[alsa-devel] Juli@ ICE1724 and 24 bit audio

Pavel Hofman pavel.hofman at insite.cz
Tue Feb 10 16:34:58 CET 2009


Vedran Miletić wrote:
> Yeah, ALSA's documentation is actually quite lacking on many fronts
> and is mostly scattered around various wikis and mailing lists.
> 
> But I guess it is as it is, complaining about it won't fix it.
> 
> On 2/9/09, Demian Martin <demianm_1 at yahoo.com> wrote:
>> Thanks, I'll try that. I guess this e-mail will serve as more documentation
>> on that feature (bug). I had not seen that factoid anywhere.
>>
>> Demian Martin
>> PDS
>>
>>
>>> -----Original Message-----
>>> From: Vedran Miletić [mailto:rivanvx at gmail.com]
>>> Sent: Monday, February 09, 2009 1:39 PM
>>> To: Demian Martin
>>> Cc: Pavel Hofman; alsa-devel at alsa-project.org
>>> Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio
>>>
>>> The plughw part is what makese it resample to 16-bit. You should just use
>>> hw.
>>>
>>> On Mon, Feb 9, 2009 at 10:29 PM, Demian Martin <demianm_1 at yahoo.com>
>>> wrote:
>>>> Pavel:
>>>> Thanks for all your help and work on this project.
>>>>
>>>> I'm using the Juli@ card because it supports 176.4KHz. The only other
>>>> candidates are much more expensive and bring their own problems.
>>>>
>>>> I'm using Xine set for dolby/dts passthrough which seems to send the
>>> audio
>>>> data at its native rate. The resampling is bypassed (Dolby Digital would
>>> be
>>>> trashed otherwise).
>>>>
>>>> I will try playing the files from the command line when I am next in
>>> front
>>>> of the system, tomorrow.
>>>>
>>>> This is the Asound.conf:
>>>> Asound.conf  (/etc)
>>>>
>>>> pcm.asym_spdif {
>>>>
>>>>       type asym
>>>>
>>>>       playback.pcm "plughw:0,1"
>>>>
>>>>       capture.pcm "plughw:0"
>>>>
>>>> }
>>>>
>>>>
>>>> pcm.!default asym_spdif
>>>>
>>>> Demian Martin
>>>> PDS
>>>>
>>>>> -----Original Message-----
>>>>> From: Pavel Hofman [mailto:pavel.hofman at insite.cz]
>>>>> Sent: Monday, February 09, 2009 12:27 PM
>>>>> To: Demian Martin
>>>>> Cc: alsa-devel at alsa-project.org; pavel.hofman at insite.cz
>>>>> Subject: Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio
>>>>>
>>>>> Demian Martin napsal(a):
>>>>>> I have the driver working fine for sample rates from 44.1K to 196K
>>>>> however
>>>>>> it seems to be truncating the data to 16 bits. I wasn't sure until I
>>>>> started
>>>>>> testing with the HRx 176.4K 24 bit files that have an embedded HDCD
>>> flag
>>>>> in
>>>>>> the LSB. The files work OK on Windoze systems (with a lot of low
>>> level
>>>>>> settings tweaked) and they play fine but the flag isn't detected by a
>>>>> system
>>>>>> that can detect them. Further looking at the data stream with a scope
>>> it
>>>>>> seems the last 8 bits aren't changing. Is there anything I can do to
>>>>> control
>>>>>> the driver to confirm this problem or change the playback settings to
>>>>> make
>>>>>> it work?
>>>>>>
>>>>>>
>>>>> How do you play the files? Do you use the hw or plug:hw device? IIRC,
>>>>> the standard-setup dmix resampler is 16-bit only.
>>>>>
>>>>> I did tests with SPDIF OUT/IN and found it bit-perfect, for 24bit too.
>>>>>
>>>>> Regular ICE1724 cards do not output 176.4kHz SPDIF, only 88.2kHz. But
>>>>> Juli is not affected by that bug.
>>>>>
>>>>> Regards,
>>>>>
>>>>>
>>>>> Pavel.
>>>> _______________________________________________
>>>> Alsa-devel mailing list
>>>> Alsa-devel at alsa-project.org
>>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>>>>
>>>
>>>
>>> --
>>> Vedran Miletić
>>

What makes you guys think the plug plugin trims the data down to 16bits? 
  Sure if the rate is changed (the source code suggests only the 
low-quality linear rate algorithm supports above-16bit format - 
operation convert, the other resampling algorithms convert using the 
operation convert_s16).

Ice1724 supports all the general rates natively, the rate conversion in 
the plug plugin should not kick in for common audio formats. The only 
case I can think of would be switching Juli to external SPDIF-IN clock - 
the hw.rate_min = hw.rate_max = actual_rate and the automatic rate 
conversion could happen.

I did not use the hw device in my tests since all the tested tracks 
would have to be 32-bit for the hw device to accept the format when 
played via my favorite aplay.

My 2 cents guess is xine does the conversion.

Regards,

Pavel.



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