.

Ad AWE - using 64-bit system avoids the 3GB memory limit per linux process (2GB in windows)
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Old 14th August 2008, 03:36 PM   #586
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correct me if I am wrong but my understanding is that AWE allows a certain part of RAM dedicated for audio. This way, this physical part of RAM is untouched by other program and can be sure that the signal path from the RAM to the sound card is the same every single time? If we just create a Ram disk, this can occupy different part of the physical RAM everytime we load the programs.

This is similar to someone who claims that placing music files at different folders in the same hard disk actually sound different?
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Old 14th August 2008, 03:40 PM   #587
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The Windows policy Lock Pages in Memory option is disabled by default. This privilege must be enabled to configure Address Windowing Extensions (AWE). This policy determines which accounts can use a process to keep data in physical memory, preventing the system from paging the data to virtual memory on disk.
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Old 14th August 2008, 11:43 PM   #588
dmills is offline dmills  United Kingdom
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That sounds similar to using something like

@audio - memlock 512000
@audio - rtprio 25

in /etc/security/limits.conf to allow members of the audio group to use
memlock (MCL_CURRENT | MCL_FUTURE); to lock pages into RAM.

The second line above allows members of group audio to use the posix realtime scheduling to effectively opt out of normal process scheduling and preemption.

If you want a fixed location for a buffer in physical ram (Why?), you can pass the kernel mem=... to restrict the kernel to using the appropriate amount of memory, then access the rest via /dev/kmem or the like for your buffer. Some very fast data acquisition cards had to play this game back in the day, but it is vast overkill for audio type stuff.

I don't see any problem with writing something like the above project, except that I don't really see the point.

Regards, Dan.
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Old 30th August 2008, 04:20 AM   #589
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Do you say that all 270x based chips resample internally, even for USB > I2S data, or only when used as a DAC?

There are a lot of DAC's around which use PCM2706/7 for USB > I2S which would mean, that all of them are lossy when used as USB?

It's bad that the more I dig into perfect DAC's I find no way out. At the moment, I think the only possibilities are:
- Asus EEE PC with highly modded linux kernel + USB DAC
- Chinese I2S port on Zhoulu D3
- some experimental DAC, like ethernet DAC + mpd
- CrazyT from BD-Design, have a look, they really seem to understand what they do and they developed their own software for sending data to the DAC (like ethernet DAC), but it's not a DIY budget...
http://www.bd-design.nl/contents/en-us/d168.html
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Old 30th August 2008, 07:54 AM   #590
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Hi folks.

The only way to disconnect the DAC from the PC jitter mess is to run
asynchronous over any kind of communication media.
The receiver controls the flow and buffers the stream in this case.
If you look at the ethernet boxes a la Squeezebox - we do not talk about realtime datastreams any longer, they buffer the stream and the FPGA is kind of resending it. You won't face any bit-error issues on the ethernet.

This you'll find implemented on e.g. EMU0404, Wavelength Async DACs for USB or Squeezebox ( or similar) for ethernet equipment.

If you look at BD-Design. Bert justed jumped on that train.
Buffering the datastream in a kind of async mode is nothing new.
The BD-design solution is a propriatary solution. You need to have
a special BD-Design player.

If you run such an ethernet box you won't hear any changes done on the PC. Of course the issues are now moved further down the stream.
But for sure it is easier to solve these down there.


But there are certain shortcomings:

Most implementations are 44.1 only.
High latency.
2 channel only

I am just playing around to build a streaming client with MusicPlayerdaemon and PulseAudio.

I also just connected a EMU USB 0404 to see how this works.
( You need to update it's firmare via Windows first. Makes the 0404 play 44.1 by default. Of course you're limited by 44.1 for the time being.)

BTW: I just finished my desktop rig: The new Adam A5 speakers (with stands) and the EMU 0404 USB. Now I spend even more hours at the PC. The sound is just great.

Cheers
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