Sunday, August 2, 2020

On Keyboards

I want to tell a story about my current computer keyboard. A couple of weeks ago I switched to Kinesis Advantage2 keyboard. This device looks very unusual compared to most regular keyboards:

As you can see, the design brings the ergonomics to the extreme and fulfills several goals:

  • provide integrated wrist rests;
  • make better use of thumbs by putting more keys under them;
  • achieve more natural wrist positions;
  • provide similar path length from main 4 fingers to alphabet keys.

However, with this design a lot of keys were moved away from their usual positions. The most notable differences are for the arrow keys, brackets, and the keys that are located at periphery positions on traditional keyboards like functional keys, tilde, and plus.

I'm not new to ergonomic keyboards. For a very long period I used another keyboard by Kinesis called "Freestyle2":

As you can see, this keyboard is also ergonomic—it consists of two halves that can be positioned at various angles. However, its layout is more traditional and even somewhat superfluous—I've never made a good use of the "shortcut" keys on the left (mostly because I use actual shortcuts for all the actions these keys intend to provide, and I can't use them for anything else as the keyboard doesn't allow for remapping).

I'm in fact a big fan of keyboard shortcuts and learn them in any application I use more or less frequently. I can't imagine using GMail without shortcuts—at my job I receive from 100 to 200 emails per day. While I was still using Freestyle2 I started noticing that my right hand is aching after I had been using the bracket keys while going through my inbox (the bracket keys are used in GMail to archive the current email and go to the next one) and after active use of the arrow keys for navigating in Emacs.

The Advantage2 keyboard was sitting in my drawer for a long time (at the beginning of the COVID lockdown I had managed to get it at my company's expense) but I was hesitating to start using it due to its unorthodox layout. And as I've got the hand ache from using Freestyle2 I decided that it's a good opportunity to give Advantage2 a try.

I must admit, it took about 2 weeks until I could finally use it without an extra mental effort for finding keys. The most infuriating was "re-learning" the key combinations in Emacs. For a lot of them over the years I simply forgot what the actual keys are and was invoking them using muscle memory exclusively. Since the key layout on Advantage2 is different, I had to recall what are the keys used for the combinations and had to learn where to find them on this keyboard.

Now I can clearly see benefits of using Advantage2: no wrist or arm aching and typing feels more comfortable than on a traditional keyboard. Also, unlike Freestyle2 the Advantage2 keyboard is programmable, meaning that you can remap keys and even record macros. I actually did a couple of key remappings to make my typing more comfortable.

Key Remappings

First, I remapped the back space key (it's under the left thumb, mirroring the space key under the right thumb) to serve as the second space key. This is after my habit of having two separate space keys on Freestyle2—I noticed that I end up pressing the back space key on Advantage2 when I wanted to type a space.

OK, but the back space is a really useful key after all—where should it go? I moved it to the nearby "Delete" key. It is still convenient for an active operation. Good, but "Delete" is also useful in various file managers as it allows deleting stuff. I found that on the left side Advantage2 has an "extra" key—another copy of the backslash key:

It's also used as the "Insert" key when you turn on virtual numeric keypad. So I thought it's a good semantical link between "Insert" and "Delete", and assigned the latter to this extra backslash key.

Finally, if you use Unix shells a lot you'll notice that the tilde key is used very frequently. On traditional keyboards it's located to the left from the number row. On Advantage2 for some reason this place was given to the "+/=" key and the tilde key was moved to the bottom:

The natural solution is to swap these two keys. So this is the layout I ended up using:

And yet another cool feature of this keyboard is that all remappings and macros are available as text files exposed on keyboard's internal USB drive. For example, this is how my remappings are specified:


This is what makes this keyboard a truly professional one.

Practicing Typing

For a long time I practice my typing on Tipp10 site. I know, there are myriads of similar touch-type training sites. What I like about Tipp10 is their scoring system which penalizes for typos. This stimulates you to type slower but with less mistakes and build up speed slowly. The site also allows for uploading your own typing lessons.

Building typing speed is something that doesn't come easy for me. I'm trying to find any book or instruction that would be based on a research on neuro-motor skills, but so far found none. The closest thing I found is this free book on playing piano called "Fundamentals of Piano Practice." It gives some practical advises on the exercises for developing finger muscles and improving finger coordination, which helps to build up speed and accuracy.

