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.

Inputs

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.

Outputs

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.

Configuration

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.

Verification

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.

Conclusions

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;
  • 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.

Monday, June 8, 2020

Switched to Markdown

After writing about 50 posts I decided to do something about how I typeset them. Previously I was using Blogger post editor in "Compose" (WYSIWYG) mode. It allows to get job done, however there was no complete control over the details of formatting. For example, I like to use non-breaking spaces between values and their units, as in "1 kHz", so they don't end up on different lines. However, Blogger editor doesn't show "special" characters. They can only be viewed in HTML mode, however the text looks overwhelming with all the extra tags and attributes that Blogger's WYSIWYG editor throws in.

Another huge missing feature of the Blogger editor is "find and replace". There is "find" function built into the browser but no "replace". Again, you can work around by copying the HTML source into a capable editor, doing all the work there, then pasting back. Hopefully you haven't screwed up the HTML tags.

I realized that I would like to use my favorite editor for writing posts and then convert them into HTML (just once!), paste the result into Blogger and be happy. These days Markdown is the standard way for typesetting moderately complex pages, and its minimalist nature makes the page source look very readable even without syntax highlighting.

So Markdown it be. Where is it convenient to store Markdown sources? GitHub pages is a good place since GitHub offers a built-in renderer for them. The renderer also adds some nice "extensions" to basic Markdown. Decided—I will use GitHub pages for storing the Markdown originals and continue posting them on Blogger, because people actually do read the posts there.

Converting old pages

As an experiment in feasibility of this approach I decided to convert my existing blog pages to Markdown and "distill" them back into HTML. This would help to establish the process and iron out all the possible issues. This also ensures that the blog "mirror" on GitHub doesn't have dangling links to old posts.

I downloaded the archive of this blog via Blogger's "Back up content" function. It provides a huge XML file containing all the posts in HTML format, so it's easy to cut out their content for further processing.

For conversion I used Pandoc tool which among numerous formats supports both HTML and GitHub "flavor" of Markdown. So, for the old pages the process was as follows:

  1. Save the post as HTML file, convert it into GitHub markdown using Pandoc:

    pandoc input.html -f html-native_divs-native_spans \
    --shift-heading-level-by=-1 --atx-headers -t markdown_github \
    -o output.md

    By trial and error I figured out that I like the results of the deprecated markdown_github converter better than its gfm replacement. For some pages I used --shift-heading-level-by because I was using <h3> HTML headers and needed to have them "level up"-ed.

  2. Clean up the converted Markdown: remove trailing whitespace, extra line breaks, make sure all non-breaking spaces are in place, etc.

  3. Preview the Markdown file using excellent grip tool. This saves from unnecessary uploads to GitHub.

  4. Convert the Markdown back to HTML for Blogger:

    pandoc output.md -f markdown_github -t html -o distilled.html
  5. Paste the "distilled" HTML back to Blogger.

  6. Upload the Markdown to GitHub.

  7. Compare the looks and make necessary adjustments to Blogger styling.

The last step also helped me to resolve long standing annoyances with the default CSS styles used by "Awesome Inc" Blogger theme. I put my CSS overrides into "Advanced > Add CSS" section in the theme editor.

BTW, I'm not exaggerating about the converted back HTML being "distilled". Blogger puts so much superfluous formatting that the size of a file containing a post from Blogger typically reduces by 25–50% after converting back and forth via Markdown!

Of course, the conversion isn't without flaws, and Markdown does in fact offer less formatting capabilities than Blogger. Let's consider the differences in detail.

Post links

I decided to use the same file structure for Markdown posts, this makes converting links easier. The conversion is needed because GitHub uses names of the Markdown files—md extension, while Blogger uses html. I made all the post links to be "site relative" (starting from /) so it doesn't matter where the page is actually hosted.

This way, a link to a previous post in Markdown looks like this:

[as shown in the previous post](/2019/06/previous-post.md)

and when "distilling" Markdown source to HTML I replace md with html.

Pictures

There are a lot of pictures in this blog, I decided to leave them hosted on Blogger. The reason is that Blogger server can resize the picture to a smaller size from the parameters specified in the image URL. These smaller images are used for previews in the article. After clicking on the preview a full size picture is served. This is more efficient than serving a full picture only and sizing it down in the browser.

