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ian.au
Guru
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Joined: 07 Apr 2011
Posts: 508
Location: Australia

PostPosted: Tue Oct 15, 2019 1:12 am    Post subject: Reply with quote

Fitzcarraldo wrote:

Gentoo is not as popular as it used to be, and there is no way of dressing it up any other way. But, as I've written before, that does not mean it is dying or 'nearly dead'. It simply means it has settled down to a core group of enthusiasts who like using it and appreciate its features. There is nothing wrong with that.

It's pretty anecdotal to try and determine the user-base from the the basis of new threads here; although when there are a lot of new installs going on, the 'Installing Gentoo' section seems much more active than of late.

There are also a lot more binary distro options than there were 20-years ago, and a hell of a lot more experienced linux users than back then. It's altogether likely people coming to Gentoo now won't need quite the amount of support to get a booting system as was required back in the early 00s, and the faintly curious will find a binary flavour that suits them rather than dedicating a day or so to an install.

The best you could probably say is, it's relative. when I first installed gentoo in the early 00's I didn't have a forums account, nor the *nix knowledge to ask a sensible question if I'd had one. It took me a few years to learn and build a gentoo system I was really happy with. I didn't post here until years later when I broke a production system and needed to bring it back up in a hurry.

If you want to know how many are still quietly sitting on the sidelines with systems humming along, the dev's will have to break something fundamental, that rarely happens now. Whenever it has happened in the past a surprising number of old handles magically reappear for a time, some unseen for months/years prior.

I'd be really surprised if anyone who has any real depth of experience with Gentoo could be happy completely abandoning it for a binary distro, the convenience of being able to edit a flag or package (main tree or overlay) in most cases to extend functionality and let portage sort out the dependency chain beats any other distro in my experience.

I can't see it dying anytime soon, but I do get a grin out of the annual obituary announcement ;)
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Juippisi
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Joined: 30 Sep 2005
Posts: 494
Location: /home

PostPosted: Tue Oct 15, 2019 12:09 pm    Post subject: Reply with quote

^ TLDR: There's so much more info on the internet nowadays, and people are more capable of searching it on their own.

I'd also like to remind that there are more discussion networks now, like discord, reddit etc.
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Fitzcarraldo
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Joined: 30 Aug 2008
Posts: 1873
Location: United Kingdom

PostPosted: Tue Oct 15, 2019 1:37 pm    Post subject: Reply with quote

I think former Gentoo developer and Council member Donnie Berkholz got it about right, back in 2013: Ranking Linux distributions, and the decline of the traditional distros

Discussion on Reddit in 2016: Why did Gentoo peak in popularity in 2005, then fade into obscurity?
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erm67
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Joined: 01 Nov 2005
Posts: 634
Location: EU

PostPosted: Thu Oct 17, 2019 9:50 am    Post subject: Reply with quote

Ridrok wrote:

Are you sure mpd supports DSD format? I never tested since my Pi DAC supports 192/24 maximum, then I provide it with DSD converted to 88/24 in FLAC format usually.


Sorry for the long delay, yes of course mpd does supports dsf it also supports dop (dsd over pcm) that you are very likely using with your DAC, unfortunately you have to specify in the config that the dac wants data in the dop format and not a dsd stream.
Beware of sacd_extract bugs a lot of the stuff that you find online was extracted with a buggy version and contains a lot of noise, it might be interesting for the multichannel version but on my equipment a 2 channels versions, 96/24, from a BDA
sounds better than some 2 channels SACD stuff that can be downloaded. Also the algorithms used in dsf2dsd are questionable, even the README in the sources says they could be better. I am also not 100% certain that converting from a DSD bitstream to DSD packed into PCM is truly lossless, and most importantly the SACD stuff was originally encoded @44.1 with a very high bitrate .... and a SACD of Post Malone will probably be the same as the 128k mp3 anyway.


Code:
audio_output {
        type            "alsa"
        name            "My ALSA USB DAC"
        device          "hw:1,0"        # optional
        mixer_type      "hardware"      # optional
        mixer_device    "hw:1"  # optional
        mixer_control   "PCM"           # optional
        auto_resample   "no"
        auto_channels   "no"
        auto_format     "no"
        dsd_usb         "yes"
        dop             "yes"

}


It will also decode dsf dff and dop but doen't use automatically a moltitude (44.1 to 88.2) so better not use it that way.