Of course, typing and playing piano are very different activities, however I believe that they share some goals. I've learned a couple of things from this book.

First is what they call "Hand Separate" practice. I've found that my right hand is somewhat weaker and less accurate than the left one, partially because I'm left-handed, and also because I had a minor injury of my right hand resulted from winter biking. So it helped to train this hand more intensely. This is where the design of Advantage2 is a real advantage—the hand zones are clearly separated.

The second technique is what the book calls "Parallel Sets" (PS for short). Their strong side is that they can serve both as diagnostic tests for the finger movement fluidness and as an exercise for developing it. The practice of the parallel sets is described to length in the book, so I will not repeat it here. This is just an example of custom typing exercises that I had created for myself in Tipp10 (for QWERTY layout):

PS Exercise 1

aaaa x8, jjjj x8, ssss x8, kkkk x8, for all front row keys.

PS Exercise 2

asdf fdsa asdf fdsa asdf fdsa asdf fdsa
as as as as sa sa sa sa
sd sd sd sd ds ds ds ds
df df df df fd fd fd fd
asd asd dsa dsa asd asd dsa dsa
sdf sdf fds fds sdf sdf fds fds

And similar sequences for the right hand. There is also a variation where I type left and right hand interleaved, for example:

ajsk ajsk ajsk ajsk ajsk ajsk
skdl skdl skdl skdl skdl skdl

PS Exercise 3

ad ds sf ad ds sf ad ds sf ad ds sf
ads dsf ads dsf ads dsf ads dsf
adsf adsf adsf adsf adsf adsf adsf adsf

And similar sequences for the right hand. This exercise is actually challenging. I've found that it really helps to do it for each hand in isolation first.

In fact, all these exercises help to feel difference between keyboards. I was doing them in parallel on Advantage2 and Freestyle2 keyboards, and I immediately felt the convenience of curved hand wells of Advantage2 that helped to make the glide of fingers more fluid.


I'm glad that I have switched to this keyboard. Among positive sides I would specify:

  • great ergonomic design;
  • flexibility in configuration which makes this keyboard a real professional tool.

And on the flipside:

  • unorhodox layout which can interfere with muscle memory developed while writing program code and using code editors. This isn't a bummer, just requires some time for updating your muscle memory;
  • high price (but good quality).

I think Kinesis had understood these problems and at some point they have introduced "Freestyle Pro" keyboard which has the design of Freestyle2 and the customization engine of Advantage2. It is also cheaper than Advantage2, so seems like a good alternative to me.

Thursday, July 9, 2020

DIY Headphone Equalization


Back in time I already experimented with commercial packages for headphone equalization: Morphit by Toneboosters and Reference by Sonarworks. They offer means for correcting the frequency response of selected models of headphones in order to bring the sound closer to either a "reference" target curve or to the sound of some other model of headphones.

Although I still own Morphit I decided to try to devise my own way for headphone equalization. I had the following reasons for this:

  1. Although the range of headphone models measured by Toneboosters is quite wide, some of the models I do use: Audeze EL-8 Closed and Open are missing.

  2. I intend to use parametric equalizers with limited number of filters available—the equalizer built into MOTU AVB card. I also want to avoid using computers to save myself from the noise of their fans.

  3. I wanted to be sure that I apply correction that is relevant to my pair of headphones and to my head. Sonarworks emphasize that there are variations between different pairs of the same model and offers a service to measure your pair, or to sell you a pair of headphones which they have measured on their rig. I would also add that the low end performance of over-ear cans would differ depending on the state of the pads and the shape of the head of the person wearing them.

Thus my plan was to perform my own measurements of the headphones I have and try to bring their sound closer to each other. The practical task I had at hand was to bring the sound of Beyerdynamic T90 and Shure SRH1540 to the sound of Audeze EL-8 I mentioned above. Why to do that? One word—comfort. Let's compare how much do weight the headphones I own (with cable):

Model Weight, grams
Audeze EL-8 Closed 576
Audeze EL-8 Open 533
Beyerdynamic T90 394
Shure SRH1540 331
Massdrop 6XX 317
AKG K240 Studio 290

Although I really love the sound of EL-8 (both variants) their weight is killing me! So my plan was to equalize T90 to sound like the open model and SRH1540 to sound like the closed model. I chose T90 and Shure because their sound feels as "spacious" as in EL-8s, and it's only their tuning that feels wrong to me: T90 have too much highs, and it's unnatural and fatiguing, while SRH1540 have a very pronounced V-shaped tuning which I wanted to "flatten".