This approach also works when links to images host on Blogger are used in Markdown arcticle on GitHub. As I've figured out, GitHub in addition makes a copy of any externally hosted image for serving from its own CDN, so it really doesn't make sense to pull out images to GitHub manually.

One notable loss is that Markdown doesn't allow specifying alignment and interaction with text for pictures, so they are always aligned to the left and can't have text fills on the size.

Code

Up to the redesign Blogger wasn't offering dedicated code formatting. I used monospace font with non-breaking spaces for sequences of multiple spaces. While converting, I changed all those code fragments to use Markdown fenced code blocks.

Tables

Similar thing for tables. I used tabulated monospaced formatting. This wasn't super convenient. I converted these ersatz tables into Markdown tables which translate into actual HTML tables for Blogger. This looks better. The only inconvenience is that GitHub Markdown doesn't allow "headerless" tables.

Colors

Markdown doesn't have means for colorizing text. It's actually good for accessibility (think screen readers, color blind people), but I used to highlight text with colors when discussing graph. Now I will have to provide more annotations on the graph itself.

Miscellaneous

  1. In Markdown the header of the post is specified on the first line using # style (heading level 1). In Blogger the header stored separately.
  2. Special characters like "non-breaking space", "em dash" need to be written using corresponding Unicode characters in Markdown. Note that the sequence of three dashes --- is used in Markdown for horizontal breaks.

Writing a new post

I'm writing this post in Markdown and the life feels good. The only culprit is adding pictures. I still want them to be stored on Blogger. For example, I want to post an image of the same post in Blogger and on GitHub. Here is what I have to do. After preparing the image, I upload it to Blogger and insert into the post draft. Then I copy the link and transform it into Markdown link format. This is the result:

The GitHub mirror of this blog is now located here: https://mnaganov.github.io

Testimonials

Both Pandoc and grip are awesome tools that helped me a lot with converting my posts into Markdown and back into HTML. I highly recommend them for any document conversion work and Markdown authoring.

Saturday, May 2, 2020

Sennheiser Ambeo Headset Applications

As I have mentioned in the previous posts about optimizing audio in our Mercedes GLK, I used Sennheiser Ambeo Headset as a measurement device in the car. In this challenging acoustical environment it allowed to achieve better channel matching than a conventional measurement microphone. I decided to make a dedicated post about this headset because I've found some interesting applications for it.

Overview

I discovered this headset at the AES Headphones conference where it was used in conjunction with Magic Leap's One AR glasses. By the time when I decided to buy it for my experiments, Sennheiser had already abandoned its production. Nevertheless, it's still possible to buy leftovers from the stock and used gear.

This is how this device looks:

By comparing it with the image on the packing box it's easy to spot a marketing trick. On the box the controlling unit is pictured from the side, making an impression that it's thin and long. However, in reality this unit is pretty thick and looks a bit ugly:

The headphones themselves are designed to be worn around ears, sports-style. They don't however feel sturdy enough like a real sport-style headphone should—yet another perceptual mismatch. Overall, the look of these headphones isn't too exciting, certainly not as appealing as "iconic" Apple earbuds.

Speaking of the technical side, the only connection option offered is Apple Lightning connector. There is also a companion iOS app, however so far I was only using this headset with Android devices and laptops. This becomes possible using Anker's Lightning-to-USB-C adapter which is a must have device if you happen to own any good Lightning headsets and plan to connect them to other devices besides your iPhone. Anker's connector tech specs explicitly lists the Ambeo headset as a compatible device. As a side note, the adaptor also works great with Lightning cables by Audeze.

The controlling unit has a lot of buttons. Besides three usual media controls, there is also a rocking switch toggling between active noise cancelling, "normal" mode, and "transparent hearing"—when the device uses its built-in microphones to allow any external sounds in. This mode is useful because the headphones are designed for in-ear insertion and actually provide a good noise isolation even without active noise cancelling.

Another switch on the controlling unit activates "padding" for the stereo microphones. The designers intended it for use at concerts to avoid clipping during recording.

