Pavel: Thanks for the insight. I thought I was doing something wrong since no one else has noticed. I tried your suggestion to use plughw:0,1 and the verbose output looks fine, the data rate shifts fine (I'm watching the L/R clock on the input to the DAC on an external DAC) but no audible sound comes out the converter unless it's a 16 bit 44.1 file. The 24 bit stuff is not getting converted into something useful to the DAC. Should I check to see if the data clock is 48X the LR clock?

What info do you need from me to look deeper?

I can supply a sample clip of a 176.4 KHz recording from Reference Recordings if it would be helpful. I'm also testing with the samples from Linn and Gimmel.

I looked at the Plug plugin but have not been able to figure out how to configure it for my needs. There is way too much marginal information on the web and little that directs me to an optimum setting. -Demian

From: Pavel Hofman pavel.hofman@insite.cz Subject: Re: [alsa-devel] Via VT82xx VT1708 ICE1724HT and higher sample rate/bit depth files through SPDIF To: Demian Martin demianm_1@yahoo.com Cc: alsa-devel@alsa-project.org Message-ID: 47C423E4.4060900@insite.cz Content-Type: text/plain; charset=windows-1252; format=flowed

Hi Martin,

First- sample rate

Even though the documents claim that it will play at 176.4 KHz and 192KHz whenever I play a file at these rates it down converts to ? the rate

(88200

and 96000). I tested this with the ICE1724HT (Juli@) which has almost the same core, I never could get 176.4KHz at the SPDIF output. I confirmed

that

the ICE1724 will play the file at 176.4KHz in Windows so its not a

hardware

limitation. I tried to do the same with the VT1708 (on a Via CX700

chipset)

and the Windows driver wouldn?t let me at anything other than 48000.

I spent a few days on this issue and confirm that ICE1724HT cannot output 176.4kHz from its internal SPDIF transmitter, WHEN clocked by its internal clock circuits. A scope hooked to corresponding pins/outputs revealed the following:

All I2S lines output correct 176.4kHz signal, i.e. D/A codecs receive proper signal.

However, I did not find a way to force ICE1724 to output 176.4kHz, it always drops to 88.2kHz, no matter what MT01/MT02 combination is. The same happens with windows drivers of my Prodigy 192. The sequence is: 44.1, 48, 88.2, 96, 88.2, 192

I have not seen any manufacturer of Envy-based cards revealing this deficiency. The "preliminary" datasheet we all use mentions only that the chip forces the 128x mode when switched to 176.4kHz. I believe the output of 88.2kHz is an undocumented feature or perhaps even a bug of the chip.

Juli is a card whose manufacturer may be aware of the problem. It is not clocked by ICE1724 internal clocks fed by the two crystals. Instead, the crystals also provide signal to external circuits generating clock signals of required frequency, feeding the chip in slave mode. The circuitry is controlled by ice1724 GPIOs.

This way the ICE1724 is not aware of its clocking frequency and cannot thus mis-configure its SPDIF transmitter. The windows driver does output proper 176.4kHz.

I am working on Juli driver and have been analyzing this beast for a few nights. The special clocking is already working at home :)

Second, the only way I can get the 24 bit files out in a form that the

card

can play is to pass them through dmix. But I need to reconfigure dmix for each sample rate. And I?m not convinced that it isn?t resampling twice,

both

at the input and the output. Or how to configure the buffers etc. to

prevent

underruns even when the only conversion is to pass a 24 bit file to the card.

ICE cards use 32bits format. Try aplay -v -D plughw:0 24bit.wav and you will see the conversion to 32 bits done by the plug plugin, no dmix involved.

I?m trying to build a high quality music server that will pass the files

to

either the internal or an external DAC at the bit depth and sample rate

they

were recorded at. I have tried to understand the various plugs and

settings

but it seems they are all focused on mixing various streams. I want to

send

the files unmixed to the output. I can?t figure out what set of plugs and settings in asound.conf would enable this.

Check the plug plugin.

Regards,

Pavel Hofman. -----------------