Using a TC Electronic DBMax for live streaming

This post describes using a TC Electronic DBMax to enhance the audio of a live stream.

In a live streaming situation, control of audio levels is critical to providing listeners and viewers with a great auditory experience. Inconsistent levels, or levels that are too loud or too soft result in a suboptimal experience, which degrades the performance.

The TC Electronic DBMax (Digital Broadcast Maximizer) is an ideal device for stabilizing and enhancing audio for live streaming and live recording situations. It supports up to three pre-dynamic inserts (AGC, Parametric EQ, 90 deg. mono, Dynamic Equalizer, Stereo Enhance, Normalizer, MS Decoding and MS Encoding), an expander, compressor and limiter, and a single post-dynamic insert (Transmission Limiter or Production Limiter). The EQ, compressors, and limiters are all 5-band, and provide broadcast quality results. It is basically a Finalizer 96K on steroids.

This image has an empty alt attribute; its file name is DBMAX-INCL.-DIGITAL-BYPASS-BROADCAST-MAXIMIZER-II_P0CXM_Front_L.png

In my role as recording lead at ICF, I use two separate DBMax devices to solve two separate problems. 1) stabilize the FoH signal for video, and 2) boost the stabilized signal for streaming. Once I put these two devices in, the video team I work with thanked me multiple times as they have a stable audio level they can rely on for single every event, no matter what kind of event, and they love the audio quality.

The first DBMax balances audio levels between a live worship session, where audio levels during worship are 10–15dB louder than the normal speaking volumes, and speaking sessions. The goal is to provide a signal that roughly meets the -23 LUFS broadcast standard in Europe, which means the audio recorded for video needs no additional post-processing for release to public TV (which we do for special events).

  • The feed coming from the FoH board is limited (with the Waves L2 Ultramaximizer), which provides a known maximum audio level.
  • The first insert is the Normalizer, which gives a small 3dB boost to the the overall audio level. The focus is to bring speaking parts into a better working range for later processing.
  • The second insert is the AGC (Automatic Gain Controller), which automatically raises the volume by up to 3dB (for soft speaking), or reduces it by up to -20dB (for loud band numbers).
  • The third insert is Stereo Enhance. The FoH provides signal for the main audience, but the stereo image in a live situation is typically reduced so that audience members on the far left/right sides don’t hear only a mono left or right signal. This insert provides a wider and more natural signal for TV viewing.
  • The 5-band expander is not used.
  • The 5-band compressor does a decent amount of compression, typically several DB. The advantage of a 5-band compressor is that kick and tom hits will not result in the overall signal being compressed, giving a more transparent sound.
  • The 5-band limiter catches loud peaks that might have made it past the compressor and AGC.
  • The last insert is the Production Limiter, which limits the overall sound signal. It is rarely triggered, and is mostly there to catch anything left.

The second DBMax takes the audio from the first DBMax, and boosts it to roughly -14 LUFS for live streaming on internet, and for streaming to TVs throughout the building (e.g. for parents in the children’s area). The -14 LUFS level was chosen as both YouTube and Apple both use this standard, and signals recorded from this DBMax require no additional post-processing for release to those streaming platforms (for video or podcasts).

  • The first insert is the Parametric EQ, which boots the signal 9dB, and applies a shallow high-pass and low-pass filter to strip off extra energy that laptop speakers and most headphones cannot reproduce.
  • The second insert the Spectral Stereo Image, which widens signals above 50 Hz, making for a better experience on headphones or laptops.
  • The third insert is the AGC, which makes minor signal adjustments of ±3dB to produce more consistent results.
  • The 5-band expander is not used.
  • The 5-band compressor provides fast and light 2.0:1 compression, and a 3.9 dB boost.
  • The 5-band limiter catches loud peaks that might have made it past the compressor.
  • The last insert is the Production Limiter, which limits the overall sound signal. It is rarely triggered, and is mostly there to catch anything left.

For those who are curious, the DBMax introduces roughly a 5ms delay into the audio signal chain, which must be compensated for when aligning the audio and video signals together. The video signals I’m working with have an 80ms latency delay, so I apply an additional 75ms delay using a Behringer X32 console, which was a console already available to me.

If you are interested in buying a used DBMax, they can frequently be found on eBay in the $700–1500 price range. New they are $4000+, so the used price is quite good. With patience, I’ve purchased three in the $700-$800 range. Try to purchase a device with the v2.60 or v2.90 firmware as the v2.02 firmware has a several minor issues that were fixed in the v2.60 release.

If you’d like to know more, let me know in the comments. For the actual settings, see this Google spreadsheet.