Speaking of the microphones, since this device was conceived for "3D" recording, besides the usual headset style mono microphone on the right earphone wire, it also has a microphone housed inside left and right earphone:

Before I bought this device I was thinking that the microphones are behind the grilles on the sides of the earphones, but actually the microphone is placed on the inner side of the earphone and faces the reflecting cavity of the pinna:

Overall, from a regular consumer's point of view, the appealing features of this headset are its noise cancelling function and the ability to create entertaining 3D "dummy head"-style recordings. However, the build of the earphones and bulkiness of the controlling unit (and probably relatively high price) most likely worked against its wide adoption.

Earphones

I didn't plan to actively use this headphone for listening to music, but it's still interesting to check what it is capable of. For comparison I'm using very well known and widespread Shure SE215 in-ear phones.

What you will immediately notice with the Ambeo headset is that it's very bright, up to the point when listening to vocal recordings with a bit of extra sibilance becomes unpleasant. My usual tracks for checking this are "Little Wing" performed by Valerie Joyce on "New York Blue" album, and Madonna's "Hang Up" from "Confessions on a Dance Floor".

On the other hand, this brightness also provides a very strong sense of spatiality that can be heard on Hol Baumann's "Endless Park" theme from "Human" album which sounds much duller and more two-dimensional on SE215.

I don't have a rig for measuring headphones, however I was able to capture reliably the high-frequency part of the transfer function of both Ambeo and SE215 by moving them in a free air close to a measurement microphone (a variant of MMA averaging)—the measurements are only valid starting from about 2 kHz. Then I simply divided these transfer functions and found this huge bump around 9 kHz on Ambeo:

To validate my finding, I used an equalizer first to add more high-end to SE215 and then to reduce the harshness of Ambeo, and it worked. The setting of the high-end equalization on Ambeo is extreme. The right setting seems to be somewhere in the middle between Ambeo and SE215—to add a wide peak of +6 dB Q 0.7 centered at 9 kHz to playback via SE215, and to apply a good dip when playing via Ambeo.

The difference in 1–5 kHz region can also be seen and it results in a more "distanced" perception of vocals. I tried adding a -2.5 dB Q 0.7 filter centered at 2 kHz and this helped adding some "depth" to the sounding of SE215 trading for some loss of clarity. Looks like these two settings result in a more "ambient" perception of an audio program. I suppose the reason for this equalization on the Ambeo headset is due to intention to use it primarily for immersive audio playback—playing back the "3D" sound captured with its microphones.

As a side note, I also liked that I found this equalization curve for Shure SE215, which by default sounds more "closer" and two-dimensional. It works even better if crossfeed is added. This experiment has rekindled my interest in SE215.

One problem that I've found at least with my particular Ambeo headset is the mismatch of the earphones transfer function at high frequencies. First I thought that this was due to my bad measurements—I used a DIY coupler to simulate an ear canal, so positioning of the earphone wasn't super precise. But then I also tried the averaging measurement method mentioned above. With both methods, I was always able to match left and right speakers on other in-ear headphones, except for Ambeo which always yielding rather different curves for the left and right earphones (below is the MMA measurement):

So I came to a conclusion that it must be the headset's fault. However, I can't say that I can hear this mismatch clearly, (especially the one in high frequencies). Still, for a headset of this price which has built-in DSP processing leaving this fairly obvious (via measurements) mismatch between left and right channels seems strange to me.

Microphones

Since my primary intended use of this headset was for "dummy head"-style measurements, I was curious to see how well the left and right microphones are matched and how they are tuned. Note that when this headset is connected to a PC (or Mac), it offers both "mono" and "stereo" recording modes. My expectation was that the "mono" mode uses the headset microphone (located on the right earphone wire) which is intended for communications. However, it turned out that the "mono" mode simply uses the left earphone microphone only. So I'm not sure how to activate the headset mic—perhaps when this headset is connected to an iOS device directly, it uses some special mode not available via the Anker adapter. Not a big loss though.

After seeing the mismatch between the outputs of the left and right earphones I was worried whether left and right "3D" mics are suffering from the same issue. I validated them by placing as close as possible to each other in a fixture (not on my head) and measuring the same sound source. Turned out that the mics are actually matched quite well, and we can see very close measurements when coherence is good. On the picture below the measurements are blanked out when coherence is less than 85%:

The tuning of the mics seems to be for "diffuse field"—with a prominent bump at high frequencies. This is important to know as would I try to tune a sound system to a "flat" curve using these mics, this will result in an excessively bright sound. Here is the comparison of measuring the same sound source in the same conditions using a Beyerdynamic MM-1 microphone with "free field" (0 degrees) calibration:

We can see that microphones of Ambeo start sloping up after 2 kHz at approximately 2 dB/octave rate. I wouldn't be paying much attention to other differences as they are likely due to differences in the microphones placement.

