Re: audioquest dragonfly does not play 88.2 & 96 khz files properly under Ubuntu 12.10

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Hi Daniel and list;

A bit more experimenting this afternoon, please see below.

On Wed, Feb 6, 2013 at 1:20 PM, chris hermansen <clhermansen@xxxxxxxxx> wrote:
> Hello Daniel, list;
>
> Thanks for the reply and the ideas.  I have some more information,
> please see below.
>
> On Wed, Feb 6, 2013 at 8:22 AM, Daniel Mack <zonque@xxxxxxxxx> wrote:
>> On 06.02.2013 17:02, chris hermansen wrote:
>>> On Wed, Feb 6, 2013 at 7:18 AM, Daniel Mack <zonque@xxxxxxxxx> wrote:
>>>>
>>>> Hi,
>>>>
>>>> On 06.02.2013 00:13, chris hermansen wrote:
>>>>> I wonder if any of you have any experience yet with the Audioquest
>>>>> Dragonfly, specifically under Ubuntu 12.10 which is running this kernel
>>>>> GNU/Linux 3.5.0-23-generic x86_64
>>>>>
>>>>> This device sounds fine with 44.1 kHz / 16bit files but the 88.2 and 96
>>>>> kHz / 24bit files do not.  Specifically, I can hear the music for both
>>>>> of those files, but there is a loud kind of "static" thing going in the
>>>>> foreground.
>>>>>
>>>>> A "cat /proc/asound/DragonFly/stream0" when playing at all three bit
>>>>> rates seems to show "reasonable" parameters.  Also, the LEDs on the
>>>>> Dragonfly show the correct color corresponding to the bit rate.
>>>>>
>>>>> One thing I note is that Alsa seems to want to run at 24 bits even for
>>>>> the 16 bit files ie one is forced to use plughw:1,0 for the output.
>>>>
>>>> What is forced exactly? The Linux driver sends data in 24bit if the
>>>> device requests 24bit sample format, and it lets the driver know through
>>>> its descriptors. I wonder why specifing a certain output device should
>>>> change anything in that regard.
>>>
>>> It seems that the combination of driver and Dragonfly only offer S24_3LE,
>>> so a file that is S16_LE needs to be converted to S24_3LE.
>>>
>>> This doesn't seem right to me, as according to this author for instance
>>>
>>> http://www.stereophile.com/content/audioquest-dragonfly-usb-da-converter-measurements
>>
>> I only read briefly over this, but that article doesn't seem to measure
>> the actual samples on the bus. If you tell CoreAudio (or ALSA for that
>> matter) to operate on 16 bits, the software layer will cut off the lower
>> 8 bits and that will of course affect the audio performance in
>> applications. That has nothing to do with actual hardware format spoken
>> to the device). Same happens on Linux when you use plughw:.
>>
>>> the Dragonfly accepts 16 bit data as well as 24 bit data.
>>
>> No, the software layer does, not the device itself.
>
> Since I am nearly completely clueless on the above, I certainly would
> not argue with you!
>
>>
>>> Anyway, the evidence:
>>>
>>> When I try
>>>
>>> sudo aplay -vD hw:1,0 06*.wav
>>>
>>> I get
>>>
>>> Playing WAVE '06_-_Amadou & Mariam_-_Artistiya.wav' : Signed 16 bit
>>> Little Endian, Rate 44100 Hz, Stereo
>>> aplay: set_params:1081: Sample format non available
>>> Available formats:
>>> - S24_3LE
>>>
>>> Conversely, with
>>>
>>> sudo aplay -vD plughw:1,0 06*.wav
>>>
>>> I get
>>>
>>> Playing WAVE '06_-_Amadou & Mariam_-_Artistiya.wav' : Signed 16 bit
>>> Little Endian, Rate 44100 Hz, Stereo
>>> Plug PCM: Linear conversion PCM (S24_3LE)
>>
>> Because ALSA converts that to S24_3LE.
>>
>>> The hw:1,0 device operates with a 24 bit file, giving
>>>
>>> Playing WAVE '2L50SACD_tr1_96k_stereo.wav' : Signed 24 bit Little
>>> Endian in 3bytes, Rate 96000 Hz, Stereo
>>> Hardware PCM card 1 'AudioQuest DragonFly' device 0 subdevice 0
>>
>> And here, no conversion is necessary.
>>
>>>>> When I was debugging my Schiit Bifrost (still not working the way I want
>>>>> FWIW) I recall some patching that Daniel Mack was applying to the kernel
>>>>> in 3.6 (I think).  Perhaps I need a newer kernel...?
>>>>
>>>> Do you see similar behaviour with the DragonFly than with the Bifrost,
>>>> in a way that stopping and restarting the stream would recover it?
>>>
>>> Stopping and restarting the stream does not improve things.
>>>
>>> Also, the LED colour, which indicates the bit rate, is correct on the
>>> first try and does not change on second or subsequent tries.
>>>
>>>>
>>>> Also, did you test the device with Mac OS X maybe, without installing
>>>> any third-party driver?
>>>
>>> I don't have access to a Mac unfortunately.  I will try with a Windows XP
>>> we have here and report back later today.
>>