Regarding gentoo decline, well clearly a lot of stuff that people badly wants is missing and stuff that nobody cares for is overhyped.
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Ridrok
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Joined: 26 Jan 2014
Posts: 53
Location: France

PostPosted: Sun Oct 20, 2019 10:15 am    Post subject: Reply with quote

You are right, sound is converted :(

If I play a .dff file which is 5.6Mhz, my DAC switches to 192khz/32bits and not DSD format.
Code:
Vangogh ~ # cat /proc/asound/card0/stream0
FOSTEX FOSTEX USB AUDIO HP-A4 at usb-0000:0b:00.3-3, high speed : USB Audio

Playback:
  Status: Running
    Interface = 1
    Altset = 1
    Packet Size = 288
    Momentary freq = 192031 Hz (0x18.0100)
    Feedback Format = 16.16
  Interface 1
    Altset 1
    Format: S32_LE
    Channels: 2
    Endpoint: 1 OUT (ASYNC)
    Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800
    Data packet interval: 125 us
  Interface 1
    Altset 2
    Format: S32_LE
    Channels: 2
    Endpoint: 1 OUT (ASYNC)
    Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800
    Data packet interval: 125 us
  Interface 1
    Altset 3
    Format: S32_LE
    Channels: 2
    Endpoint: 1 OUT (ASYNC)
    Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800
    Data packet interval: 125 us

Moreover, I should see a Format: SPECIAL on one Altset which is the DSD support but I don't. So it seams the DAC is not yet fully supported.
On Windows I am able to make it switch to DSD when playing the same .dff but with fubar2k only and it's a real pain to make it work.