Measurement Methods

I used two methods I had previously mentioned in this post: moving microphone averaging (MMA) and on-head measurement using Sennheiser Ambeo Headset. Since the time I've made that post I've got a couple of updates on them.

Clock Drift with Ambeo Headset

One problem I had with Ambeo Headset is that due to lack of external synchronization use of Ambeo's ADC for input and an external DAC for output resulted in clock drift which creates a spectral shift when doing long (lasting several seconds) sweep measurements.

I have partially solved this problem by using the feature of Mac OS X audio system called "Aggregate Device". It allows combining two or more digital audio devices into a single one, and what's important, takes care of synchronizing their clocks:

This actually fixes the problem with shift of the measured amplitude across spectrum. However, because the drift correction only happens at certain periods, there is still phase shift occuring, especially at high frequencies. Due to this care must be taken when averaging multiple measurements.

MMA and Ambeo On-Head Methods Accuracy

Recall that I tried using the MMA method for reverse engineering the equalization applied by Audeze Cipher Cable for EL-8. Recently I cross-checked these measurements by performing an electrical measurement of this cable into a dummy load. Here is the FR graph acquired by QA401:

It confirms that there is a bass boost, but no other modifications, whereas my MMA measurements were also showing a "scoop" at middle frequencies. So turns out that scoop is a measurement error.

I decided to figure out the accuracy and usable range of both methods by doing several measurements in a row, averaging the result, then repeating the same process and comparing the averages. For the MMA method I came up with the following graph:

As we can see, the usable range is from 200 Hz and the variance is within +/- 0.5 dB. While on-head measurement using Ambeo headset gives the following:

So, the usable range is up to 4 kHz with the same variance. That means, the range from 200 Hz to 4 kHz can be used for judicious merging of the curves. Note that the resulting curve can't be compared with measurements obtained using standard headphone rigs. However, it can be used for comparing the tuning of various on-ear headphones and deriving equalization between them.

Note that even measurements done using different "standardized" head and torso simulators still can't be compared directly, as it can be seen from the past exchanges between Tyll Hertsens (ex-Innerfidelity) and Head-Fi, and Audeze.

Also note that before averaging the measurements done using Ambeo Headset I had to convert them to minimum phase first, because the clock drift I mentioned above produces skewed phase:

Thus we can only average the magnitude data. But that's OK considering that the MMA method, being a single channel measurement also provides magnitude data only.

The Equalization Process

This is the process I used:

  1. Obtain an averaged measurement of the headphones' left driver from 5 MMA measurements. I was performing these measurements by waving slowly the headphone earcup playing pink noise in front of the Beyerdynamic MM-1 microphone while capturing RTA 1/48 octave measurements in REW with infinite aveaging until I've reached 100 averages.

  2. Obtain an averaged measurement of the same driver from 5 on-head reseatings, this time using a 1M measurement sweep from the headphone into Sennheiser Ambeo Headset microphone.

  3. Merge the averaged measurements somewhere between 1 kHz and 2 kHz. I understand that this brings uncertainty in the process. However, the point can actually be found by comparing the slopes of the two curves. I applied 1/12 octave smoothing to both curves to simplify the process.

  4. Repeat the same process with another pair of headphones.

  5. Calculate the equalization curve by performing "A / B" operation in REW, where A is the curve of the target headphones and B is the curve of the headphones being equalized.

  6. Approximate this curve by adjusting PEQ in MOTU AVB equalizer.

Now here is an example of applying these steps to equalize Beyerdynamic T90 to sound like Audeze EL-8 Open back.

Below are the averaged graphs obtained for T90 from MMA and on-head measurements:

The merge point is at 2 kHz. And below are the averaged graphs for Audeze EL-8 Open:

Note that the 5–8 kHz drop that Tyll and Audeze were arguing about is absent from the MMA measurement. Whereas the notch around 5–6 kHz seen in on-head measurements done using Ambeo headset is the artefact of this headset, it appears on almost all measurements I've done using this technique.

