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Thread: Output from phono stage too low for good digital recordings

  1. #11
    Join Date: Jan 2017

    Location: Hampshire

    Posts: 18
    I'm Tony.

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    Quote Originally Posted by cre009 View Post
    Some observations from my own experience.
    .
    Ah. most interesting comment about Linux and the recommendation to use Jack - I hadn't thought of these factors before. Thanks.

    I had read that the phono stage, especially for MC, is a bit 'entry level' on the GT40a so, having invested in a quite good MC cartridge, I think I'll give that one a miss. The Korg device sounds the most exciting - primarily because it's software appears to be able to identify the music being recorded and then do automatic looks-ups of recorded music databases to get the 'EXIF-like' data for the recording (Album Title, artists, composer, tracks etc) and package each 'track' (equivalent to movement I guess) for you. This saves a huge amount of work. Sadly none of this works under Linux though, which means that if I want to use this device I have to buy a refurbished MAC (being seriously concerned about cyber-crime I absolutely refuse to use that operating system apparently designed for and ideally meeting the needs of cyber criminals, aka Windows).

  2. #12
    Join Date: Jan 2017

    Location: Hampshire

    Posts: 18
    I'm Tony.

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    Quote Originally Posted by Arkless Electronics View Post
    Cheers Gary

    The OP rang me yesterday actually for advice on this and we discussed this option.... I also suggested that he could take the output from his pre amp (the main outputs, not the tape outs which he was trying to use) and thereby, assuming the line stage has gain, boost the output from the phono stage to a suitable level.
    The Arkless 640P has much higher output than the standard one (an AOS member with one on order has actually asked me to reduce the gain a bit on his!) at 64dB but I can customise for any gain at no extra cost.
    This would vastly increase the quality of the sound as well of course! Much more so than a better sound card!
    Hello Jez, I guess I'll report back to you here on the forum rather than taking your valuable time on the phone. I couldn't try your suggestion until this morning when I collected my Rega P7 TT + Cartridge after being serviced by my local hifi adviser here in Winchester. My amplification consists of an AudioLab 8000Q driving an AudioLab 8000P. The 8000Q has two identical RCA line output socket pairs , so I was able to connect the Behringer UAC-202 to the 8000Q line output which was not being used, allowing me to continue to hear the music while digitising it via the Behringer.

    The results have baffled me: the 'histogram' in Audacity was now an almost perfectly straight line. It could be adjusted using the volume control on the 8000Q (obviously) but only became a recognisable recording 'plot' when the volume from the speakers was at an uncomfortable level. Playing back this recording either through the Behringer to a tape input on the 8000Q or via the in-built speakers on the laptop or via the speaker output connector on the laptop into Tape-In on the 8000Q produced a barely audible output on my main speakers (Spendor SP1s). I swapped the connections on the 2 line-outs on the 8000Q - no difference on the volume on the main speakers and no difference to the recording levels on the laptop either - still just a straight line, much much quieter than connecting the Behringer UAC-202 to the output of the 640P. I tried by-passing the Behringer entirely, using the mic-in connector on the laptop connected to the output of the 640P. The recording volume was now higher - but no higher than when using the Behringer taking the signal into the laptop via USB (but of obviously poorer quality using the ADC backing the mic-in on the laptop),

    Bottom line: I think I need your re-engineered quasi-640P running with about 64 dB gain!. But I'll talk to you outside this forum to agree the details - is that OK?

  3. #13
    Join Date: Jan 2017

    Location: Hampshire

    Posts: 18
    I'm Tony.

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    Quote Originally Posted by Rothchild View Post
    I wouldn't let it bother you, 0dBvu = -18dBFS ie when the record was cut and the needles on the meter went through 'zero' that would only make -18 on a 24bit digital meter.

    If you're recording at 24bit you're capturing all the range on the record anyway, further analogue gain just risks pushing more analogue noise in to your recording. If you're really unhappy about the appearance of the file, or you want to have the files at a level where they match other sources just normalise them in audacity.
    Actually, because of a long-track record of total ignorance of recording techniques, exacerbated by ill-thought out farting about, I'm not sure what it is that I'm worried about or trying to achieve. But I certainly feel quite dis-satisfied with what I am achieving. Aside from the (reasonable) limitation in the Behringer UAC-202 - 44.1kHz, 16 bit only - I think I'm enervated by the idea that the essentially flat plot in Audacity means that when I normalise it I'll be boosting the noise just as much as the 'real' signal (whatever that is). Everything tells me that the signal to noise ratio will be much improved if I'm driving the recording to the (once only) maximum of -1dB - which I take to mean dBvu. Or is my ignorance overwhelming my sense of judgement here?