Device
Code:
Bus 005 Device 002: ID 1019:0011 Elitegroup Computer Systems (ECS)
Device Descriptor:
  bLength                18
  bDescriptorType         1
  bcdUSB               2.00
  bDeviceClass          239 Miscellaneous Device
  bDeviceSubClass         2
  bDeviceProtocol         1 Interface Association
  bMaxPacketSize0        64
  idVendor           0x1019 Elitegroup Computer Systems (ECS)
  idProduct          0x0011
  bcdDevice            0.10
  iManufacturer           1 FOSTEX
  iProduct                2 FOSTEX USB AUDIO HP-A4
  iSerial                 3 00001
  bNumConfigurations      1
  Configuration Descriptor:
    bLength                 9
    bDescriptorType         2
    wTotalLength       0x00f0
    bNumInterfaces          2
    bConfigurationValue     1
    iConfiguration          0
    bmAttributes         0x80
      (Bus Powered)
    MaxPower              500mA
    Interface Association:
      bLength                 8
      bDescriptorType        11
      bFirstInterface         0
      bInterfaceCount         2
      bFunctionClass          1 Audio
      bFunctionSubClass       0
      bFunctionProtocol      32
      iFunction               0
    Interface Descriptor:
      bLength                 9
      bDescriptorType         4
      bInterfaceNumber        0
      bAlternateSetting       0
      bNumEndpoints           0
      bInterfaceClass         1 Audio
      bInterfaceSubClass      1 Control Device
      bInterfaceProtocol     32
      iInterface              2 FOSTEX USB AUDIO HP-A4
      AudioControl Interface Descriptor:
        bLength                 9
        bDescriptorType        36
        bDescriptorSubtype      1 (HEADER)
        bcdADC               2.00
        bCategory              10
        wTotalLength       0x002e
        bmControls           0x00
      AudioControl Interface Descriptor:
        bLength                 8
        bDescriptorType        36
        bDescriptorSubtype     10 (CLOCK_SOURCE)
        bClockID               16
        bmAttributes            1 Internal fixed clock
        bmControls           0x07
          Clock Frequency Control (read/write)
          Clock Validity Control (read-only)
        bAssocTerminal          0
        iClockSource            0
      AudioControl Interface Descriptor:
        bLength                17
        bDescriptorType        36
        bDescriptorSubtype      2 (INPUT_TERMINAL)
        bTerminalID             1
        wTerminalType      0x0101 USB Streaming
        bAssocTerminal          0
        bCSourceID             16
        bNrChannels             2
        bmChannelConfig    0x00000003
          Front Left (FL)
          Front Right (FR)
        iChannelNames           0
        bmControls         0x0000
        iTerminal               0
      AudioControl Interface Descriptor:
        bLength                12
        bDescriptorType        36
        bDescriptorSubtype      3 (OUTPUT_TERMINAL)
        bTerminalID             2
        wTerminalType      0x0301 Speaker
        bAssocTerminal          0
        bSourceID               1
        bCSourceID             16
        bmControls         0x0000
        iTerminal               0
    Interface Descriptor:
      bLength                 9
      bDescriptorType         4
      bInterfaceNumber        1
      bAlternateSetting       0
      bNumEndpoints           0
      bInterfaceClass         1 Audio
      bInterfaceSubClass      2 Streaming
      bInterfaceProtocol     32
      iInterface              0
    Interface Descriptor:
      bLength                 9
      bDescriptorType         4
      bInterfaceNumber        1
      bAlternateSetting       1
      bNumEndpoints           2
      bInterfaceClass         1 Audio
      bInterfaceSubClass      2 Streaming
      bInterfaceProtocol     32
      iInterface              0
      AudioStreaming Interface Descriptor:
        bLength                16
        bDescriptorType        36
        bDescriptorSubtype      1 (AS_GENERAL)
        bTerminalLink           1
        bmControls           0x05
          Active Alternate Setting Control (read-only)
          Valid Alternate Setting Control (read-only)
        bFormatType             1
        bmFormats          0x00000001
          PCM
        bNrChannels             2
        bmChannelConfig    0x00000003
          Front Left (FL)
          Front Right (FR)
        iChannelNames           0
      AudioStreaming Interface Descriptor:
        bLength                 6
        bDescriptorType        36
        bDescriptorSubtype      2 (FORMAT_TYPE)
        bFormatType             1 (FORMAT_TYPE_I)
        bSubslotSize            4
        bBitResolution         16
      Endpoint Descriptor:
        bLength                 7
        bDescriptorType         5
        bEndpointAddress     0x01  EP 1 OUT
        bmAttributes            5
          Transfer Type            Isochronous
          Synch Type               Asynchronous
          Usage Type               Data
        wMaxPacketSize     0x0400  1x 1024 bytes
        bInterval               1
        AudioStreaming Endpoint Descriptor:
          bLength                 8
          bDescriptorType        37
          bDescriptorSubtype      1 (EP_GENERAL)
          bmAttributes         0x00
          bmControls           0x00
          bLockDelayUnits         0 Undefined
          wLockDelay         0x0000
      Endpoint Descriptor:
        bLength                 7
        bDescriptorType         5
        bEndpointAddress     0x81  EP 1 IN
        bmAttributes           17
          Transfer Type            Isochronous
          Synch Type               None
          Usage Type               Feedback
        wMaxPacketSize     0x0004  1x 4 bytes
        bInterval               4
    Interface Descriptor:
      bLength                 9
      bDescriptorType         4
      bInterfaceNumber        1
      bAlternateSetting       2
      bNumEndpoints           2
      bInterfaceClass         1 Audio
      bInterfaceSubClass      2 Streaming
      bInterfaceProtocol     32
      iInterface              0
      AudioStreaming Interface Descriptor:
        bLength                16
        bDescriptorType        36
        bDescriptorSubtype      1 (AS_GENERAL)
        bTerminalLink           1
        bmControls           0x05
          Active Alternate Setting Control (read-only)
          Valid Alternate Setting Control (read-only)
        bFormatType             1
        bmFormats          0x00000001
          PCM
        bNrChannels             2
        bmChannelConfig    0x00000003
          Front Left (FL)
          Front Right (FR)
        iChannelNames           0
      AudioStreaming Interface Descriptor:
        bLength                 6
        bDescriptorType        36
        bDescriptorSubtype      2 (FORMAT_TYPE)
        bFormatType             1 (FORMAT_TYPE_I)
        bSubslotSize            4
        bBitResolution         24
      Endpoint Descriptor:
        bLength                 7
        bDescriptorType         5
        bEndpointAddress     0x01  EP 1 OUT
        bmAttributes            5
          Transfer Type            Isochronous
          Synch Type               Asynchronous
          Usage Type               Data
        wMaxPacketSize     0x0400  1x 1024 bytes
        bInterval               1
        AudioStreaming Endpoint Descriptor:
          bLength                 8
          bDescriptorType        37
          bDescriptorSubtype      1 (EP_GENERAL)
          bmAttributes         0x00
          bmControls           0x00
          bLockDelayUnits         0 Undefined
          wLockDelay         0x0000
      Endpoint Descriptor:
        bLength                 7
        bDescriptorType         5
        bEndpointAddress     0x81  EP 1 IN
        bmAttributes           17
          Transfer Type            Isochronous
          Synch Type               None
          Usage Type               Feedback
        wMaxPacketSize     0x0004  1x 4 bytes
        bInterval               4
    Interface Descriptor:
      bLength                 9
      bDescriptorType         4
      bInterfaceNumber        1
      bAlternateSetting       3
      bNumEndpoints           2
      bInterfaceClass         1 Audio
      bInterfaceSubClass      2 Streaming
      bInterfaceProtocol     32
      iInterface              0
      AudioStreaming Interface Descriptor:
        bLength                16
        bDescriptorType        36
        bDescriptorSubtype      1 (AS_GENERAL)
        bTerminalLink           1
        bmControls           0x05
          Active Alternate Setting Control (read-only)
          Valid Alternate Setting Control (read-only)
        bFormatType             1
        bmFormats          0x00000001
          PCM
        bNrChannels             2
        bmChannelConfig    0x00000003
          Front Left (FL)
          Front Right (FR)
        iChannelNames           0
      AudioStreaming Interface Descriptor:
        bLength                 6
        bDescriptorType        36
        bDescriptorSubtype      2 (FORMAT_TYPE)
        bFormatType             1 (FORMAT_TYPE_I)
        bSubslotSize            4
        bBitResolution         32
      Endpoint Descriptor:
        bLength                 7
        bDescriptorType         5
        bEndpointAddress     0x01  EP 1 OUT
        bmAttributes            5
          Transfer Type            Isochronous
          Synch Type               Asynchronous
          Usage Type               Data
        wMaxPacketSize     0x0400  1x 1024 bytes
        bInterval               1
        AudioStreaming Endpoint Descriptor:
          bLength                 8
          bDescriptorType        37
          bDescriptorSubtype      1 (EP_GENERAL)
          bmAttributes         0x00
          bmControls           0x00
          bLockDelayUnits         0 Undefined
          wLockDelay         0x0000
      Endpoint Descriptor:
        bLength                 7
        bDescriptorType         5
        bEndpointAddress     0x81  EP 1 IN
        bmAttributes           17
          Transfer Type            Isochronous
          Synch Type               None
          Usage Type               Feedback
        wMaxPacketSize     0x0004  1x 4 bytes
        bInterval               4
can't get device qualifier: Resource temporarily unavailable
can't get debug descriptor: Resource temporarily unavailable
cannot read device status, Resource temporarily unavailable (11)
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Ridrok
n00b
n00b


