Challenges While Performing Crossfeed Measurements

Since I've started working on the SPL Phonitor Mini crossfeed filter replication, I've learned so many things that I will need several posts in order to describe them all. Let's start with the analog part.

Channel Imbalance

As I've mentioned in the previous post, I have a goal of replicating Phonitor Mini crossfeed settings using DSP in HA-DSP. The first task I was faced with was performing accurate measurements of Phonitor's filters. They are quite delicate—the amplitude of the filters doesn't exceed 3 dB, and the attenuation of direct vs. opposite channels need to be replicated. Not surprising, this task required a lot of attention to the details. Especially when using budget hardware (I use MOTU MicroBook IIc as my capturing tool) which has limits on the precision it can provide.

In addition to being delicate, crossfeed filters involve signal summing due to partially mixing signals from the left and right input channels. This can be a problem when using inexpensive sound card for measurement, thanks to slight channel imbalance. As an example, below are the measurements of all the combinations of "Main" output and "Line" input channels on MicroBook:

It's easy to see that no pair matches exactly in the recorded signal level. What that means to our crossfeed measurements, is that even if we ignore own channel imbalance of Phonitor Mini (which is also present in reality), the records of the sum of left and right channels (from crossfeed) captured by left and right inputs of MOTU will have different levels.

Unfortunately, the input and output attenuation controls on MicroBook, despite being digital, do not provide required resolution in order to compensate these offsets. This also means that one needs to be careful with performing calibration of the soundcard—there need to be separate calibration profiles per input / output channel combination.

I wasn't particularly happy with the discovered lack of balance, so I decided to try other inputs and outputs, as MicroBook has several of them. This time I used unbalanced line output, and "Guitar" input. Since it's a mono input I used a Y-cable 3.5 mm TRS into two 1/4" TS, connecting each one of them in turn. This time the channels balance was much better, just 0.003 dB offset:

It may appear as if the second graph has a much steeper rolloff on ends, but this is only because this time I've magnified more along the vertical scale. When placed next to each other, the graphs do not show such a dramatic difference:

Using the "Line to Guitar" configuration, I connected MOTU's unbalanced line out to Phonitor's unbalanced input, and one channel of Phonitor's headphone output again via a Y-cable to the guitar input. This is how "flat" Phonitor (no crossfeed applied) measures in this setup:

Note that the channel offset is now about 0.015 dB instead of 0.003 dB of the loopback connection, due to slight imbalance introduced by Phonitor.

On headphone amplifiers the channel balance usually skews as volume level changes (unless it's a super expensive super precise amplifier, apparently the Mini isn't one). It's anyway unknown how the crossfeed circuit affects channel balance, so I decided not to try to find a volume knob position providing a smaller offset, having that this volume position gives me the highest signal level without the need to engage the amplifier on the guitar input, which would contribute to non-linearity and noise.

It's worth mentioning that Phonitor didn't change the shape of the curve. Plotted on the same graph, its response covers the loopback response exactly. That's great—this is what you expect from a "transparent" audio equipment.

Noise

Another problem I faced was noise. Since for measurements I have to connect several pieces of electrical equipment together, this creates a possibility for ground loops to appear. There is a great article by Bill Whitlock on the origins of ground loops, available for free download here (need to apply for a free registration first).

For example, when measuring Phonitor (which is a stationary amplifier), I had to disconnect the laptop running measurement software from AC power, since otherwise a ground loop was created via two connections to mains power.

More challenging was avoiding ground loops when measuring HA-DSP. It runs on battery power, but unlike Phonitor contains its own DAC, which means I had to connect it to USB port of the same laptop. This creates a ground loop via two USB connections (from MOTU and HA-DSP). In fact, the noise level introduced by this loop was about 30 dB(Z), I had to get rid of it. I've considered several ways to do that:

