New Zylia Software Releases Enhance Functionality of Innovative ZYLIA ZM-1 in 360 Sound Recording

Zylia, the Poland-based manufacturer of audio recording technologies, today announced a series of software releases for the ZYLIA ZM-1 single-mic portable recording studio. The company’s new ZYLIA Ambisonics Converter plug-in and new releases of ZYLIA Studio (1.7.2) and ZYLIA Studio PRO (1.3.1) software give musicians and other audio professionals even greater control and flexibility in capturing and working with 360 sound.

“Our ZYLIA Ambisonics Converter and enhanced ZYLIA Studio and ZYLIA Studio PRO software make it easier than ever to capture and output 3D/360-degree audio,” said Piotr Szczechowiak, Zylia co-founder and chief operating officer. “Leveraging the unique design of our innovative ZYLIA ZM-1 microphone system along with these new software offerings, we deliver unparalleled 360 audio recording and processing functionality at an unprecedented price point.”

The ZYLIA ZM-1 is a lightweight, compact, and elegantly designed recording solution that uses 19 omnidirectional microphone capsules to deliver up to 48 kHz/24-bit resolution while capturing the full spatial sound scene. The new ZYLIA Ambisonics Converter plug-in converts multichannel recordings made with the ZYLIA ZM-1 into Higher Order Ambisonics (HOA) full-sphere surround sound recordings. This software tool makes it easy for users to prepare 3D audio recordings for playback on the Facebook 360 and YouTube 360 platforms. To bring added height and depth to surround sound, the ZYLIA Ambisonics Converter offers a B-format signal set that includes first-, second-, and third-order Ambisonics. Users can choose between the FuMa and ACN component ordering formats, as required by the target platform, and between ambiX and TBE output formats. The converter also allows for quick correction of microphone orientation in postproduction.


Both ZYLIA Studio software (standard with the ZYLIA ZM-1) and the ZYLIA Studio PRO virtual studio technology and audio unit (VST/AU) plug-in have been enhanced with tailored spatial filters for a 3E microphone model so that they support the latest ZYLIA ZM-1 system design. In addition to bringing spatial filtering and automated signal separation directly into the digital audio workstation (DAW), ZYLIA Studio PRO now also boasts a new source localization function that makes it easy to find sound sources within the 360 sound scene. ZYLIA Studio PRO uses virtual microphone technology (software-defined microphones) on the 19-channel ZYLIA ZM-1 recording to separate sound sources and record them as individual tracks. With the plug-in’s new source localization feature, users can quickly add virtual microphones to just the right 3D position.

ZYLIA Ambisonics Converter

ZYLIA also has bundled its ZYLIA Control Panel with the ZYLIA ZM-1 driver for macOS operating systems. For ZYLIA ZM-1 users, this translates to more convenient brightness adjustment of the LED status light encircling the recording device.

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Broadcast Pix Commander Now Offers Full-Motion, Browser-Based Switching for BPswitch Systems

“Video production workflows are changing. The traditional control panel is still in use, but many organizations prefer a touchscreen approach,” explained Tony Mastantuono, product manager. “Commander has already revolutionized how people switch live shows. Now, your touchscreen isn’t limited to your control room – full-motion, cloud-based remote control video production has arrived.”

The Commander platform allows BPswitch customers to switch live productions from any location using a laptop, tablet, or smartphone. Ideal for corporate, government, religious, and educational facilities, Commander allows live productions to be switched from a seat in the audience, an office across town, or even a satellite facility across the country. The control-over-IP option offers low latency and is accessed through the BPNet™ IP ecosystem, which is powered through the ioGates cloud-based media management platform.

With its streamlined user interface, Commander makes it easy to execute sophisticated productions. An integrated PTZ camera control overlay, for example, is well suited for touch interfaces. Plus, file-based macros allow users to create scripted commands that produce complex combinations of on-screen elements, including motion graphics and DVE compositions with multiple sources.


The BPcommand Toolkit option allows users to create and modify custom interfaces for browser-based production, so users can create an elaborate interface for experienced TDs or a simple interface a novice can master in minutes. Broadcast Pix also offers a customization service.

In a closed network environment using a LAN, Commander is accessed through a browser securely using passwords. When connected to the internet outside of a LAN for remote access, no special router or firewall configuration is needed. Instead, a BPNet account routes control network traffic and live video through a secure “tunnel” to the remote browser.

Broadcast Pix also announced enhancements to its SoftPanel for BPswitch for local or remote use. With the same secure, browser-based access as Commander, the SoftPanel looks like a virtual 1000, 2000, or 5000 control panel.

The BPswitch family of integrated production switchers offers up to 22 SDI inputs and 12 outputs, with up to eight internal channels for clips and graphics. Other features include built-in NewBlueNTX multi-layer 3D motion graphics, dual-channel clip server, audio mixer, streaming to Facebook Live and other CDNs, recording, file-based macros, customizable multi-view, chromakey with virtual sets, and optional remote camera control. The updated Commander option will be available by the end of the year and priced at $800.

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Interoperability of FM Composite Multiplex Signals over IP

by Junius Kim – GatesAir

Two methods of MPX interconnection are digital and analog. Each present a tradeoff in terms of required network bandwidth, signal quality, and compatibility. Digital MPX over AES offers the possibility of an all-digital processing chain, while analog MPX can offer greater flexibility and compatibility with legacy equipment. With the recent emergence of digital MPX, there is a need for bridging and interoperability between newer digital MPX equipment and older analog FM plant MPX infrastructure.

