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Weiss Enginering Ltd.

DSP501/DSP502

4.5

Digital level control and Dithering

Digital Level Control – done the right way. With sound examples.

In high-end HiFi circles a level control done in the digital domain is often viewed as being inferior to one
operating in the analog domain. Let’s look on how a digital level control works and why it can be an excellent
solution if it is properly implemented.
A level control is a multiplication of the audio signal with a constant, the ”gain factor”. The gain factor usually
is in the range of zero (signal fully off ) to one (signal untouched).
A factor of

0.5

then means that the audio signal is attenuated to half of its amplitude. What exactly happens

when we multiply two numbers?
If we e.g. multiply a 2 digit and a 3 digit number, the resulting number can be up to 5 digits long (the sum 2
plus 3). As an example: 30 times 500 equals 15000. 2 digits times 3 digits yields a 5 digit result.

In digital audio, the numbers are represented in the binary system, not the decimal system. A decimal
number consists of digits 0 through 9, a binary number of digits 0 and 1.
So a binary number may look like this:

1011 0011 0101 1101

This is a 16 digit or 16 bit binary number, the grouping into 4 bit chunks is for better readability. The audio
samples on a CD are represented with such a binary number system with each sample value represented
with 16 bits.
Now let’s assume we have an 8 bit gain factor for a level control. If we apply that to a signal coming from a
CD we multiply an 8 bit gain factor with a 16 bit sample value. The result is up to 24 bits long (the sum of
the word-lengths of the two factors).
An example:

0100 1001 x 1001 0110 0111 1011 = 0010 1010 1110 1001 0001 0011

The question now is what do we do with the 24 bit long result? The digital to analog converter which converts
the samples after the level control may only be capable to handle 16 bit wide samples. Thus what should
we do with the excessive 8 bits? The simplest solution is to truncate the 24 sample to 16 bits, i.e. to cut off
the 8 less significant bits. The truncated 24 bit result above then would look like this:

0010 1010 1110 1001

i.e. the first 16 bits of the 24 bit result above. The remaining bits

0001 0011

are discarded. If these bits are discarded, an error is introduced. This error is called a quantization error,
because the 24 bit result is requantized to 16 bits.
Unfortunately the quantization error is part of the audio signal – and if we take that part away from the signal,
the signal undergoes some distortion, the so called quantization distortion.

The sound example at the link below shows how such a distortion sounds. In this music example a 16
bit signal is truncated to 8 bits. 8 bits in order to clearly show the effect. Notice how the noise (distortion) is
modulated by the music signal.

https://www.weiss.ch/linked/digital-level-control/nodither.mp3

This is how a badly implemented digital level control works....
Fortunately there is a better way to handle the re-quantizing. One solution would be to use a Signal Proces-
sor with a higher word-length, e.g. a 24 bit converter, to accommodate for the 24 bit samples coming out of
the level control. This of course would already help a lot, but there is another technique: dithering.
The idea about dithering is to de-correlate the quantization error from the audio signal. As we have seen in
the example above, the quantization error depends on the audio signal, i.e. it is correlated with the audio
signal. On the other hand, if dither noise is added to the 24 bit sample after the level control and before
the re-quantization to 16 bits, the quantization error can be fully de-correlated from the signal. This means

User Manual and

26

White Papers

Summary of Contents for DSP501

Page 1: ...DSP501 DSP502 User Manual and White Papers Software Version 2 5 1r2997 Date February 15 2022...

Page 2: ...Headphone 14 2 3 12 Snapshot Storage 16 2 3 13 Global Presets 17 2 3 14 Firmware Update Download 18 2 3 15 M S Control 18 3 Technical Specification 19 4 Signal Processing Algorithms 21 4 1 The Weiss p...

Page 3: ...0 year history in Digital Signal Processor design In that time span we have learned a thing or two about algorithm design The DSP50x is the essence of our experiences Figure 2 The DSP501 The DSP502 fi...

Page 4: ...gs This opens a menu which allows to select one of the DSP algorithms to be changed Note the rotary knob allows to scroll through the menus The algorithms can be operated via the touch screen but we r...

Page 5: ...the DSP Presets is given in the web interface description below Figure 3 IR remote control 2 3 The Web Interface As last but not least possibility to operate with your DSP50x we offer you the Web Inte...

Page 6: ...Weiss Enginering Ltd DSP501 DSP502 Figure 4 Web interface overview User Manual and 4 White Papers...

Page 7: ...the high frequency content of music So this plugin can be beneficial if your favourite recording contains some unpleasant harsh sounds But sharpness can also arise during mastering or as a results of...

Page 8: ...ted To switch a plugin on off just tap on the enable bypass tab The selection is highlighted in red By moving the slider you can adjust the user parameter Saturation The Saturation parameter adjusts t...

Page 9: ...izer plugin A little example should give you an idea of the functionality and the manifold opportunities the EQ offers The 1st band is set to low shelving with a boost of 0 5dB at 147 Hz i e frequenci...

Page 10: ...nsider the following instructions on how to set up the Room EQ with a simple procedure without using any measurement In an alternative setup scenario we will use the room measurement software of the I...