Joined: 26 Jan 2014
Posts: 53
Location: France

PostPosted: Sun Oct 20, 2019 12:30 pm    Post subject: Reply with quote

Hi again

I managed to patch the kernel to activate DSD on this DAC, see DSD_U32_LE above on Setting 1.
Code:
Vangogh ~ # cat /proc/asound/card0/stream0
FOSTEX FOSTEX USB AUDIO HP-A4 at usb-0000:0b:00.3-3, high speed : USB Audio

Playback:
  Status: Running
    Interface = 1
    Altset = 1
    Packet Size = 288
    Momentary freq = 192000 Hz (0x18.0000)
    Feedback Format = 16.16
  Interface 1
    Altset 1
    Format: S32_LE DSD_U32_LE
    Channels: 2
    Endpoint: 1 OUT (ASYNC)
    Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800
    Data packet interval: 125 us
  Interface 1
    Altset 2
    Format: S32_LE
    Channels: 2
    Endpoint: 1 OUT (ASYNC)
    Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800
    Data packet interval: 125 us
  Interface 1
    Altset 3
    Format: S32_LE
    Channels: 2
    Endpoint: 1 OUT (ASYNC)
    Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800
    Data packet interval: 125 us

I did activate it for Altset 1 to see, but I don't know how to make use of it. Whatever dff I send it's played in PCM. I will need a little help from you to debug this.
Then I plan in finding the proper Altset, then if it works I need to know how to submit a kernel patch.

It may need more work anyway in kernel, reading the specs of the DAC:
Code:
• [USB] connector
  • Interface: USB 2.0 high speed
  • Supporting sampling frequency:
     (PCM): 44.1 kHz, 48 kHz, 88.2 kHz,
            96 kHz, 176.4 kHz, 192 kHz
     (DSD):  2.8 MHz, 5.6 MHz
  • Supporting quantization bit width: 16 bit, 24 bit

So the 3 settings are most probably 16bits, 24bits and DSD.

Edit
comparing the 3 sections of LSUSB, I found:
Code:
    Interface Descriptor:
[...]
      bAlternateSetting       1
[...]
      AudioStreaming Interface Descriptor:
        bLength                 6
        bDescriptorType        36
        bDescriptorSubtype      2 (FORMAT_TYPE)
        bFormatType             1 (FORMAT_TYPE_I)
        bSubslotSize            4
        bBitResolution         16
[...]
    Interface Descriptor:
[...]
      bAlternateSetting       2
[...]
      AudioStreaming Interface Descriptor:
        bLength                 6
        bDescriptorType        36
        bDescriptorSubtype      2 (FORMAT_TYPE)
        bFormatType             1 (FORMAT_TYPE_I)
        bSubslotSize            4
        bBitResolution         24
[...]
    Interface Descriptor:
[...]
      bAlternateSetting       3
[...]
      AudioStreaming Interface Descriptor:
        bLength                 6
        bDescriptorType        36
        bDescriptorSubtype      2 (FORMAT_TYPE)
        bFormatType             1 (FORMAT_TYPE_I)
        bSubslotSize            4
        bBitResolution         32
[...]

So I confirm setting 1 is 16bits PCM, setting 2 is 24bits PCM, setting 3 is DSD, now I need to code a bit.

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PostPosted: Sun Oct 20, 2019 5:20 pm    Post subject: Reply with quote

Sorry for spam, but my Sunday was a good day :)

https://i.imgur.com/9DGo39t.jpg

DAC is now finaly in DSD mode, but I probably did not need any path at all.