And after we have obtained two merged curves it's time to divide them and obtain the suggested equalization curve. Below this curve is superimposed with the actual curve I've ended up using applying using the parametric equalizer of MOTU AVB. It doesn't follow the suggested curve precisely, it's a compromise that takes into account capabilities of the DSP on MOTU and my subjective judgement obtained by fast switching between these headphones on various tracks:

As we can see the equalization does shave off some high frequency from the factory tuning of T90 making them sounding more neutral.

And the similar procedure has been applied to equalize Shure SRH1540 to sound more like Audeze EL-8 Closed back. The suggested and resulting equalization curves are below:

The major correction here is to "straighten" the V-shape of the factory tuning.

The downside of the equalization done using IIR PEQ filters is non-uniform group delay. Ideally we would want to use linear phase filters. This is something to consider for future improvements.


Personally I liked the result of re-tuning which allowed me to combine the comfort of one pair of headphones with the sounding of another pair. As with any equalization we need to understand its limits. Of course, the properties of the drivers used in the headphones have great influence on the perceived sound "quality". As an example, initially I tried to make Massdrop 6XX to sound like EL-8 Open (recall that 6XX are second lightest headphones among those I have), however I was missing the sense of spatiousness and of having the soundstage wider than the headphones. T90 replicates this feeling better.

What are the alternatives for the approach I used here?

  1. Use ready-made equalization toolkit like Morphit. Pros are obvious: you just select the source and the target headphones and start "morphing". However, its database is not complete, and after all, you don't know how close your two pairs of headphones are to their measurements.

  2. Use some database provided by an enthusiast. Similar to the previous one but this time the equalization curve must be derived by yourself. This is doable if the database provides the data in some numerical format, not just pictures. Here the same concern about the variability between headphone instances applies. Be sure not to mix measurements provided from different sources. I know it appears compelling to compile graphs from various sources in order to create the most complete database, but this just doesn't make sense since curves from different rigs are not directly comparable. Reading this post by Crinacle explains a lot of things that can deviate from rig to rig.

  3. Order calibration for your headphones from a company like Sonarworks. I would do that if I were using headphones for professional music production, but for entertaiment purposes this seems like an overkill. Also, at least Sonarworks provide calibration data in a proprietary format only usable with their software. And fulfillment of measurement orders takes weeks, so it would be wise to approximate the equalization on your own before committing to that.

  4. Buy or build a complete measurement rig. Yes, that's the way to go if you intend to perform measurements and equalizations routinely. Here the question is about the reliability of measurements. For example, I've heard from owners of the miniDSP EARS rig that it's not very accurate at high frequencies, thus either a lot of averaging is required, or use of a different method like MMA is needed for obtaining measurements in that region. Whereas rigs from GRAS and other established measurement companies are pricey.

Saturday, June 20, 2020

Marantz AV7704 as Audio Hub

I have a Marantz AV7704 A/V receiver that I was using for some of my work projects. I know Marantz well for their classic "Hi-Fi" equipment: CD players and receivers. Originally an American company, it was acquired by its Japanese competitor Denon, forming a "D&M" holding. Then the holding was bought by "Sound United" which now owns "Classé", "Denon", "Marantz", and "Polk" brands. We can only hope that all these corporate games didn't degrade the quality of the products.

Up until the last month I was considering this receiver for work usage only but lately I decided to give it a bit more use and hook it up for my daily listening. My goals were:

  • eliminate computers and any other equipment with fans from the playback chain;
  • have convenient remote controls;
  • ensure that audio path is clean and works to its full performance.

So let's see how this receiver performs. The documentation on its technical capabilities is quite scarce, it will be useful to fill up missing information on measurements.

First Look

This is quite a versatile receiver. If you look at its back panel there is no shortage of inputs and outputs:

AV7704 supports 3 audio zones, and its remote has 14 buttons for selecting an input. The inputs utilize a range of technologies from good old analog to digital wireless. This is somewhat overwhelming. I decided to wear a consumer hat first and see what functions I can utilize.