Page 11: ...ries you may visit this website and calculate the potential room mode frequencies by entering the dimensions of your listening room https amcoustics com tools amroc This can help with determining the...

Page 12: ...ction without boomed bass Note you may have gotten used to the acoustics of your playback setup and then when correcting it with the Room EQ it may sound strange at first But keep it and listen to it...

Page 13: ...32 105 42 32 130 106 41 33 128 107 41 34 126 108 40 35 124 109 40 36 122 110 39 37 120 111 38 38 118 112 38 39 117 113 37 40 115 114 37 41 113 115 36 42 111 116 36 43 109 117 35 44 108 118 34 45 106 1...

Page 14: ...wede In acoustics this loudness describes the subjective perception of sound pressure1 Schwedes EQ design then considers this subjective perception to create a more pleasant playback regarding the sou...

Page 15: ...rall the same loudness impression To adjust how strong this dynamic range reduction should affect the playback you can set the Dynamic Level If you want to have less dynamics left in the output choose...

Page 16: ...ers and Distance Listener faders If the geometry is not very well suited for the XTC playback the background of the two fader sections turns yellow The XTC may still work but may be compromised Note t...

Page 17: ...phone plugin allows adjusting the Amount level between 0 and 100 in order to achieve your preferential setting A higher Amount results in stronger impact of the plugin s effect on the played back audi...

Page 18: ...so part of its setting and will be stored in the snapshot Figure 14 Snapshots maintenance In order to recall a stored snapshot select it by name from the list and tap on Snapshot Load Store a plugin s...

Page 19: ...SP algorithms are bypassed The plugin sequence within the composer matrix reflects the actual sequence of the DSP processing Pre set snapshot dropdown lists of plugins which are currently not applicab...

Page 20: ...onnect to our Server to see whether there is any new device firmware available The DSP50x has to be connected via Ethernet to your internet router If new firmware is available the pad allows to downlo...

Page 21: ...200 240 V automatic voltage selection Fuse rating 500 mA slow blow Power consumption 10 VA max Power consumption in standby 2 2 VA max Size DSP501 Depth 30 cm Width 18 8 cm Height 6 6 cm Height with...

Page 22: ...BU outputs Sampling frequency selectable between 88 2 kHz 96 kHz 176 4 kHz 192 kHz Toslink input 44 1 kHz 48 kHz 88 2 kHz 96 kHz Input word length max 24 32 Bits Digital Outputs 2 RCA Connectors 2 XLR...

Page 23: ...dependent bands Each of them can be set to a mode namely low cut high cut to get rid of disturbing low or high frequencies low shelf high shelf to boost or cut low or high frequencies or peaking to bo...

Page 24: ...low cut filter also called high pass filter is displayed Figure 21 Low pass and high pass filter scheme To finalize this section you find an example chart that visualizes examples of frequencies for v...

Page 25: ...y that frequency then can be suppressed to some extent with the room EQ See section 2 3 6 for instructions about how to find room modes with a slow sine wave sweep If a room mode is very pronounced th...

Page 26: ...e on headphones The main impressions you will get with the XTC based playback are Large stereo stage much wider than the space between the speakers A feeling of depth i e a 3 D like presentation Very...

Page 27: ...additional resonance frequencies specific noise at various frequencies specific crosstalk between left and right channels specific effects caused by the RIAA emphasis specific amplitude modulation ef...

Page 28: ...it long result The digital to analog converter which converts the samples after the level control may only be capable to handle 16 bit wide samples Thus what should we do with the excessive 8 bits The...

Page 29: ...it converters no question that a level control with a 24 bit word length easily rivals the best analog level controls By the way 24 bits means 16 777 216 quantization steps The last example below togg...

Page 30: ...should effect the audio signal To be able to use the plugin properly it is helpful to understand how the levels of the plugin effect the audio signal There are two main levels the Dynamic Level and th...

Page 31: ...different recordings with a wide range of loudness levels choose a lower and if needed negative Dynamic Level The lower this level is the more the dynamic content is limited The value 0dB is our defa...

Page 32: ...l untouched This idea is based on the so called m s stereophony Which is a coding process that transforms the signals of the left and right into a different representation The mid signal is the sum of...

Page 33: ...define the Amount of the chosen De Essing effect Independently a special normalization is performed to create a result that is individually matched with the current sound of the music This is needed o...

Page 34: ...ow the different modes sound For the parameter Amount a good starting point could be around 6dB Futhermore in the beginning it is more easy for getting to know the sound of the De Esser and finding yo...

Page 35: ...f such a headphone frequency response measurement is plot ted in red In reality such measurements are far more complex This exemplary headphone only creates an amplification imbalance in two frequency...

Page 36: ...cert hall So Schwede s concept tries to overcome these non linearities of the human sense of hearing with a equalizer design which creates an optimized compensation curve based on the psychoacoustic a...

Page 37: ...Weiss Enginering Ltd DSP501 DSP502 5 Contact Information Weiss Engineering Ltd Daniel Weiss www weiss ch weiss weiss ch User Manual and 35 White Papers...

Page 38: ...treatment overview 22 23 Dummy head microphone 23 24 XTC scheme 24 25 Block diagramm of the Weiss Vinyl Emulator 25 26 Image processing original left quantized without dithering middle quantized with...

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