1. Instead of USB, use optical input on HA-DSP. But the issue with this approach is that MicroBook doesn't provide an optical output, only coaxial, so I had to add another piece of equipment in order to do conversion. Also, my preliminary experiments using the optical output of Mac Mini have shown that HA-DSP has an additional brickwall filter at about 24 kHz on this signal path (even if the TOSLINK connection is operated at 96 kHz).
2. Instead of USB or optical input, use the analog output on MOTU connected to analog input on HA-DSP. Since MOTU provides a good separation of audio and USB grounds, this connection doesn't introduce a ground loop. But the obvious drawback is that it involves additional A/D conversion on HA-DSP, and it's also limited to 24 kHz frequency top range.
3. Use USB, but insert an isolating transformer after analog output of HA-DSP. The issue with an isolating transformers it that they are non-linear, except for really expensive ones. Since we are doing audio measurements, it would be inconvenient to insert a distorting component.
4. Use USB via USB isolation. Obviously, doesn't introduce any analog distortions, but the issue with the majority of USB isolators is that they are based on Analog Devices ADuM3160 and ADuM4160 chips, which are limited to USB "Full Speed". In theory, Full Speed should provide bandwidth enough to pass 96 kHz / 24-bit stream, but in practice, High Speed USB DAC chips fall back to 48 kHz / 24-bit if the connection can't provide bandwidth enough for 192 kHz / 24-bit. This is true for HA-DSP. There are a couple of High Speed USB isolators, but they cost about 4x times compared to a normal Full Speed isolator.

I decided to go with the option 4, and bought a Full Speed isolator from USConverters. After all, 48 kHz impulse response can be upsampled to 96 kHz (this is the operating rate of HA-DSP) easily.

Below is the plot of measured channel imbalance of HA-DSP output from USB, at 48 kHz, with DSP bypassed:

As it can be seen, the channel offset is about 0.04 dB—that's considerably more than of Phonitor. Another issue is more noise—it can be seen on an unsmoothed plot. And that's even after making 5 consecutive measurements and averaging them.

Yet another issue is different roll-off at high frequencies:

As it can be seen, the difference at 20 kHz is about 0.9 dB! Recall that the same input was used on MOTU, so the difference is definitely due to HA-DSP. Fortunately, this can be compensated as part of filter design process.

UPDATE 9/14/2017: Turns out, the early roll-off for HA-DSP is an artefact of averaging of measurements. In fact it's not that bad, I'll publish updated measurement graphs in the next post.

Conclusion

I think that's enough for now, and it's just the beginning of my findings! Will publish more soon.

MiniDSP HA-DSP First Impressions

Being an audio geek I couldn't ignore this promising gadget from MiniDSP. To me, MiniDSP products were always associated with loudspeaker and room digital correction, but unexpectedly they have tried their design skills in a relatively niche area of portable DACs and headphone amplifiers. And thanks to their vast DSP experience, the resulting product has come out very interesting and highly unique.

There is a very good description of HA-DSP from the manufacturer. I will give my own version of it, coming from the possible range of functions that this device can perform:

1. HA-DSP can be used as a regular portable headphone amplifier taking analog input. Unfortunately, HA-DSP can't also be used as a power bank despite presence of a deceptively looking USB-A port on it.
2. Since HA-DSP has USB inputs and is USB Audio Class compliant, it can also be used as a portable DAC for mobile devices and computers. However, I've encountered issues when I tried using it with some Android devices leading to mobile device reboots.
3. The 3.5 mm analog input port of HA-DSP also accepts mini-TOSLink jack, allowing to connect HA-DSP to audio gear that has optical output. Note that both analog and optical input has a cutoff frequency at 22050 Hz, even if the output device provides a full range signal.

However, you can find numerous other portable DACs that can fulfill all these scenarios and don't have the drawbacks I've mentioned. What makes HA-DSP unique is the "DSP" part in it's name. If you know how to create FIR and IIR filters, this little box offers a lot of room for audio experiments, for example:

Make precise output frequency response adjustments using 10 bi-quad IIR filters per channel.

Create arbitrary crossfeed effects with two parallel blocks of FIR filters and cross-commutation.

Apply headphone target curve correction with FIR filters.

Switch between 4 configurations of the filters that can be tied to different headphone models and crossfeed settings.

My own plan for this box is to replicate Phonitor Mini crossfeed filter, and also to experiment with headphone correction based on the headphone measurements from Inner Fidelity database and Isone MorphIt filter.

I've already mentioned some drawbacks of HA-DSP. Another issue I discovered while making measurements is some non-linearity of the frequency response. My measurements of HA-DSP using MOTU Microbook IIc have shown early roll-off in high frequency range. Below is a frequency response of HA-DSP (blue) compared to Microbook's own (orange):

On this graph, the HA-DSP's response plot uses Microbook calibration from a loopback measurement. The good part, however, is that this roll-off can be corrected using the on-board DSP as part of FIR filters design.

Conclusions

HA-DSP is definitely a device targeted to audio geeks. One would probably need to carefully consider whether they would be able to use it to full extent. If designing filters isn't your hobby, there are definitely better alternatives in terms of cost and quality.

