Gain matching issues

[Thaddeus Smith]

So I bought a Topping D10s to use as a USB to SPDIF bridge...

 

 

between my music server and Yamaha SP2060...

 

 

 

Which then feeds my amps...

 

 

 

Without changing anything else in my system I suddenly found myself with a usable range between 1 and 5 on the PC volume, out of 100. Greater than 5 resulted in possible hearing loss. So I bumped down the attenuation on my amps to just 1 notch off of infinity, which gave me a usable range on the volume slider of about 1 to 10 on the PC. I don't have any meaningful attenuation on the output channels in my DSP, which left me with the option of attenuating the input channels. I found sort of a sweet spot around -20dB on the input signal, which gives me a more usable range of about 1 to 60 on the PC volume slider. Is that the right way to go about this, or is there a better way to achieve the desired results?

 

The input range of my DSP is fairly wide..

 

 

I've read some of those PDF's floating around the web regarding gain matching, I just never seem to know where I should be applying it - and it would seem my particular signal chain of gear is fairly sensitive.

 

[Chris A]

10 hours ago, Thaddeus Smith said:

Is that the right way to go about this, or is there a better way to achieve the desired results?

If the DAC/converter (i.e., the Topping D10s) doesn't have a digital gain control to the S/PDIF output, then the Yamaha is the next best place to do it, and the input channel gain controls is where I'd go so that you're not overdriving the Yamaha's output channels before attenuating the signal.

 

This is good news.  I tried another USB-->AES3 converter last year (from mainland China) that turned out to be a brick (Windows 7 and 9 PCs never recognized it), so I have another path to do it--that's even less expensive:  https://www.audiosciencereview.com/forum/index.php?threads/topping-d10s-usb-dac-and-bridge-review.14859/

 

I assume that Topping DAC is being used to buffer the USB input data stream and the internal clock(s) used to control the USB bus-jitter down to reasonable levels.  This is why I was looking for a clocked/buffered USB-->AES3 (XLR) box in the first place.

 

How are you converting the wiring of S/PDIF to AES3 (XLR)?

 

10 hours ago, Thaddeus Smith said:

I've read some of those PDF's floating around the web regarding gain matching, I just never seem to know where I should be applying it - and it would seem my particular signal chain of gear is fairly sensitive.

Your ears and mind are probably the best resources you've got in terms of setting up the gain structure.  If you're in the middle of the scale in terms of adjustability of the Yamaha's input gains, then I'd say you're there.  Additionally, how's the background noise level?  Are you having to carefully route any cables to keep the noise levels down now?

 

If this works well, then I might have to restart my computer room setup using one with the Yamaha that I've got (AMT-1s sitting on top of Khorn-clone [Shinall] bass bins).  Since the Yamaha can handle it, I might have to try to add a midrange horn/driver between the bass bin and the AMT-1 for grins, and listen for a while.

 

Chris

[Chris A]

This thing (the Topping DAC/converter) would also work well between a PC and a miniDSP 2x4 HD or 4x10 HD to the S/PDIF (TOSLINK optical) interface.

 

Chris

[Thaddeus Smith]

Thanks for confirming Chris. It seemed logical with such a wide variance range on the input channels, but wanted to make sure I wasn't robbing dynamic range.

 

My interconnects between dsp and amps are short and balanced. Speaker wire is longer and grouped together, but tidy. Power cables are short and I try to keep them away from signal lines, but there's no real effort applied to separating everything.

 

I don't hear any background noise.

 

As for the dac to dsp connection, it's just a 1m rca to XLR cable. I spent a good half hour researching and it seems that going AES to spdif is a bit more fiddly and requires proper conversion equipment. Any direction between toslink and AES does as well. Spdif coax to AES however, especially over short runs, is a bit more forgiving if your equipment is robust enough to handle it. Recommendations were simply to try it and see.

 

I did, and it seems to work just fine.

[Bubo]

Digital Source >> Topping USB-Toslonk >> Yamaha Processor DAC >> Amps >> bi-ampd speakers

 

Topping = USB to Toslink optical isolation

should be a buffer for managing the USB incoming data steam for the CRC ARQ

Then a second buffer between the bus and the toslink for outbound packets or framed bits.