>> That doesn't help, as Windows does not ship with any usable USB audio
>> driver at all. So vendors are forced to ship their own, proprietary one,
>> which only has to work for their own hardware of course. Consequently,
>> they can ignore all the crucial details in the descriptor and hard-code
>> whatever constants they want in the driver.
>
> I recall reading in some bit of info that this device is a USB-1.0 device.
>
> Does this mean special drivers are not required for Windows?  I think so.
>
>>
>> OS X is different, as they have a fully compliant driver natively. Hence
>> it would be interesting to see whether it works there.
>>
>>>> You can of course, if you're able to, hack the driver and force a
>>>> certain output format, just to see if that stops the static noise for
>>>> you. Then you know where exactly to look for possible misbehaviour of
>>>> the driver. Most likely though, we need to work around a hardware bug
>>>> with a quirk here.
>>>>
>>>> Could you send the output of 'lsusb -v', please?
>>>
>>> http://paste.ubuntu.com/1616929/
>>
>>       AudioStreaming Interface Descriptor:
>>         bLength                20
>>         bDescriptorType        36
>>         bDescriptorSubtype      2 (FORMAT_TYPE)
>>         bFormatType             1 (FORMAT_TYPE_I)
>>         bNrChannels             2
>>         bSubframeSize           3
>>         bBitResolution         24
>>         bSamFreqType            4 Discrete
>>         tSamFreq[ 0]        44100
>>         tSamFreq[ 1]        48000
>>         tSamFreq[ 2]        88200
>>         tSamFreq[ 3]        96000
>>
>> The only audio streaming format offered in the descriptors is 24 bits
>> (bSubframeSize == 3), so the driver does the right thing by sending only
>> that format on the wire.
>>
>> Anyway, I doubt this is the reason for the broken audio on higher sample
>> rates.
>>
>> Lacking a good explanation, the only hint I can give you is to boot a
>> Windows instance in a Virtual box, pass-through that USB device to the
>> guest OS and use usbmon to trace the communication for 96KHz streaming.
>> Then do the same thing with Linux and look for the differences. We might
>> most probably end up with a quirk for that device :-/
>
> I don't have a Windows I can run in a virtual box, so I tried a few other
> things, which may narrow down the problem.
>
> On Windows XP using Foobar 2000 (erm) I get LED colours that match the
> sample rates of the music and NO CRACKLING.  Ie seems to work there.
>
> Because I can, I guess, I tried the Dragonfly on a couple of 32 bit Ubuntu
> machines I have around, one a server 12.10 and one a fresh desktop install
> of 12.10.
>
> Both of those work just fine!  Ie I can play 16 bit / 44.1 files through plughw
> and 24 bit / 88.2 and 24 bit / 96 through hw or plughw and I get fine sound,
> what appears to be the correct behaviour from aplay -v, and the colours of
> LEDs that are expected.
>
> So something is either weirdly configured on my 64 bit machine or ...?
>
> One other thing I have tried on the 64 bit machine - using sound settings
> to make the Dragonfly the active card and playing Guayadeque through
> the default device, I get 44.1 files playing at that bit rate (according to
> the colour of the LED on the Dragonfly), and 96 files playing at 48 kHz
> (according to the colour of the LED on the Dragonfly).
>
> Neither the 44.1 nor the 48 music includes static.
>
> The above behaviour seems to make sense, as Pulse is configured to
> use 44.1 as the default sample rate and 48 as the alternative sample rate.
>
> So I am left wondering if I have some kind of weird configuration issue
> where pulseaudio is somehow interfering with the Dragonfly, or some
> similar thing.
>
> Thanks again for any ideas.

Further to my comments above, my "other project" is to get my Schiit Bifrost
working properly.  Today I tried a brand-new ASUS DX in my older Dell desktop
hooked up to the Bifrost with a TOSLINK "cable".

What does this have to do with the Dragonfly etc above?

Well, the ASUS DX -> Bifrost on my Dell (32 bit Ubuntu 12.10) behaves the
same weird way as the Dragonfly on my System76 (64 bit Ubuntu 12.10), ie
fine on 44.1 / 16 bit files but weird loud static on the 96 / 24 bit files.

All the weirder because the Dell + Dragonfly seems to work perfectly.

Now more puzzled than ever.  Could this be Pulse Audio weirdness?



-- 
Chris Hermansen · clhermansen@xxxxxxxxx

C'est ma façon de parler.

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