My usage of AV7704 is 95% for audio playback. I have the following "use cases":

  • streaming lossy stereo audio (Google Play Music and YouTube);
  • playing lossless stereo audio from a home server (FLAC files);
  • playing surround audio from a home server (DTS, MKA and MKV files).

Currently I have a stereo setup but nevertheless I enjoy listening surround re-issues of famous albums downmixed into stereo for headphones (binaural). Sometimes surround remixes reveal background details that I missed on the original stereo mixes.

What AV7704 can offer to me? It has a built-in HEOS player which supports some streaming services and Internet radios, however Play Music is absent from the list. Not a big problem—I have a Chromecast HDMI dongle and NVidia Shield TV Pro set-top box that I can connect to HDMI inputs.

Playing local stereo is of course supported by HEOS, and the most convenient way for making files from a local server to be accessible to HEOS seems to be via a Plex server. In theory HEOS can connect to network shares directly, but I couldn't make it work.

Unfortunately, HEOS doesn't support surround audio files and neither does Chromecast. Shield comes to the rescue offering a Plex client and VLC apps. Both support "pass-thru" mode for sending encoded surround audio to the HDMI output of Shield directly.

AV7704 also has support for Bluetooth and AirPlay. However, Bluetooth is obviously lossy and limited to stereo, and AirPlay requires using a computer or an iOS device—not my option.


Here I've got somewhat atypical demands. I need optical output to feed the miniBox for LXminis and the subwoofer, with volume control! I need parametric equalizers for interfacing SPL Phonitor mini headphone amplifiers.

This is where AV7704 falls short for me—it only offers HDMI outputs and analog outputs (line and headphone), no SPDIF. Also, the EQ on this unit is a classic "Graphic EQ" with fixed bands and no adjustment for the "Q" value of filters. It's good that this receiver at least offers tone controls, I will need to use them when playing some records.

It is possible to split off SPDIF from an HDMI output by using one of numerous "HDMI Splitter" boxes. I was considering that until I discovered that AV7704 only offers volume control on its analog outputs—not when sending audio via HDMI.

Failure? Not really—I have a trump in my sleeve—MOTU Ultra Lite AVB card which I was previously using for my surround setup. This card has 6 high quality line inputs, DSP, and both analog and digital outputs. So I can use to complete the HDMI receiver and Dolby / DTS processing functions of AV7704, great! And MOTU AVB can work on its own, without a computer, thus my initial requirement is still fulfilled.

The remote control requirement is fulfilled by AV7704, the companion HEOS app for Android, and obviously other apps on Android that can work with Chromecast.


This is how I hooked things up:

I decided to use Zone 1 output for headphones (driven by SPL Phonitor minis). XLR outputs of AV7704 connected to inputs of MOTU AVB for headphone equalization. Since the headphones are connected to Phonitors which have volume controls, I don't need to control volume on AV7704. So potentially I could send audio to HDMI, split it out as SPDIF and send that to MOTU optical input. I considered that option but found it inconvenient because first, this will require adding yet another electronic box to the configuration, and second, this will force MOTU AVB to be clocked at the same sampling rate as HDMI audio, which is normally 48 kHz. So using the analog XLR output is more robust, although it adds an extra D/A->A/D conversion.

Specifically for surround downmixes I would prefer to use the headphone output of AV7704 (HPH on the diagram) because typically there are differences in how Dolby and DTS downmix to speakers vs. headphones since the latter offer much better channel separation.

And Zone 2 output (only RCA is offered for it) is used for LXminis. Since the miniBox is about two meters away from AV7704 and is powered from a different outlet I decided to use optical connection between AV7704 and miniDSP in miniBox for full isolation and less noise. For the speakers I have to use the volume control of the AVR so the analog output is the only option here. In order to convert analog into TOSLink I also use MOTU AVB.

With all these extra D/A and A/D conversions and use of analog outputs on AV7704 it is important to verify that there is no signal quality degradation due to noise, output or input overload, or any other issue. Also, AV7704 offers options like "direct" output which clams to provide "purer" output and I'm curious to validate these claims.