Code:
--- sound/usb/quirks_origin.c   2019-10-20 19:13:36.143961381 +0200
+++ sound/usb/quirks.c  2019-10-20 19:24:22.927892105 +0200
@@ -1423,6 +1423,10 @@
                if (fp->altsetting == 2)
                        return SNDRV_PCM_FMTBIT_DSD_U32_BE;
                break;
+       case USB_ID(0x1019, 0x0011):  /* FOSTEX HP-A4 */
+               if (fp->altsetting == 3)
+                       return SNDRV_PCM_FMTBIT_DSD_U32_LE;
+               break;
        default:
                break;
        }

That was needed or no I have to check, but to play DoP DSD, I needed more software.
Then I pulled and compiled dsf2flac from here
I had to edit also the source to remove many printf in dsf_file_reader.cpp due to unsupported tags in my dsf files.
Then finally added this bash script found online in path:
Code:
#!/bin/bash
dsf2flac -d -i "$1" -r 176400 -o - 2>/dev/null | ffmpeg -i - -r 176400 -c pcm_s32le -f alsa hw:0


Et voila!

Edit
Installed mpd and cantata and got same result after configuration :)
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PostPosted: Mon Oct 21, 2019 4:26 am    Post subject: Reply with quote

If you read a bit around you'll find out that most SACD were just 16bit CDs upsampled and overpriced .... and yes mpd doesn't support SACD ISOs, you'll have to extract the tracks, there is an mpd fork that reads SACD isos but probably is not worth since most SACD suck. Maybe they sound good for you because of some limitations of your DAC.

DSD is a dead technology that made a comeback recently with the advent of cheap sigma-delta modulator DAC that unlike multibit DACs play natively the DSD format and use a hardware interpolator stage to convert incoming PCM data to DSD. Some chipsets like the AKM4452 and AKM4490 (but also other) have a direct path that skips the hardware interpolator if the incoming data is DSD or DoP.

So basically if you send a PCM 96khz/24bit to one of those (very common) one bit DAC it will be converted internally to DSD by a hardware 8x interpolator and played, see the block diagram of one of them for example: AKM4452.

According to some people the hardware interpolator in most one bit DAC chipsets sucks and the music will be a lot better if upsampled to DSD by a good software program (there is a SoX fork that does it) and sent as DoP to one of those one bit DACs that has a direct path for DSD data, that is the main reason why everybody talks about DSD today. The conversion of PCM to DSD is lossy and I think mpd doesn't support it on the fly, of course the conversion is lossy also when done in hardware and this explains why multibit DACs that can play PCM data without interpolation are a lot more expensive. This also explain why PCM converted to DSD might sound better on those chipsets, the software conversion might be less lossy than the hardware one. And why SACD sound a lot better on those DACs, in fact it's PCM that sounds worse than it should.

Another thing to take into consideration is digitalization noise, that for sigma delta modulators is in the high frequencies, the higher the bitrate the higher is the frequency of the digitalization noise, this is why one bit DAC are using always higher rates like DSD512: the digitalization noise is at at frequencies so high to be not hearable at that bit rate. The digitalization noise in the original DSD64 format used in SACD was too close to the hearable frequencies to be cleaned efficently so also SACD can sound better if upsampled to DSD512.

Apparently also your "High-End" DAC is one bit and uses a hardware interpolator internally: https://www.ti.com/lit/ds/symlink/pcm1792a.pdf, it is not imediately clear from the block diagram if DSD data will be played directly without going through the interpolator, but it is very likely since for you DSD sounds better than PCM; so maybe the trick of DSD upsampling will work also for you.

Basically since your DAC is going to convert everything to DSD internally using a hardware interpolator of unknown quality you can try to do a high quality conversion in software of all PCM data and send only DSD256 to it. look for a SoX fork if you want to try.

I am thinking about a modi multibit DAC for my birthday ;-) Now that so many web radios are streaming in flac quality: chillout


PS Sorry if I disappointed you, after all the efforts you put in your quest for bit perfection it is not nice to discover that all that perfection is going to be grinded and maimed into an hardware interpolator, but I felt the same when I discovered it :-
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PostPosted: Mon Oct 21, 2019 11:32 am    Post subject: Reply with quote

What comes out of a PC is not music, however its encoded because it was destroyed before it went in :)
The class-D amplifiers just finish it off.

It doesn't matter to me any more with my ears high frequency cut off at 7.5kHz on the good side and 6.5kHz on the other side.
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PostPosted: Mon Oct 21, 2019 12:57 pm    Post subject: Reply with quote

It is not just a matter of frequencies but also clock jitter and other noises introduced by digitalization .... one bit modulator DACs are (theoretically) immune to clock jitter (that allegedly plagued early CD players) and do not need an extremely precise voltage so even cheap devices are usually decent. Paired with a vintage AA amplifier and some good entry level HI-FI speakers they can do wonders, I got some HD tracks of the early The Cure recently and they sent me a shiver down my spine ... like the crystals on the cookies, probably they sound better now than they did on the CD player in the '80, and I had an HIFI CD player from Technics back then ...
Try to find some second hand vintage AA ampli, the only real improvement on modern class AB(AA) probably is the direct-digital button found on modern ampli that bypasses the pre-amp stage that could theoretically introduce some noise.
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PostPosted: Mon Oct 21, 2019 5:09 pm    Post subject: Reply with quote

NeddySeagoon wrote:
What comes out of a PC is not music, however its encoded because it was destroyed before it went in :)
The class-D amplifiers just finish it off.