I would run all data streams in native mode with no up-sampling. 24bit 96Khz is supposed to be the Holy Grail.....Use a glass cable for the tosslink to minimize bit errors since there is no recovery.

If the clocks are stable on Topping and Yamaha, phase jitter should not be an issue.

 

The Yamaha incoming Optical interface will derive the clock from the inbound signal or framing bits and stream the bits onto the bus, No CRC ARQ on Toslink. Part of the bit stream defines amplitude by changing coordinates per sample or other. I haven't studied all of the various encoding protocols, so I'm not sure how they handle volume and error correction if any. USB is a real data protocol.

 

Yamaha to amp is an analog recreation of the bit stream to the amps. It's functioning as the DAC. It would be interesting to run frequency sweeps through it. The DAC is probably the most significant part of your audio quality in your set up.

 

I would leave the entire digital stream set at 100% lossless and the analog gain attenuater on the amps set at 50%. This should preserve dynamics and allow some use of your volume control while providing the strongest signal to the amp.

 

I keep the gain controls on all of my amps at 12 o'clock aka 50%, so I can have some granularity on the volume control.

 

YMMV

[babadono]

16 minutes ago, Bubo said:

at 12 o'clock aka 50%,

What amps are at 50% when gain controls are at 12 o clock? Just curious.

[Bubo]

On 2/10/2021 at 6:12 AM, Chris A said:

If the DAC/converter (i.e., the Topping D10s) doesn't have a digital gain control to the S/PDIF output, then the Yamaha is the next best place to do it, and the input channel gain controls is where I'd go so that you're not overdriving the Yamaha's output channels before attenuating the signal.

 

Somewhere in the recording process, limiters were already used

Wouldn't 100% be native lossless mode ?

Is there such a thing as digital clipping ?

 

On 2/10/2021 at 6:12 AM, Chris A said:

 

 

On 2/10/2021 at 6:12 AM, Chris A said:

I assume that Topping DAC is being used to buffer the USB input data stream and the internal clock(s) used to control the USB bus-jitter down to reasonable levels.  This is why I was looking for a clocked/buffered USB-->AES3 (XLR) box in the first place.

USB supports CRC ARQ and must buffer

Internal architecture on the bus unknown as is outbound buffer for toslink

If the outbound clock is unstable the whole thing turns to s$$t

 

On 2/10/2021 at 6:12 AM, Chris A said:

 

Your ears and mind are probably the best resources you've got in terms of setting up the gain structure. 

 

If you're in the middle of the scale in terms of adjustability of the Yamaha's input gains, then I'd say you're there.

 

Try lossless and run out of the Yamaha at 90-100%, and see how it sounds ?

 

[Bubo]

8 minutes ago, babadono said:

What amps are at 50% when gain controls are at 12 o clock? Just curious.

My McIntosh MC-2205

Yamaha MX-600, MX-800, M-80, M-85 etc

[Thaddeus Smith]

I'm good on the rest of the setup, thanks. Just needed some clarity about how I was applying input attenuation in the SP2060.

[babadono]

4 minutes ago, Bubo said:

Is there such a thing as digital clipping ?

yep. When signal gets up to 0dBFS there are no more bits to represent a higher level signal. You have clipped your A/D and D/A converters. And it sounds like schiit.

[Bubo]

9 minutes ago, babadono said:

yep. When signal gets up to 0dBFS there are no more bits to represent a higher level signal. You have clipped your A/D and D/A converters. And it sounds like schiit.

 

If I read this correctly

 

Limiting and calibration occur at the point of A to D conversion aka the recording and mastering. this is where this issue appears at encoding, not decoding, it is addressed to avoid clipping the AD converter into an undefinable range aka greater than 1 at the time of encoding. Once it's clipped at AD it's clipped all the way through playback, you can't unclip something that isn't there

 

Interesting possible discussion analog tape being more dynamic than digital recording, but you still have to put it on something to play it like vinyl etc....

 

Thanks to Lavry for their nice write ups

http://www.lavryengineering.com/?cat=10

0dBFS

Overview

The term "0dBFS" is used to describe the absolute peak level of a digital audio signal and is an abbreviation for "zero decibels full scale."

