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MDAC First Listen (part 00110011)

Discussion in 'audio' started by adamdea, Nov 6, 2016.

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  1. JohnW

    JohnW pfm member


    No problem to power On/Off via a switched power strip.
  2. JohnW

    JohnW pfm member

    Does not include the SOC power requirements.

    Thank you for the offer, but I'll need to source solution for the MDAC2 PSU - we have some ideas...
  3. adamdea

    adamdea You are not a sound quality evaluation device

    Some cds are now mastered in a way that the uadiophile market does not like (largely down to dynamic range compression). It seems that at least some people like this though.

    It doesn't affect most of the music I listen to

    FWIW I personally do not buy into the idea that vinyl is the gold standard or that there is anything wrong with 16/44 pcm. Perhaps vinyl replay has improved but there are huge flaws with the medium as whole: for example the hole in the middle of the record (pretty much all records) being off centre creates timing inaccuracies hugely greater than the tiny errors arising from jitter or anti/alias/ anti imaging filters. Even test records cant avoid this which is why wow and flutter figures are often given with a low pass filter. This may be fair enough when comparing turntables, but not IMHO when considering the medium itself
  4. mrflibble

    mrflibble pfm Member

    I agree, so many recordings are spoilt, even completely ruined, by stupidly excessive compression. RHCP's Californication and Metallica's Death Magnetic are infamous examples.

    I enjoy the composition of some popular music (Iggy Azalea, David Guetta), but I can't listen to it because of the horrible distortion on the vocals. Such a shame. It can be heard quite clearly on my setup even when listening to the Spotify and YouTube versions - for example Iggy Azalea's track "Black Widow".

    And vinyl isn't always any better because it is often (not always) mastered to a similar level. When the vinyl is mastered correctly and the CD is not, then the vinyl will sound better because of the mastering not because it is on vinyl. The vinyl may or may not sound better for other reasons, however (depending on your point of view).
  5. XpanD

    XpanD New Member

    Thanks! Definitely going to try this soon, should save a good bit of time. :)
  6. adamdea

    adamdea You are not a sound quality evaluation device

    upsampling to dsd should take the dacs anti imaging filter out of the equation and maybe the modulators, but if the dac is not converting in exactly the same format who knows what's going on? It might end up being converted back to pcm or remodulated.
  7. Julf

    Julf Facts are our friends

    Except vinyl can't actually physically take the same kind of ultra-loud, ultra-compressed mastering we see on many CDs.
  8. mrflibble

    mrflibble pfm Member

    Whilst it doesn't fix the problem of the decimation filters in the delta-sigma ADC used during the recording process, the use of a Non-oversampling DAC or a multibit DAC should preserve the samples present in the digital signal. These types of DACs tend to sound more analogue like. For those interested, it may be worth trying a cheap NOS or multibit DAC to hear if it is to their taste.

    There is a chap in France who makes little TDA1543 NOS DACs and sells them on ebay, searching for "nos tda1543" on ebay will bring them up. They are a bit lacking in resolution but are easy to listen to.
  9. mrflibble

    mrflibble pfm Member

    Yes, you are right. But I believe the vinyl of Death Magnetic was still pretty bad?

    If the original recording is performed digitally with an ADC, then there will a further conversion back to analogue to record the signal onto vinyl. The vinyl would likely exhibit many of the issues that the CD has - the difference being the digital to analogue conversion has happened at the production phase rather than the playback phase. Of course, in the good old days, I believe master recordings were made to analogue tape and the vinyl was cut from these.
  10. Julf

    Julf Facts are our friends

    Indeed, but thos good old days are long gone - these days pretty much all recordings are recorded, mixed and mastered digitally.
  11. 44000khz

    44000khz Active Member

    Hello JohnW,

    I've tried to send a private message but your inbox is full.
  12. Richard Kimber

    Richard Kimber pfm Member

    Has this changed? I thought you said earlier that MDAC2 was DSD256. I wish there was a definitive list of MDAC2 specs available.

    - Richard
  13. JohnW

    JohnW pfm member


    We have DSD512 working on the bench - but no OS / Software can support such a data rate - I dont recall how we tested it.

    Basically the hardware can support DSD512, but the PC software / OS will take time to catch up.
  14. jirij

    jirij Virtual Member

    It's one more step up - DSD512 is commercially available, DSD1024 is possible in "lab conditions". Google gives me ie. http://www.diyinhk.com/shop/audio-k...to-i2sdsd-pcb.html#/xmos_option-xmos_768k_pcb amongst the other high-end DACs.

    Driver wise, I don't see any rates hardcoded for native DSD on Linux, so it can theoretically support anything the (USB) device announces. That is, as long as it fits ie. 32bit signed integers, I assume some code uses those for Hz. Still, that gives you some ~2GHz sample rate to work with, though you might want to use USB 3.1 for that.

    edit: Another cheap one - http://www.ebay.com/itm/Amanero-USB...port-DSD-512-for-LKS-ES9018-DAC-/141316861730 ... somebody should buy one and figure out how it plugs into the linux/sound subsystem. :) (Might do it after xmas.) I remember looking at how they implemented native DSD and it was along the lines of defining it as a single-bit PCM or something like that. As the kernel just passes samples around, it presumably works fine.
    ./uapi/sound/asound.h:#define   SNDRV_PCM_FORMAT_DSD_U8         ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */
    ./uapi/sound/asound.h:#define   SNDRV_PCM_FORMAT_DSD_U16_LE     ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */
    ./uapi/sound/asound.h:#define   SNDRV_PCM_FORMAT_DSD_U32_LE     ((__force snd_pcm_format_t) 50) /* DSD, 4-byte samples DSD (x32), little endian */
    ./uapi/sound/asound.h:#define   SNDRV_PCM_FORMAT_DSD_U16_BE     ((__force snd_pcm_format_t) 51) /* DSD, 2-byte samples DSD (x16), big endian */
    ./uapi/sound/asound.h:#define   SNDRV_PCM_FORMAT_DSD_U32_BE     ((__force snd_pcm_format_t) 52) /* DSD, 4-byte samples DSD (x32), big endian */
    Hmm, looks like it's just 2/4 byte blocks. Reminds me of DSF/DFF which really just contains blocks of DSD bits. :)
  15. JohnW

    JohnW pfm member


    Its just my opinion, but I dont consider Linux a typical everyday users OS... its too specialised - you need to be a computer geek to use it :)

    I'm getting too old to want to spend any more of my life fighting computers.. it was ok when I was 15 years old with nothing better to do with my life!

    Jarek had DSD512 working here in the lab and hes the kind of guy that likes to mess with such detractions :)
  16. Richard Kimber

    Richard Kimber pfm Member

    The new Matrix Audio X-SABRE Pro claims to be capable of 1024

    - Richard.
  17. Richard Kimber

    Richard Kimber pfm Member

  18. JohnW

    JohnW pfm member

    Its not really a hardware issue - but the Host software, we have no means to test above DSD512 but when there is a solution then there's no reason we cannot support it.

    I'm not aware of any "true" DSD1024 capable ADC's so at the moment the feature is rather moot.
  19. misterdog

    misterdog Not the canine kind

    + 1 billion or more.
  20. Julf

    Julf Facts are our friends

    There are linux distros that are perfect for everyday users. It's just that the distros you want to pick as a base for a special embedded application are of course geared towards experts.
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