Additional Resources

Willow Creek FAQ: Broadcast Audio Process. This article describes how Willow Creek uses the DBMax.

Upgrading TC Electronic DBMax firmware

This post provides steps for updating TC Electronic DBMax firmware.

The latest firmware (v2.90) along with installation instructions are available from the tc electronic Music Tribe site. The instructions only describe using a M5000, which I don’t have, so I needed another solution.

The DBMax has a built-in PCMCIA card reader for storing settings, and which can be used to upgrade the firmware. According to the manual, the DBMax supports Type 1 PCMCIA cards with a minimum of 64KB SRAM and a maximum of 2 MB. Although settings can also be stored and recalled via MIDI, the firmware can only be updated via PCMCIA.

TC Electronic devices interact with the PCMCIA card as a raw storage device, and do not utilize a filesystem such as FAT. This means that the firmware must be copied as raw data to the device using software capable of doing so.

The solution I’ve found for copying the firmware to a PCMCIA card with a CSM OmniDrive USB2 Professional PCMCIA card reader. Unfortunately, PCMCIA cards are no longer common as they once were, and finding a drive proved to be both difficult, and expensive ($350 on eBay), but it works. Alternatively, an old laptop with a PCMCIA card reader should work, or with a TC Electronic M5000 (not the M5000X) and a 3.5″ floppy drive.

OmniDrive USB LF SD
OmniDrive USB LF SD

The OmniDrive site provides downloads for Windows which include a Software Driver (v3.3.4) that enables reading/writing of a PCMCIA card as a normal drive, and PC Card Manager (PCM) (v3.1.1) which enables reading/writing of a PCMCIA card as a raw storage device. The PC Card Manager is required for updating firmware.

To copy the firmware to the PCMCIA card…

  1. Unzip the dbmv290.zip file, which should provide a dbmv290.wiz file.
  2. Open the PC Card Manager software.
  3. Click the “Copy file(s) to a PC Card” icon.
  4. Select “New” to start a new Copy Job.
  5. Choose the dbmv290.wiz file as the Source File, and click “OK”. If the file isn’t listed, make sure file dialog is showing “All files”, not just “Images (*.PCC;*.PCA)” files.
  6. Click “Copy”. This should take <1 second.

To copy the firmware to the DBMax…

  1. Insert the PCMCIA card into the DBMax.
  2. Power the DBMax on while holding the “Help” button.
  3. Press the “OK” button to initiate the firmware upgrade. This should take <10 seconds.
  4. Power cycle the device.

That’s it!

Avid S3L and 3rd-party AVB Devices

This article describes how to connect and use MOTU AVB audio interfaces with the Avid S3L-X console.

For years, I have wanted to connect my MOTU AVB audio interfaces to my Avid S3L console, but have had no luck. After recently finding some information on the internets, I’ve found a way to reconfigure the S3L to talk using 8-channel AVB streams, which my MOTU devices require, and with some effort I now have bi-directional audio working!!

I’ve written up a document, and shared it as a public Google doc. I’ll eventually write it up here, but don’t feel like messing with WordPress right now.

Avid S3L-X and 3rd Party AVB (a Google Doc)

I look forward to any feedback!!

Testing Phantom Power

I’m the proud owner of an Earthworks M30 30kHz measurement microphone, which I use it to measure and calibrate sound systems using Smaart v8 from Rational Acoustics, along with a MOTU UltraLite mk3 audio interface.

Background

A few of years ago, I was attempting to measure a system, but had the strange behavior that after 15s or so, the signal from the microphone faded away completely, making it impossible to calibrate the system. I eventually realized that by swapping the 10m XLR cable I was using for a 3m cable, the problem went away.

As I’d used the setup without issue in the past, I assumed the issue was the with the particular XLR cable itself, but eventually realized that the longer cables produced hit-and-miss results, and that I needed to dig deeper into the problem. I also realized that I had always connected the mic through the console of the system I was measuring (to ensure I was calibrating the full path), but on this occasion I was calibrating directly with the interface as customers were expected to provide their own console, and only the system was a fixed installation.

As my father has decades of experience in audio, he was my first source for troubleshooting. He suggested that the mic had a high impedance, and that the 10m cable would cause issues with such a high impedance. Contacting MOTU, they also suggested that the mic might be the issue. The M30 model I own is a 600Ω impedance model which I’ve owned for 7y+.

With this information, I contacted Earthworks directly. After some discussion of the issue via email, Earthworks suggested that the impedance of the older M30 models might only be part of the issue, and that my audio interface might not be capable of maintaining 48V phantom with the longer cables. They provided me with a test procedure which I could use to verify my setup experimentally.