Digital Inputs

Need to recall two issues that can happen with digital recordings that do not leave enough headroom due to aggressive mastering. The first is clipping of intersample peaks during resampling. The problem illustrated below:

If a record is digitally mastered in a way that puts non-peak values of waveforms to maximum (or minimum) values of a particular integer representation (16-bit or 24-bit integers), then resampling can yield values that are outside of the domain of the integer representation, which means clipping. This is what I have encountered with Google Nexus Player with its mandatory resampling of 44.1 kHz content to 48 kHz.

Presence of this problem can be detected purely in digital domain by capturing the digital output of the player. I decided to check Chromecast, HEOS, and Shield whether they have this issue. For that I used the same test files as back in 2017: a sine wave phase shifted by 45° and "normalized" to 0 dBFS digitally.

Recall that this isn't just a DSP geekery but rather a real issue encontered in commercial CD recordings that were engineered to sound "louder".

This is the test setup I used:

I was capturing the digital output digitally by sending audio to HDMI and using a splitter. The optical output from the splitter was captured by MOTU AVB. What I've found is that Chromecast and HEOS do not attempt to resample the input signal and hence do not clip it, whereas Shield Pro always opens the HDMI output at 48 kHz and resamples 44.1 kHz inputs to 48 kHz with clipping. Thus, the conclusion is—avoid using Shield Pro for music playback except for encoded surround audio which is sent to the AVR directly for further decoding, or if you are sure the audio is at 48 kHz already.

I also checked if I can ditch HEOS in favor of Chromecast for local playback too, but quickly discovered that VLC can glitch when casting to Chromecast, while HEOS always plays flawlessly.

AV7704 Analog Outputs

What I wanted to verify is whether the quality of the XLR, RCA, and the headphone output of AV7704 are on par with each other. I used Cambridge Audio DacMagic Plus as a reference. I verified that its XLR and RCA outputs in fact have the same linearity and I was expecting the same from AV7704.

However, as I started measuring I found that the RCA output of AV7704 is much noisier than XLR. The fact that the noise was fluctuating as I was touching the unit's screws at the back lead me to the conclusion that it is missing proper grounding. Indeed, the power input of AV7704 is two-pronged so the enclosure if "floating". I can understand why the manufacturer has done that—it's in fact typical for consumer equipment which normally uses unbalanced connections and thus there is a high chance of creating a ground loop. However, instead of simply not grounding the enclosure I would prefer to have a "ground lift" switch as the last resort for solving ground loop issues.

After I grounded the box by connecting a copper wire to one of the screws on the back with one end and to the power strip enclosure on the other, the noise situation has become much better and indeed XLR and RCA started showing similar performance. It seems that Cambridge Audio DAC is engineered better than AV7704 since it performs great without requiring to be grounded.

As for the headphone output, I measured its output impedance and found that it's quite high—39 Ohm which means it can only damp well headphones with high input impedance—300 Ohm or higher. Recall that I plan use the headphone output for surround renderings, and my preference is to use IEMs in this case, as they have less interaction with my ear pinnaes. Since IEMs typically have very low impedance, I ended up connecting the headphone output of AV7704 to the line input of MOTU AVB which constitutes a perfect load for this headphone output.

Yet another thing to consider is what is the optimal output level from AV7704. This receiver in fact provides several options here:

  1. Attenuated output: 0 dB down to -79.5 dB.

  2. Amplified output: 0 dB up to +18 dB in case if the digital program level is too low.

  3. Pure Direct output mode which bypasses processing circuitry and turns off all analog video circuits in an attempt to lower the noise.

The hardware test setup was essentially my playback setup. I only added one extra connection: TOSLink output from MOTU into AV7704 "CD" input. Here is how XLR, RCA (Zone 2), and the headphone output are seen by MOTU AVB when the output level on AV7704 is set to -6 dBFS. I was using REW tone generator to produce a sine tone of 1 kHz at maximum dBFS:

As we can see, the headphone output (red) has the highest output level and also the highest level of noise and harmonics. It's interesting that only the headphone output has a small spike around 60 Hz which didn't went away after I grounded the receiver.

The most linear output is XLR (green). It seems that -6 dBFS is the sweet spot for it, as reducing attenuation to 0 dBFS significantly degrades its linearity and in "amplifying" modes performance is unacceptable.