Basics

In digital audio; the possible range of recorded audio extends from digital silence where all bits representing audio signal voltage levels equal zero, to full scale where all bits representing audio signal voltage levels equal one. Digital silence, as the name implies, is the total absence of any audio signal. Full scale is the loudest level an audio signal can reach before some of the original information is lost due to clipping. The effect is very similar to the effect of bipolar analog circuitry clipping the top and bottom off of the audio waveform.

Because PCM digital audio encoding is linear, the most important reference level when recording is the loudest level. In the vast majority of cases, the digital level is displayed with "0db" representing full scale and all other levels represented as a "minus" value in decibels below that level. For more information on digital audio reference levels, please see dB.

PCM digital audio is typically encoded using the Two's Complement system. There are a number of advantages to this approach, which is beyond the scope of this discussion. This system uses all but one of the bits in a digital audio word to represent the voltage level of the encoded audio of either the positive or negative portion of the signal waveform. The other bit represents the “sign” which determines if the voltage was positive or negative at the point in time the audio was sampled.

Overview

The term "dB" is used to describe a ratio between two audio levels. As such; it has no absolute value. Due to the non-linear nature of human hearing, the logarithmic dB scale approximates the relationship of the measured value to the perceived change in acoustic level.

Basics

Please note: due to limitations in supported text characters; in the following discussion a value such as "two squared" is described as "2 raised to the power of 2" for clarity.

The decibel may be defined in this manner: two amounts of power differ by 1 decibel when they are in the ratio of 100 raised to the power of one-tenth. The term was used originally in early telephony to measure loss in a standard mile of telephone wire. In honor of Alexander Graham Bell, and to indicate the "decimal" power relationship; the unit was named the "decibel."

The ratio can be expressed as 10 raised to the power of (dB x 0.1); so a 6dB difference between two signals means the larger signal equals the value of the smaller signal multiplied times (10 raised to the power of 0.6) or "ten to the six-tenths power."

In order to give a dB measurement an absolute value; it must have a “zero reference.” One analogy is temperature in “degrees.” Without knowing what “zero degrees” is; we can only talk about the difference between two temperatures (a form of ratio). Unfortunately; the analogy fails when you bring in the differences between Fahrenheit and Centigrade because (unlike the decibel) the “one degree” has different definitions in each system!

Click here for more detailed information

SPL

One of the more common forms of “absolute” dB scales is Sound Pressure Level or “SPL.” In this case, the “zero reference” is a very small value and the scale only extends in the positive value direction. The zero reference acoustic level is considered to be the “threshold of human perception” and SPL is thus a scale that tells us how much louder a sound level is than the quietest sound one can perceive.

dBu (dBm)

In audio electronics, the “0dBm” standard for power was 1 milliwatt; because most early audio transmission utilized 600 Ohm impedance-matched systems. This power level was achieved when an RMS voltage of 0.775 volts was applied to a 600 Ohm load. In most contemporary audio systems, the signal appears as a voltage waveform; thus the dBu voltage scale is used instead of the dBm power scale. To make this system applicable to pure voltage level measurements; the “dBu” scale is used with the same zero reference of 0.775 Volts.

VU

When the use of VU meters became prevalent in the USA; the VU meter’s “0” was defined as “+4dBm,” and in contemporary systems is defined as “+4dBu.” The designation “VU” is an abbreviation for “Volume Unit” because the VU meter was intended to provide a useful means of relating the metered level of complex audio program to the perceived volume. "0dBVU" = +4dBu = 1.228 Volts rms

dBV

The dBV scale is typically used in consumer audio measurements and ratings. The zero reference for dBV is 1 Volt rms.

“+4” versus “-10”

Professional line level is often referred to as “+4” as versus the consumer line level of “-10”; which is source of confusion. This is because “+4” is “+4dBu” and “-10” is “-10dBV;” and the dBu and dBV scales uses different “zero dB” reference voltages! The result is that, rather than the “apparent” difference of 14dB between “+4” and “-10”; the actual difference is closer to 12 dB (11.8dB).

dBFS

Definition

The term dBFS or “dB Full Scale” is used to describe the level of a linear PCM digital audio signal relative to the highest (peak) level that can be encoded, and has no fixed relationship to dBu or "analog level."