It doesn't matter to me any more with my ears high frequency cut off at 7.5kHz on the good side and 6.5kHz on the other side.

Wow. And there was me getting depressed because I can't hear anything above 10 kHz! Well, on a good day, one ear can perhaps get up to 14 kHz (I think).
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PostPosted: Mon Oct 21, 2019 5:36 pm    Post subject: Reply with quote

Fitzcarraldo,

As an asthmatic since birth, my ears had an unusually wide frequency response. (it goes hand in hand)
They were good to over 22kHz when I started high school.
It was never established if that was the limit of the test gear or my hearing.
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PostPosted: Mon Oct 21, 2019 5:40 pm    Post subject: Reply with quote

http://www.noiseaddicts.com/2009/03/can-you-hear-this-hearing-test/

https://www.npr.org/sections/therecord/2015/06/02/411473508/how-well-can-you-hear-audio-quality

First one tests your ability to hear sound of different frequencies. Old fogies will be doing well if they can hear up to 15 kHz. A lot of older people (me, for example) can't even hear up to that (If the wind is blowing in the correct direction I think I can hear 14 kHz in one ear above the tinnitus).

Second one tests your ability to discern audio quality (quantisation and sample frequency). I got 33%!
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PostPosted: Mon Oct 21, 2019 6:17 pm    Post subject: Reply with quote

Hi,

I don't feel DSD are better than Flac 196khz on my DAC, but relatively small monitoring speakers only with my PC, not big detailed ones, but both DFF and FLAC@176 are so much better (coming from good SACD of course) than MP3 @384kbps. Everyone listening my PC is impressed, then no I don't feel disappointed.
I don't know the internal of the DAC, it was not a $100 like the one you show but 4 times more. I did search a lot before investing in it and I don't regret it whatever it converts or not internally.

When I want to play music on a big system I use a PI, this may looks stupid to you but a PI II with a cheap DAC hat and a proper (old classic regulated one from the 90" and fixed to 5.00V on load / not a switching one) power supply has an incredible neutral and natural sound with high frequency flacs and so good plugged to an antic Arman Cardon AVR7000 and Triangle Lyrr 222 speakers.

In conclusion, I wanted my PC DAC (Fostex HP-A4) to work fully on Linux and it does now. Sound is so good compared to most PC systems that I don't feel the need to change it. Maybe you can try to find one, I find it very good in neutrality and sound details despite the maybe questionable internals.

Ridrok

Edit I did the above tests, and seeing I did 33% too to the second test and 14khz is complete silence for me (my son hears them all) on the 1st one, I feel that at 44y old I don't need to search anymore for better sound system since what I have is beyond what I can ear 8O
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PostPosted: Mon Oct 21, 2019 10:44 pm    Post subject: Reply with quote

Ridrok wrote:
I did the above tests, and seeing I did 33% too to the second test and 14khz is complete silence for me (my son hears them all) on the 1st one, I feel that at 44y old I don't need to search anymore for better sound system since what I have is beyond what I can ear 8O

Yes, those tests are eye-openers, aren't they?! All my family did a lot better than me on those tests, especially the younger members. One of them could hear above 20 kHz (I think it was 22 kHz) in the first test, and got 100% on the second test (which is exceptional because the others all got 50%, so even young people struggle to hear the difference).
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PostPosted: Tue Oct 22, 2019 7:40 am    Post subject: Reply with quote

Ridrok wrote:
Hi,

I don't feel DSD are better than Flac 196khz on my DAC, but relatively small monitoring speakers only with my PC, not big detailed ones, but both DFF and FLAC@176 are so much better (coming from good SACD of course) than MP3 @384kbps. Everyone listening my PC is impressed, then no I don't feel disappointed.
I don't know the internal of the DAC, it was not a $100 like the one you show but 4 times more. I did search a lot before investing in it and I don't regret it whatever it converts or not internally.

When I want to play music on a big system I use a PI, this may looks stupid to you but a PI II with a cheap DAC hat and a proper (old classic regulated one from the 90" and fixed to 5.00V on load / not a switching one) power supply has an incredible neutral and natural sound with high frequency flacs and so good plugged to an antic Arman Cardon AVR7000 and Triangle Lyrr 222 speakers.

In conclusion, I wanted my PC DAC (Fostex HP-A4) to work fully on Linux and it does now. Sound is so good compared to most PC systems that I don't feel the need to change it. Maybe you can try to find one, I find it very good in neutrality and sound details despite the maybe questionable internals.