The next validation was to see how the transfer function of the microphones differs when they are inserted into ears. Due to the microphone placement, the incoming sound is now transformed by reflections from the pinna and torso. Below is the graph comparing freestanding vs. in ear microphone placements for the same sound source:

As we can see, the main difference is the prominent dip at approximately 4.8 kHz. I'm not a big specialist on anatomy of human hearing, so I can't say what it is caused by exactly. I tried putting a sound absorbing material on my shoulder and this changed nothing, so I suppose this dip is caused by some interference within the pinna. The wavelength corresponding to 4.8 kHz is approx. 7 cm, so half and quarter wavelength fit ear size.

There is a 2–3 dB boost in the speech range (300 Hz to 4 kHz)—I suppose this is thanks to the design of the pinna. And also noticeable a significant loss in high frequencies starting from approximately 14 kHz. This can actually explain why I'm not hearing well the mismatch between the left and the right earphones.

The differences in low frequencies are most likely due to variations of placement of the freestanding vs. in-ear and need to be ignored.

While writing this post I've looked up other reviews of Ambeo headset and found that on iOS it's possible to record at 24/96. Unfortunately the Anker adapter only supports 24/48. However, that's enough for my applications.

Applications

Now let's consider a couple of applications for this headset.

Sound System "offline" Evaluation

It can be useful to capture the produced sound field of a sound system for evaluating it later, perhaps in a more comfortable setting. This is similar to the original function of this headset—capturing 3D sound fields for realistic playback recreating the original environment.

There are great notes by S. Linkwitz of how much our perceptual system can ignore the room and focus on the direct sound of the speakers. However, if we reproduce a binaural recording of the system in a room back using the same system, we immediately start noticing all the room contributions (see the paper "Room Reflections Misunderstood?", Section 5). This is a really interesting experiment to try with this headset.

Note that since Ambeo is a binaural headset, not a spherical stereo microphone, the pinnaes of the person making the recording inevitably color the sound. As we have seen in the section above, the filtering by the pinna is non-negligible. I found that it's best to play back these recordings either on Ambeo itself (no surprise here), or on IEMs with close to direct field equalization. Playing on over-ear headphones or via speakers will "apply" the pinnae filter once again.

"Dummy Head" or "Spherical Microphone" Measurements

This is what I was doing when tuning audio in the car. Since the "room" is very small, and the presence of a human body introduces a significant change in the acoustic environment, using in-ear microphones for left and right speakers alignment produced better results than use of a measurement microphone.

To reiterate, I was using Ambeo only for matching the sound arriving into the left ear from the left speaker to the sound arriving into the right ear from the right speaker by equalizing the speakers. The final tonal adjustment was done using MMA averaging and double-checking with known music tracks. As we saw from the measurement of Ambeo's stereo microphones are well matched and are equalized for diffuse field. The dip around 4.8 kHz that occurs when they are inserted into ears (at least, my ears) must be ignored during sound sources matching.

A note of caution here. Ambeo headset is a digital device, not an analog microphone, and unlike pro audio interfaces it lacks external clock input. Since the audio output from Ambeo only goes into its earphones, one will need to use another digital audio interface for audio output. This is where the problem comes in—with two digital devices not synchronized via "world clock" feed there inevitably will be clock drift between them. To illustrate how bad the resulting measurements can be affected check the graph below:

The red trace is the original EQ filter (BTW, it's the SE215 "improvement" filter I was discussing in the Earphones section, with a bit of bass added), the magenta trace is the same filter as measured by Ambeo headset from a playback done via a separate audio interface. As we can see, there is a very serious spectral shift.

What to do about it? In REW the solution is to use the shortest test impulse (128k):

Using a shorter impulse has worse signal-to-noise ratio (there is more visible noise on the green trace) but at least there is almost no spectral shift. In fact, it's a well known problem with REW when it's used with USB microphones like miniDSP UMIK-1. I've seen several threads on forums where people were wondering why the results of their measurements using different log sweep lengths but otherwise the same setup didn't match. I'm really curious why REW allows for multi-device measurements by default.