0dBFS

0dBFS is sometimes referred to as “digital clipping” level. Because this is the only “fixed” level in digital audio, the dB level scale typically starts at 0dBFS and all other levels below 0dBFS are in “negative dB” (for example -14dBFS).

Calibration levels

Depending on the calibration of the AD converter used to encode the audio signal; a wide range of analog input levels can result in the same digital level. Because no signal information is retained once the signal level exceeds 0dBFS; the most important level for calibration purposes is peak level.

In a similar manner, there is no fixed relationship between digital level (relative to 0dBFS) and the analog output level of a DA converter.

Professional level analog audio equipment typically has a peak output level of between +18 and +24dBu. Because the standard for VU meters is “0dBVU = +4dBu,” this results in “0dBVU = -14dBFS” for a peak level of +18dBu and “0dBVU = -20dBFS” for +24dBu. The use of “-14” as a calibration standard for digital audio gear thus means the analog input is calibrated to accept a peak analog level of +18dBu. Lavry converters are set at the factory for a reference level of “-20” where “0dBFS = +24dBu.” Which reference calibration level works best depends on the application. For example; the “-14” level works well for Mastered audio program where the difference between “VU” (or “average”) level and peak level has been controlled with compressors and/or limiters. For recording “live” tracks that are not compressed or limited; the “-20” reference level may be a better choice. There is no standard in the professional audio industry for this calibration; only a range of calibrations that are typically used in recording, mixing, and mastering. The important point is to match the peak analog output level of the source device to “0dBFS” digital level.

Ideally; the analog source should have 1-3 dB of “headroom” (peak output level capability) above the analog input level that results in 0dBFS, so that the analog output does not “clip” or significantly increase in distortion before reaching 0dBFS in the AD converter. For example; if the analog source has a peak output level capability of exactly +24dBu, using “-19” or “-18” as the reference level would achieve this goal.

In some cases there are other consideration; such as preservation of dynamic range. For example; the analog source device has an output level control, and turning it to the “full up” position results in a more noise that setting it to a somewhat lower level. The input of the AD converter should be calibrated so the peak output level with the source output level control at the optimum position results is 0dBFS when the audio reaches peak level. It may take some experimentation to determine this level; and changing the source device may require a change to the calibration to optimize the system with the new device.

In a similar manner, the output level of a DA converter can be adjusted so that the same digital level can result in a wide range of analog levels. The DA converter is typically calibrated to the same peak analog level as the AD converter, so that the digital audio recording system is calibrated for “unity gain.” This results in the playback of the digital audio recording system to be the same level as the analog signal feeding it.

 

There may also be an advantage to setting the output level of the DA to a lower level for some applications. One example is when feeding a monitor system with high gain. By reducing the level at the output of the DA converter, the monitor Volume control can be operated at a higher, more ideal setting. Lower gain settings can also allow matching of the apparent level when switching between sources.

 

http://www.lavryengineering.com/wiki/index.php/DB

 

 

[babadono]

@Thaddeus Smith so signal comes out of the 10S on the yellow connector S/PDIF and goes into an AES XLR input on the Yammie? With just an RCA to XLR adapter?

[Thaddeus Smith]

32 minutes ago, babadono said:

@Thaddeus Smith so signal comes out of the 10S on the yellow connector S/PDIF and goes into an AES XLR input on the Yammie? With just an RCA to XLR adapter?

 

Correct. Like I said earlier, I didn't kill too much time researching before deciding to just try it. Here's a couple of threads that prompted me go to that route since I wasn't dealing with BNC or Toslink.

 

https://www.gearslutz.com/board/connectors-cables-stands-and-accessories/1137971-spdif-coaxial-aes-cable.html

 

https://www.gearslutz.com/board/so-much-gear-so-little-time/1022735-spdif-gt-aes-cable.html

[Thaddeus Smith]

Also, my Yamaha limits the digital input to 24/96 so I just set those upper thresholds in Roon for playback. Everything at or below those thresholds play natively and everything above get's downsampled. I think I have maybe 6 albums which are 24/192.