I was curious whether "Pure Direct" mode can deliver better performance for Zone 1 outputs, however the results practically didn't change at all. However, I don't use analog video inputs and outputs (I'm curious who would these days), so perhaps there is no interference from them in the first place. To me, the "Pure Direct" mode looks like a heritage of the old days, and I would prefer Marantz to remove the analog video I/O at all rather than adding this mode.

In contrast, the Zone 2 RCA output (red) provides better S/N ratio when amplified (at the cost of a slightly higher distortion), but only up to a certain point. For it, +9 dBFS is the frontier of linear behavior.

The summary of THD and noise for different outputs of AV7704 is in the table below. Note that I ran MOTU at 96 kHz sampling rate and didn't use a low-pass filter, thus the THD and noise figures are across the whole range up to 48 kHz.

Output, mode 1 kHz RMS (Z) THD Noise THD+N
Z1 XLR, -6 dB -13.2 dBFS 0.0019% 0.0021% 0.0029%
Z1 HPH, -6 dB -9.2 dBFS 0.0018% 0.0027% 0.0033%
Z2 RCA, -6 dB -21.8 dBFS 0.0034% 0.0063% 0.0072%
Z1 XLR, 0 dB -7.2 dBFS 0.012% 0.0044% 0.013%
Z2 RCA, +9 dB -6.7 dBFS 0.0047% 0.0036% 0.0059%

The official specs of AV7704 specify distorion at 0.005% over 20 Hz–20 kHz range (not specifying signal level), so it seems that my measurements are in the same ballpark.

Yet another problem that can be encountered with DACs is lack of headroom for intersample peaks. Even if there is no resampling involved, DAC still can clip intersample peaks on aggressively mastered tracks. As we can see below, putting non-peak values of waveforms to maximum / minimum integer values can result in having the peaks between samples to reach +2.6 dBFS:

Presence of this problem is checked by using the same files that I used to detect intersample peaks clipping in digital domain. I checked AV7704 and it doesn't have this problem, good!

AV7704 Tone Controls

Tone controls are available for Zone 1 only and offer modification of bass and treble in the range from -6 dB to +6 dB. I was also interested in their operating frequency range and slope. Below are the graphs of the transfer function for the tone controls:

The slopes of the tone controls are gentle, which is good. There is some phase distortion which indicates that the tone controls are implemented as recursive (IIR) filters, however due to gentle nature of the phase changes the resulting group delay is zero.

I'm pretty sure the tone controls are implemented in a DSP as they are very precise (unlike JDS Labs Subjective 3), and it seems strange to me that the control steps are 1 dB. I would like to have a better precision, at least by half of a dB.


All in all, Marantz AV7704 offers good quality analog outputs. Even the secondary zone offers good performance. From my experience, this receiver works reliably and predictably. I haven't encountered any serious glitches during a couple of months I was using it. The built-in HEOS player is useful and offers good quality playback.

Being a "consumer-oriented" (not a pro device), this receiver has some useless extras, like the analog video I/O and "Pure Direct" mode. These are seemingly relics from past models, and Marantz, being a part of a big consortium isn't very good at trimming extra functionality. I would gladly trade these "features" for a digital audio output with digital volume control which I could use for connecting LXminis.

Some annoyances that I have noticed with AV7704:

  • turning connected TVs and monitors on and off interrupts audio playback; I guess, the AVR attempts to recognize the capabilities of the connected unit, however I'm not sure why the interruption happens even when the unit is being disconnected;
  • interruption of audio also happens when changing audio modes and settings;
  • HEOS app on Android can't play album tracks in the album sequence, and this is ridiculous as D+M is aware of this, and the fix is supposedly one line of code; at least, the version of HEOS app built into the receiver doesn't have this problem; UPDATE:HEOS Android app from Jun 6, 2020 (1.562.200) plays album tracks in correct sequence, thanks D+M for the fix!
  • HEOS app is limited to stereo tracks only;
  • there is no indication of the current Zone 2 settings neither on the AVR panel nor as OSD on the Zone 2 TV, and this is very inconvenient; for example, to set up the output level of Zone 2, I had to go to the Settings menu of the unit.

Note that I haven't covered here capabilities of AV7704 in decoding surround audio and downmixing it into 2 channels, I hope to do that later. Also I haven't coverted the built in room correction module (Audissey) partly because I do it externally on miniDSP units, and it only applies to Zone 1 which I use for headphone playback only.