The clock drift problem is the reason why Acourate only allows using a single device for input and output. With Acourate it's recommended to use even longer sweeps than REW uses, so attempts to "work around" single device limitation by using drivers like ASIO4All will inevitably lead to a severe spectral shift.

SMAART has a very useful feature for tracking the impulse response delay changes automatically. This is in fact the technique I used when aligning car speakers via Ambeo. I had to experiment with averaging settings to find the one that allowed for more reactive compensation of clock drift. Usually, the shorter the averaging is, the better.

Over-ear Headphones Equalization

Here is the full story of how I obtained the graphs above. I put on the Ambeo headset and then put Audeze EL-8 closed back over-ear headphones on top of it. Then I was playing test sweeps via EL-8 and measuring their output using Ambeo. Crazy, right? I don't think anyone at Sennheiser were considering this application of the Ambeo headset. However, as the last graph demonstrates, this setup can actually be used for measuring acoustically the effects of headphone equalization.

Does it mean this $200–$300 headset plus your own head can replace a head simulator for over-ear headphone measurements? Not quite. The trick with the measurement above was that I didn't move or replace the headphones while doing it, I simply was toggling equalization on and off. This allowed for quite reliable comparison of the measurements before and after equalization. What happens if I remove the headphones from the head, put them back again, and make another measurement? The measurement will be different. As people in the headphone industry know, in order to obtain a reliable measurement of headphones one needs to re-mount them several times and then average all the measurements taken.

This is how the derived EQ looks like when I actually re-installed EL-8 several times and used averaged measurements:

The results at high frequencies are not that reliable anymore, the tolerance is only within 2 dB, which is a lot for headphone measurements. This is what will happen if one will try to compare equalization of different over-ear headphones. So, Ambeo isn't a very precise tool for this task, at least for the whole operating frequency range.

However, Ambeo still provides a good reliable output for low frequencies. And in fact, it's the low frequencies where use of a head simulator is required because headphone drivers can only deliver their full bass output when there is a closed chamber between the driver and the ear drum. That brings an idea—we can use a combined measurement of MMA for high frequencies plus Ambeo for low frequencies.

As I've mentioned in the Earphones section, this is a variation of MMA where we slowly move the headphones near a microphone and wait for the RTA measurement with infinite averaging to stabilize. To demonstrate that the produced measurement can be reliable used, here is the EQ derived from this technique, I was holding EL-8 close to MM-1 and moving it slowly, waiting for 100 sampled measurements to accumulate:

As we can see, there is much better tolerance at high frequencies, but below 1 kHz the data is unreliable. And here is where Ambeo comes to the rescue. By merging together low frequency measurement done by Ambeo on a head with the rest of measurement obtained via MMA we can measure over-ear headphones output reliably.

A note of caution—this method is only good for comparing headphones. We measure one headset, then another, then derive the differences in their equalization. There is no way for measuring absolute frequency response of headphones using this method.

For a practical demonstration I measured filtering applied by Audeze Cipher cable for EL-8. Everyone heard the debates whether or not headphone cables make the sound different. Well, in the case of the Cipher cable vs. analog cable the difference is real because Cipher cable is digital and contains a DSP in it. I noticed that even when the EQ in Audeze app is set to 0 dB at all bands, the sound via Cipher still differs from the sound via analog cable. And I was able to measure this using the technique described above:

For the analog cable measurements I used SPL Phonitor Mini, which has very low output impedance and thus provides adequate bass output. The section of the graph below 1 kHz was obtained from comparing measurements done by the Ambeo headset on my head. There is noise because I turned off FTW gating in SMAART to get a full bass extension. However, we can clearly see that the Cipher cable boosts the bass by almost 3 dB (remember, this is with a "flat" EQ setting in the controlling app!). The section of the graph after 1 kHz was obtained with MMA technique. A "scoop" at middle frequencies can be seen clearly.

I found electrical measurements of the Cipher cable done by user KaiSc on Head-Fi.org. It confirms the 3 dB boost, but not the middle section scoop. Although KaiSc also mentions compression effects from the Cipher DSP at high volume. Since I was testing EL-8 at high volume to obtain adequate free-field output, it's possible that the DSP has thrown in some compression at this point.