Testing

Phantom power test procedure

To perform the tests, I needed to build a cable. As I didn’t have a 47Ω resistor, I used a 50Ω resistor instead as it was close enough.

Rather than only check my UltraLite mk3, I set about testing the phantom power every XLR mic inputs that I had available to me at home at the time. I studied Physics in university, and doing experiments like this interest me!

The results below are based on the various document sections of the test procedure.
1.A. Measure voltage between pins 1 (neg) and 2 (pos). Expect 48V DC (±1V).
1.B. Measure voltage between pins 1 (neg) and 3 (pos). Expect 48V DC (±1V).
2.B. Measure current between resistor and pin 2. (47Ω resistor across pins 1 & 3; 47Ω resistor from pin 1). Expect >= 6.2 mA.
2.C. Measure voltage from above. Expect close to 48V DC.

1.A (V)1.B (V)2.B (mA)2.C (V)
Avid S3L-X E3 Engine48.2548.28747.59
M-Audio ProFire 610 (FireWire)49.4849.49748.17
Mackie 802VLZ447.3347.34746.8
MotU 828mk348.0148.01541.8
MotU UltraLite mk3 (FireWire)48.348.31541.5
MotU UltraLite mk3 (external)48.348.31542.16
MotU UltraLite AVB4949.05631.8

Results

The clear result was that none of my MotU devices were not capable of providing phantom power necessary to drive my Earthworks M30 mic.

After presenting the results to Earthworks, they informed me that they offer replacement circuity for the M30 to convert it from 600Ω to 150Ω. I chose to keep the microphone in its original state, and I instead purchased the ART Phantom II Pro that was recommended in the testing document. With the inclusion of that device, I have had no further problems with my setup, and I continue to use the MOTU UltraLite mk3 successfully for measurement and calibration.

As a follow-up, I contacted MOTU with the results of my tests. It turns out that all of the devices are somewhat older, and that their newer devices apparently have fixes for this particular issue. I haven’t had a chance to verify the results experimentally though.

My quest to get MIDI working on the Avid S3L

During large productions, my team uses QLab to play various sound effects, and to trigger snapshots changes on our Avid D-Show mixing console. I’d like to make use of the same triggers on the Avid S3L we use for our video mix, but unlike the D-Show it doesn’t have built-in MIDI.

According to the Avid Knowledge Base, the Roland UM-One MK 2 is officially supported, but that other class-compliant USB MIDI interfaces should also work. I don’t have the Roland, so over time I’ll try out various interfaces that I come across to see what I can get working.

If you know of a MIDI interface that works with the S3L-X, leave a comment and I’ll add it to the list.

Device Works? Tested Notes
MOTU 828mk3 No 2018-11-18
MOTU Stage-B16 No 2018-11-18
MOTU UltraLite-mk3 Hybrid No 2018-11-18 USB mode requires external power.

McDSP VENUE 6.4.0.15+ plug-ins don’t work on Avid S3L-X

I tried installing the latest versions of the McDSP VENUE plug-ins on my Avid S3L-X today, but they don’t work. The announcement says they are for the S6L, but all past versions have also worked on the S3L, so I’d hoped they would continue working. Alas, they don’t. Stick with the older 6.3.0.11 release.

The specific behaviour I see is that the plug-ins install successfully, as I’d expect, but when I open a show file using one of the plug-ins, they appear with the yellow/red triangle and are listed as “not available”. Plug-ins that I don’t have loaded in the current show file don’t even appear in the tree of available plug-ins, although they are listed on the plug-ins install page.

To get back to working plugins, I downgraded my plug-ins to the previously working version of the VENUE 6.3.0.11 bundle installer.


[Update: 2019-03-03] McDSP released a new VENUE S6L Installer v6.5.0.12. I tried it, and it doesn’t work. They continue not listing support for the S3L, so I’m saddened, but not surprised.

[Update: 2018-10-16] I fixed the link to the VENUE 6.3.0.11 bundle installer. I had incorrectly pointed to a nonexistent 6.4.0.11 version.

[Update: 2018-09-06] McDSP released a new 6.5.0.3 plug-in version for Windows that supports iLok Cloud. I tried this version (via manual install), and it also does not work.

[Update: 2018-09-04] I noticed today that the term “S3L” was removed from the VENUE installer on the McDSP Downloads page, and only the S6L is listed as supported for the 6.4.0.15 release. I’m guessing they have unofficially dropped support for the S3L, although I can’t find any other confirmation to that effect.