  4. #14
    Join Date: Sep 2012

    Location: East Anglia UK

    Posts: 1,219
    I'm Marc.

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    Tony, you seem very keen to spend money, which is cool, but you're circling a rabbit hole unnecessarily - the levels you described in your OP are perfectly adequate for capturing a good recording at 24bit, if you really insist that the waveform has to be visibly bigger just use the 'normalize' tool in Audacity and set a peak of -0.2dBFS (this will be the same but much easier than faffing around with JACK etc). Pushing the signal level at the input just takes you closer to risking clipping and distorting the recording.

    Bear in mind that 24bits gives you a dynamic range of 144dB, this means that if you had a suitable soundsystem a 24bit recording can contain data representing a sound quieter that we can hear and louder than we can tolerate, you don't really need the signal all stuffed up at the top end of this range.

  5. #15
    Join Date: Sep 2012

    Location: East Anglia UK

    Posts: 1,219
    I'm Marc.

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    Quote Originally Posted by LateJunction View Post
    Actually, because of a long-track record of total ignorance of recording techniques, exacerbated by ill-thought out farting about, I'm not sure what it is that I'm worried about or trying to achieve. But I certainly feel quite dis-satisfied with what I am achieving. Aside from the (reasonable) limitation in the Behringer UAC-202 - 44.1kHz, 16 bit only - I think I'm enervated by the idea that the essentially flat plot in Audacity means that when I normalise it I'll be boosting the noise just as much as the 'real' signal (whatever that is). Everything tells me that the signal to noise ratio will be much improved if I'm driving the recording to the (once only) maximum of -1dB - which I take to mean dBvu. Or is my ignorance overwhelming my sense of judgement here?
    The 'fun' here is caused because there are a bunch of different standards. Analogue equipment tends to be rated in dBVU (Volume Unit) digital systems/meters tend to show dBFS (Full Scale) where 0dB is the point where you run out of bits (and if you hit that point you get distortion and clipping). In general terms if you were to feed a -18dBFS signal in to an analogue box with dBVU meters the meters would sit at 0dB, conversely, if you put a signal out of an analogue box with its meter sitting at 0dB (and don't do any other gain staging to the computer input) the meter in the computer will show -18dBFS.

    So what you've described in your OP actually sounds about right levels wise. Yes if you turn the signal up digitally in audacity you'll turn the noise up too, but that would also be the case for adding gain electronically at the preamp stage.

    By way of example, when I record bands I'll set the gains going in to the system so that the peaks hit between -18 and -12dB FS, this is plenty of signal to work with, means I'm generally safe from clipping the input if something gets momentarily loud and leaves me headroom for summing tracks together.

  6. #16
    Join Date: Oct 2012

    Location: NE England

    Posts: 4,173
    I'm Jez.

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    Quote Originally Posted by LateJunction View Post
    Hello Jez, I guess I'll report back to you here on the forum rather than taking your valuable time on the phone. I couldn't try your suggestion until this morning when I collected my Rega P7 TT + Cartridge after being serviced by my local hifi adviser here in Winchester. My amplification consists of an AudioLab 8000Q driving an AudioLab 8000P. The 8000Q has two identical RCA line output socket pairs , so I was able to connect the Behringer UAC-202 to the 8000Q line output which was not being used, allowing me to continue to hear the music while digitising it via the Behringer.

    The results have baffled me: the 'histogram' in Audacity was now an almost perfectly straight line. It could be adjusted using the volume control on the 8000Q (obviously) but only became a recognisable recording 'plot' when the volume from the speakers was at an uncomfortable level. Playing back this recording either through the Behringer to a tape input on the 8000Q or via the in-built speakers on the laptop or via the speaker output connector on the laptop into Tape-In on the 8000Q produced a barely audible output on my main speakers (Spendor SP1s). I swapped the connections on the 2 line-outs on the 8000Q - no difference on the volume on the main speakers and no difference to the recording levels on the laptop either - still just a straight line, much much quieter than connecting the Behringer UAC-202 to the output of the 640P. I tried by-passing the Behringer entirely, using the mic-in connector on the laptop connected to the output of the 640P. The recording volume was now higher - but no higher than when using the Behringer taking the signal into the laptop via USB (but of obviously poorer quality using the ADC backing the mic-in on the laptop),