You know "Audiophiles" are very strange people, some of them are convinced that sending PCM data over an I2S bus will make it sound a lot better than sending the same PCM data to the same audiocard over the USB bus :-) But yeah also the 128k mp3 I downloaded from Napster sounded a lot better because I used a 48k modem instead of a VDSL line.....

Fostex doesn't produce DAC chipsets, but USB audiocards (that is the correct name for the device) that uses various chipsets inside including a DAC, a chipset that actually does the conversion job. You probably had not bought it if it was branded as a USB sound card but hey it's a Audiophile DAC .....
My links were to DAC chipsets, the second to the one used in your Fostex device. You should not be surprised if your 4 hundred $ USB sound card uses the same chipset used in 100$ (or less) cards, the chipset alone costs probably only 40$ and also integrated Amplifiers or USB sound cards a lot more expensive than yours (Onkyo Marantz Rotel) use the same DAC chipsets in their entry level models. Probably the rest of the components in your card are of better quality than those 100$ china USB sound cards that uses the same chipset.
BTW the SHIIT multibit dac costs 250$ and they claim is the cheapest multibit "DAC" on the market ...

Marketing is a shit anyway. You keep bragging about the quality of the SACDs but nobody does. If you search a bit around you will find out that only Sony Music and a few other labels actually use the DSD format to archive the master recordings, the problem is that the software used to edit the recording sessions and create the final song only work with PCM, there is no software that lets you directly edit DSD data, go figure 10 years ago when CPUs were a lot slower.
I told you already that the conversion DSD->PCM is lossy, well in the SACD you value so much the lossy conversion was done twice: DSD->PCM, editing and than again PCM->DSD now you apparently convert them again into regular flac doing the DSD->PCM lossy conversion for the third time. The best part is that your Fostex USB sound card uses internally the DSD format so it will perform a fourth lossy PCM->DSD conversion of your flacs coming from a good SACD .......
Imagine the quality you will get, but hey at least you do it bit perfectly :-) You can also use DoP inside flacs, and play DSD directly since your card can do it and now you know how to do it, or compress the dsf losslessly using wavepack.


Try to see with a graphical equalizer how much of a song you are missing not hearing those frequencies, very little since most commercial music is optimized to be played on car stereos while driving and they know very well that oldies can't hear well but has the money to buy music :-)

Try this for high quality upsampling to DSD
https://www.signalyst.com/consumer.html
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PostPosted: Tue Oct 22, 2019 7:57 am    Post subject: Reply with quote

For good quality music, break out your LPs.
An LP is a big black CD that plays both sides :)

It has to be analogue end to end.
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PostPosted: Tue Oct 22, 2019 8:04 am    Post subject: Reply with quote

NeddySeagoon wrote:
For good quality music, break out your LPs.
An LP is a big black CD that plays both sides :)

It has to be analogue end to end.


You probably have never heard about mandatory RIAA pre-amps and clock jiiter. I doubt you LP will rotate twice at the same speed .....
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PostPosted: Tue Oct 22, 2019 8:29 am    Post subject: Reply with quote

/me just smiles :)
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PostPosted: Tue Oct 22, 2019 9:49 am    Post subject: Reply with quote

The smile on your face doesn't change the truth.

The amount of information that could be stored on vinyl was not enough for music, in particular spinning @33 clock the bitrate was very low but an entire album could fit a single disk, on the other side @45 the bitrate was better but there wasn't enough space to store an entire album. So they decided to compress the music to solve the problem of low resolution (bitrate), the RIAA set the rules for the compression and the mandatory RIAA pre-amp is a decompressor that uses the algorithm set by the RIAA. So it is a compressed format, maybe a hi-fi turntable with a diamond head and stepper motor is better than 128k mp3 played on a cheap soundcard with weak clocks and DAC. However for the same money you can play your 128k mp3 a lot better, maybe interpolating it at 24bits or DSD (if the DAC inside your sound card uses that format internally) and playing it on a decent soundcard ;-) The interpolation can't be worse than a RIAA decompressor anyway, and even the the most expensive turntable head can not remove it nor recover the informations that were lost in the compression process.
How can you compare a compressed format to an uncompressed one?

My older brother that loved to listen to good music had a large collection of EP btw, same size of a 33 LP but spinning @45 (higher clock = higher bitrate) the resolution is higher and the music was a lot better, but only a couple of songs could fit the disk, not as good as on a good hifi CD player of course. Everybody back than knew that the quality of 33LP was very low apparently now this knowledge was lost and bragging about the 'quality' of vinyl LP is so trendy. I loved vinyl when the disk got a bit wobbly because of sunlight, not to mention the inevitable schratches and the dust and hair that clogged the head, and the expensive equipment to clean the disks and the head ... All that work, trouble and money to listen to shitty compressed music mixed with clicks and pops not to mention the ground loops on lower quality equipments.... maybe you cannot hear them because of your health problems .... damaged ears maybe?