Wiistar 5.1 Audio Decoder Teardown

For some experiments that require hardware decoding of Dolby Digital I've acquired a cheap Chinese 5.1 decoder on Amazon—it costs just $24 so there was not much hesitation while buying it.

The good news is that it's indeed a proper Dolby Digital (AC3) decoder, which also supports upmixing of stereo channels into 5.1 (probably using Dolby Prologic). The bad news is that the quality of the audio output is... consistent with the price of the device.

I've found a post by Alexander Thomas describing previous versions of this device. Compared to what Alexander had observed, the hardware I've bought seems to be somewhat newer:

1. Instead of CS4985 decoder chip it uses an unidentified DSP chip of a square form.
2. There is no filtering of the output signals or any "bass management" (sinking of low frequencies from the main channels into the subwoofer's channel).
3. The unit is powered from a 5 V source instead of 9 V.
4. The unit provides a 5 V USB power outlet.

There are still some similarities though:

1. LFE channel lacks +10 dB boost expected by the DD spec.
2. The board's ground is not connected to the case.

Hardware Teardown

Now let's take our screwdriver and see what's inside the box. This is how the board looks like:

Most of the components are mounted on the top side. Some of the major components can be identified:

• [1] 4558D is a stereo opamp, this make is by Japan Radio Company (JRC);
• [4] ES7144LV is a stereo DAC—the board employs three DAC / opamp pairs;
• [7] 25L6405D chip is flash memory;
• [6] NXP 74HC04D is hex inverter chip;
• [2] AMS1117 is power regulator.

There are two mystery chips:

• [5] the big one labeled VA669—I suppose that's the decoder DSP, having that there are traces coming from it to the DACs, but the actual make and model of the chip are unknown;
• [3] the one labeled "78345 / 8S003F3P6 / PHL 636 Y"—judging by its position on the board, it could be a microcontroller handling input selection and "5.1 / 2.1" switches.

And this is the bottom view:

One interesting thing to note is that the labels and holes suggest that this board can be equipped with RCA output jacks per channel, as an alternative to three 3.5" stereo jacks and the 5 V USB outlet. This suggestion is confirmed in the manual:

Measurements

I was wondering whether this device can be used in any serious setup, and for that I've hooked this device up to the inputs of MOTU UltraLite AVB audio interface.

I needed a test sound file that is AC3-encoded and contains measurement sweeps in all 6 channels. For that purpose, I took the measurement sweep file generated by FuzzMeasure, and used Audacity in order to create a 6-channel file with a sweep in each channel:

Note that ffmpeg library which is used to encode AC3 applies a lowpass filter to the LFE (4th) channel. This will prevent us from seeing the full performance of the LFE channel on the device.

Using a TOSLINK cable I hooked up the device to MacMini's optical output, played back the encoded file, and recorded the decoded analog output using MOTU.

The first thing I discovered was that the surround channels are swapped. That is, they use a reverse of the standard TRS stereo channels mapping where the left channel is on the "tip" contact plate, and the right channel is on the "ring". Instead, the left surround is on the "ring", and the right surround is on the "tip". Perhaps, this was done on purpose to undo the reversal of "left" and "right" if one sets the surround speakers facing him, and then turns around :)

The next discovery was quite a bad shape of the output waves. As one can see, the sine wave is severely clipped at bottom half-waves. This is how the source -3 dBFS sine wave has been rendered:

Input sine wave with smaller amplitude (-6 dBFS) is clipped a bit less:

This is very unfortunate, and is probably caused by a bad design of the output stage. Looks like using the 4558 opamp wasn't the best choice in the first place, and the designers of this board seriously hindered its performance by failing to drive it correctly.

After looking at these horrible output sinewaves, I wasn't expecting a good frequency response, and indeed it's quite bad. Below are the plots for the left channel from a -3 dBFS input signal (blue), and for -6 dBFS input (orange), no smoothing:

The measurements for the remaining channels are the same as for the left—at least this device is consistent for all channels. Below is left channel (blue) vs. LFE channel (yellow):

This plot confirms that the LFE channel has the same output level as other channels, lacking the required +10 dB boost.

It's very funny to look into the "Technical Data" section of the manual for this device, stating:

Frequency Response: (20 Hz ~ 20 KHz) +/- 0.5db

The authors tactfully omit the level of the input signals used in this measurement (if it actually was performed)—probably the level wasn't too high.

Conclusion

Looks like this family of devices can't be used in any serious setup. It will be interesting though to try to reverse engineer the electrical design of this board, and fix obvious flaws.