[Update: 2018-09-01] I tried manually downloading and installing the 6.4.0.14 version of the EC300 and NR800 plug-ins on a separate Windows machine, copied the installed plug-ins to a USB stick, and installed them on my S3L-X. This also did not work. (I’ve successfully used this method in the past to install the SA-2 Dialog Processor before it was included in the VENUE bundle installer, so I know it works.)

My Audioquest DragonFly Red works on macOS High Sierra again!

I don’t know what Apple is doing with audio timing in macOS High Sierra, but they have serious quality control issues in this area. See my post on Avid S3L-X, AVB, and macOS High Sierra for other troubles I’m having.

When Apple released 10.13.2, my Audioquest DragonFly Red started having strange issues. Similar to the clicking issue with AVB, I was having strange timing issues that sounded like phasing, almost like the individual waveform samples were being triggered at a different clock rate than the audio device. It wasn’t constant, but frequent and annoying enough that I gave up using the device. I wasn’t the only one having the issue.

In any case, 10.13.4 fixed my Dragonfly issues, so I’m again happy.

Avid VENUE S3L-X, AVB, and macOS High Sierra

Long story short, if you need use an Avid S3L-X with macOS and playback via AVB, do not install macOS High Sierra. macOS Mojave works fine, as does the older macOS Sierra, but High Sierra has clocking issues that manifest as constant clicking during playback, rendering the audio unusable.

If all you need to do is record via AVB, macOS High Sierra works without issue.

References

Note, all versions of macOS High Sierra through 10.13.6 are affected.

[Update 2019-03-03] I continue to have no problems with macOS Mojave (currently 10.14.3).
[Update 2018-09-29] Preliminary testing with macOS Mojave (10.14.0) and 64-channel recording and playback indicates that the AVB problems have been fixed.
[Update 2018-07-28]
Increased affected versions to 10.13.6.
[Update 2018-06-05]
Increased affected versions to 10.13.5.
[Update 2018-03-31]
Increased affected versions to 10.13.4.

Booting an Avid S3L-X remotely with Wake-on-LAN

E3 Engine

The E3 engine can be remotely powered on and started using the Wake-on-LAN protocol.

To remotely wake the E3 engine, you need three things:

  1. A computer that is connected to the same Ethernet network as the E3 engine.
  2. The MAC address of the engine. You can get the MAC address by going to the Options > Devices tab and right-clicking on the E3 engine image.
  3. The IP subnet address of the network. (Optional, depending on the software used.)

To shut the E3 engine down, use the VENUE Options > System > Shutdown button.

Software to wake the E3 engine

There are several software packages available to send the special Wake-on-LAN Magic Packet.

Mac

  • Wake On Lan by Depicus (Mac App Store, $1.99)
  • Remote Desktop (Apple, $79.99) – Also useful for controlling the S3L-X remotely.

Windows

  • MagicPacket by DecaTec (Microsoft Store, Free)
  • Wake On Lan by Sepiro Ltd (Microsoft Store, Free)

Command-line

For those comfortable with the command-line, a short Python script will also do the job. Save this script somewhere as wakeonlan.py and make it executable with chmod +x.

#!/usr/bin/env python
# https://apple.stackexchange.com/questions/95246/wake-other-computers-from-mac-osx

import socket
import sys

if len(sys.argv) < 3:
 print "Usage: wakeonlan.py <ADR> <MAC> (example: 192.168.1.255 00:11:22:33:44:55)"
 sys.exit(1)

mac = sys.argv[2]
data = ''.join(['FF' * 6, mac.replace(':', '') * 16])
sock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
sock.setsockopt(socket.SOL_SOCKET, socket.SO_BROADCAST, 1)
sock.sendto(data.decode("hex"), (sys.argv[1], 9))

Myself, I keep a copy of the script in my ~/usr/bin directory. To wake my system, I call the command like this, where 172.16.0.255 is the subnet of my network, and 00:90:fb:4a:13:9e the MAC address of my E3 engine.

$ ~/usr/bin/wakeonlan.py 172.16.0.255 00:90:fb:4a:13:9e

Stage 16 Box

The Stage 16 Box cannot be remotely power cycled without additional equipment. Some suggestions include:

  • Furman CN-1800S + Furman BB-RS232 giving control via Ethernet.
  • Furman M-8S (US) or the Furman PS-8RE III (Europe) connected to the GPIO connection from the E3 engine, along with an event (saved in the default show) to latch a GPIO when the system is started. Attempt only if you feel comfortable with electronics. If you would like me to build this setup and demonstrate it, send me an email.