Try some 78 on a grammophone they must be better.
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PostPosted: Tue Oct 22, 2019 10:54 am    Post subject: Reply with quote

erm67,

You are changing one set of objectionable audio artefacts for another.
Which is subjectively the most objectionable probably depends on the equipment you grew up with.

Calling RIAA equalisation a compressor is somewhat oversimplification of what it does.
During recording, frequencies below 1kHz are indeed reduced in amplitude. Frequencies above 1KHz are increased in amplitude.
The net result is that the modulation in the groove in the LP is approximately frequency independent.
This process is reversed on playback. Hopefully, electronically, but there are mechanical ways too.
RIAA equalisation addresses both mechanical and media issues.

With some care and attention RIAA equalisation can be flat from 20Hz in 20kHz to better than 1db.
Since the human ear struggles to discern 3db amplitude steps, that's plenty good enough. (Don't forget that db is a log scale).
That's what is all about, good enough, with some margin.

78's, with no RIAA equalisation have a much poorer frequency response. My ears can probably hear it all.
They predate recording and playback with the aid of electronics. Indeed, if you get a used 78, there may not be much left because of groove wear from the tracking weight. The high frequencies are lost first.

I've never bothered to measure the frequency response of a PC sound card.

128k mp3 is not music. The 11:1 lossy compression process destroys it. Interpolating, or trying to make up, fake if you like, the missing data is useless.
Once the data has been thrown away, its gone for good.

As you hint, 'quality' is a subjective experience. I grew up with reel to reel tape and vinyl. When CDs came along, the shot noise and quantisation error on quiet passages sounded horrible.
There were other objectionable artefacts compared to the tape and vinyl I grew up with.

I'm not afraid to wash my collection of LPs. :)
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PostPosted: Tue Oct 22, 2019 11:05 am    Post subject: Reply with quote

Not to mention that cd's are almost as vulnerable as vinyl.
You certainly don't want to leave them in sunlight, high heat, etc, and over time they simply degrade.
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PostPosted: Tue Oct 22, 2019 12:18 pm    Post subject: Reply with quote

Anon-E-moose wrote:
Not to mention that cd's are almost as vulnerable as vinyl.
You certainly don't want to leave them in sunlight, high heat, etc, and over time they simply degrade.

More vulnerable than vinyl, actually. Some of the Audio CDs I bought around 20 years ago have already suffered the well-known phenomenon of disc rot. This, despite being very carefully kept and handled in a controlled environment. Optical discs are rubbish from a longevity point of view. Vinyls, on the other hand, if kept in a controlled environment, will last almost indefinitely:

Record collector builds world's largest vinyl hoard – six million and counting

Even with my poor hearing I can hear how bad a 128 kb/s mp3 sounds, but when you get up to 320 kb/s it's a different matter, In most cases I can't hear the difference between 320 kb/s and a 16-bit 44.1 kHz Audio CD, and, as the tests in my earlier link demonstrate, most people struggle to tell the difference too: Audiophile or Audio-Fooled? How Good Are Your Ears?.

Regarding sampling theory, the video Digital Audio: The Line Between Audiophiles and Audiofools is quite good if someone does not understand why 16-bit 44.1 kHz was chosen. As to finer quantisation and higher frequencies, 'The Difference Between 24-bit & 16-bit Audio is Inaudible Noise'.

An audiophile friend of mine with a life-long passion for top-end hi-fi has an insanely expensive hi-fi system which is integrated throughout his house, with hand-built pre-amps imported from a small, specialist manufacturer, for example. His main speakers alone cost a lot more than most people pay for an expensive sound system. He moved to servers with uncompressed (FLAC and WAV) files of well-produced 16-bit 44.1 kHz Audio CDs years ago.
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PostPosted: Tue Oct 22, 2019 8:29 pm    Post subject: Reply with quote

Back a bit to gentoo, is it possible to remove things I don't need and which take too much time to update like Rust, Ruby... I don't know why these two came in months or years ago and what they are used for in gentoo if I don't plan in coding with these languages and would like to get rid of these big packages..
Python, C, C++ are enough for me to code and perl very few (and C# on Windows).
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PostPosted: Tue Oct 22, 2019 9:24 pm    Post subject: Reply with quote

Ridrok wrote:
Back a bit to gentoo, is it possible to remove things I don't need and which take too much time to update like Rust, Ruby...


This is turning into a help thread, (there's already some interesting audio discussion here - this thread would benefit from a rename tbh) why not open a thread in Portage / Programming for these issues?

As for your question, I think on my system the rust dependency sheets home only to firefox, so as long as you don't need firefox you probably could nuke it, but ie.
Code:
equery d rust
would be a good place to start. Ruby is laced through webkit-gtk so if you don't use Gnome or derivatives maybe you can get rid of that too. Building custom systems is what Gentoo is for, after all.

I'm assuming that the last rites for the 'nearly dead Gentoo' can be postponed for now? ;)
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