    Bottom line: I think I need your re-engineered quasi-640P running with about 64 dB gain!. But I'll talk to you outside this forum to agree the details - is that
    OK?
    That's absolutely fine Tony. Nice speakers BTW!
    Arkless Electronics-Engineered to be better. Tel. 01670 530674 (after 1pm)

    Modded Thorens TD150, Audio Technica AT-1005 MkII, Technics EPC-300MC, Arkless Hybrid MC phono stage, Arkless passive pre, Arkless 50WPC Class A SS power amp, (or) Arkless modded Leak Stereo 20, Modded Kef Reference 105/3's
    ReVox PR99, Studer B62, Ferrograph Series 7, Tandberg TCD440, Hitachi FT-5500MkI, also FT-5500MkII
    Digital: Yamaha CDR-HD1500 (Digital Swiss army knife-CD recorder, player, hard drive, DAC and ADC in one), PC files via 24/96 sound card and SPDIF, modded Philips CD850, modded Philips CD104, modded DPA Little Bit DAC. Sennheiser HD580 cans with Arkless Headphone amp.
    Cables- free interconnects that come with CD players, mains leads from B&Q, dead kettles etc, extension leads from Tesco

  7. #17
    Join Date: Mar 2010

    Location: Sheffield

    Posts: 2,898
    I'm Simon.

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    I do my rips on my sony pcm10, it's a little adc/dac digital recorder with line inputs, decent microphone amps and Microsd card and USB data outputs. It'll do 24/96 and has variable input gain with vu metering. It works fine with every phono stage I've tried.

    Get the best phono stage you can and try a decent digital recorder the behringer is just a toy.
    Kuzma Stabi/S 12", (LP12-bastard) DC motor and optical tacho psu, Benz LP, Paradise (phonostage). MB-Pro, Brooklyn dac and psu, Bruno Putzeys balanced pre, mod86p dual mono amps, Yamaha NS1000m

  8. #18
    Join Date: Jan 2017

    Location: Bristol

    Posts: 111
    I'm Clive.

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    If the OP is using the Pulse Audio protocol on Linux then I believe that gets relatively low resolution rate even if trying to record off a 24 bit card. I read that Pulse has much the same issue as MME direct sound on Windows. A 16 bit recording will actually achieve far less resolution than 16 bit. My understanding is that for every 6db of attenuation below the clipping level there is a loss of 1 bit in the recording resolution. If Audacity is showing a flat line then that will be a low quality recording.

    I still think he should enquire within the Linux community to establish how to get the best results.

  9. #19
    Join Date: Sep 2012

    Location: East Anglia UK

    Posts: 1,219
    I'm Marc.

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    There's a nice simple piece here: http://dbzeebee.blogspot.co.uk/2009/...-of-thumb.html explaining about levels and bitdepth (and why not trying to 'fill' the whole digital meter is going to save a bunch of issues down the line).

    I know many folk don't like Pulse but I'd never heard anything about quality issues, got a link?

  10. #20
    Join Date: Jan 2017

    Location: Bristol

    Posts: 111
    I'm Clive.

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    Nice link.

    I am going off memory from researching the issue about 18 months ago. There is a lot of material out there but a lack of clarity and enough people knocking pulse that I preferred to steer clear of it. ASIO reads direct off the Soundcard and Jack is the Linux equivalent. Pulse is apparently similar to direct sound.

    I was originally doing 16 bit recordings in Windows using direct sound but wanted 24 bit particularly when trying to fix issues using iZotope RX. It became quite clear to me that the 16 bit recordings I had been getting using a standard PC sound card were relatively mediocre. I also had a 24 bit capability with the SoundBlaster but eventually found that recordings using direct sound on that card were the same due to limitations in direct sound.

    I explored 3 options. The first was Linux but I struggled to get it up and running on my old PC. I did not try very hard because I really needed things to be portable which the PC was not. The second option was an external digital recorder as suggested by sq225917 but was not sure which device would meet my needs. The third option was ASIO on a Windows laptop which I got going with the ADL GT40a device and have been very impressed with the clarity of the recordings over what I had before. Since then I have tried WASAPI with the Soundblaster which also gave me a good result.
    Last edited by cre009; 21-01-2017 at 21:13.

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