In FM broadcasting, the multiplexed signal (MPX) contains multiple components such as main audio, pilot tone, and Radio Data System (RDS) signal. In addition, Subsidiary Communications Authorization (SCA) channels may be generated which are modulated onto higher subcarriers. The MPX composite signal frequency spectrum bandwidth varies depending on components carried, but at a minimum, with RDS, it is 60 kHz and can be up to 99 kHz.

Stereo generation can be performed in the digital domain. Such digital processing produces a discrete time representation of an MPX signal formatted for interconnection using AES3. MPX over AES is carried at a sample rate of 192 kHz. Due to the Nyquist frequency, MPX over AES is band-limited to approximately less than 80 kHz. Digital MPX offers higher RFI immunity, simpler connections and distribution than analog MPX.


A FM Studio-Transmitter Link (STL) using IP telecommunications can have several topologies. One common topology is where audio is transported from the studio to far-end transmitter site using an audio codec. The stereo generation is done at the transmitter site. Another possible STL topology is transport of the analog MPX signal using an MPX codec where the analog signal is digitized and processed. Such a codec can support different sampling rates and sampling word sizes. The sampling rate can be adjusted to transport the stereo audio plus RDS, or stereo audio plus RDS plus one or two SCA channels. This flexibility can adapt the codec to best suit the bandwidth of the STL IP network connection. The MPX over AES signal can also be transported on a codec supporting 192 kHz sampling.

STL transport of MPX offers several advantages over audio transport. The MPX generation process (stereo generation, RDS, and SCA modulation) is centralized and controlled at the studio site. With multiple transmitter sites, MPX generation is done once rather than being distributed out to the transmitter sites. With MPX over AES, a complete digital processing chain is preserved with no additional analog processes required.


When MPX over AES is transported over an STL, the codec performs an end-to-end, bit-by-bit copy of the signal with no alterations. It is only necessary to transport the AES3 left channel sample word across the STL. Other AES3 data such as parity, synchronization, and metadata can be regenerated at the transmitter site to save network bandwidth. One channel of 192 kHz, 24-bit words has a data rate of 4.6 Mb/s.

To reduce the network bandwidth for MPX over AES STL transport, techniques such as sample rate conversion and word size reduction can be used. Sample rate conversion is the digital signal process of changing the sampling rate of a discrete time signal to obtain a new discrete representation of the underlying continuous signal. The 192 kHz sampled AES signal can embody an MPX signal of up to 96 kHz. Often, the MPX signal does not contain information up to this frequency. For example, if the MPX signal contains stereo audio and RDS only, then the frequency content is up to 60 kHz. If the MPX signal contains in addition to this a single SCA channel (sub-carrier at 67 kHz), then its frequency content is up to 75 kHz. In these cases, the AES3 signal can be sample rate-converted to a lower sample rate, such as 132 kHz (stereo audio plus RDS) or 162 kHz (sub-carrier at 67 kHz) without any loss of information.

Modern state-of-the-art sample rate converters have excellent performance. The THD and dynamic range can be greater than 125 dB with a near constant group delay and an amplitude vs frequency response characteristic close to the Nyquist rate. Using sample rate conversion in an STL, the AES over MPX signal can be sample rate converted at the studio site, and then sample rate converted back up to 192 kHz at the transmitter site.
AES3 defines a 24-bit word size. The theoretical maximum dynamic range for a digital representation of an analog signal using uniform quantization is 6 dB per bit. Therefore, 24-bit quantization can provide 144 dB of dynamic range. As a practical matter, this is more dynamic range than can effectively be generated by the FM stereo generator, or utilized by an FM exciter. In most cases, word size can be reduced without loss of quality. With 16-bit sampling, dynamic range is still a robust 96 dB. Further reduction in word size is possible with some tradeoff in quality.

Using sample rate conversion and sample word size reduction, the AES over MPX STL network bandwidth can be considerably reduced. Using 132 kHz sampling and 12-bit sampling the network payload bandwidth is 1.6 Mb/s.


Analog MPX is compatible with most existing FM plant infrastructure, while digital MPX (MPX over AES) is a relatively new operating standard. For interoperability between the two, the MPX signal can be bridged from analog to digital, or vice versa. This is useful when interoperating between older FM equipment not supporting digital MPX and newer FM equipment that does. For example, an older FM stereo generator at the studio can interoperate with a new FM exciter through a MPX bridging device.


Conversely, a new FM stereo generator can interoperate with an old FM exciter through a MPX bridge.


This hybridized bridging device sits between digital and analog domains of operation. The bridging can be between co-located FM equipment or the bridging function can span an STL when it is integrated into a MPX codec.


The wide area network (WAN) payload bandwidth requirements for transporting an MPX signal varies based on sampling rate and word size. MPX over AES has a data rate of 4.6 Mb/s. This data rate can be reduced when down sampling from 192 kHz to a lower sampling rate and reducing the sample word size.

Transport of an analog MPX signal has flexibility in the payload data rate because of settable options for sampling rate and sample word size. The sampling rate selection is made based on the services needed to be carried across the WAN. For example, 132 kHz sampling can carry stereo audio and RDS data, 162 kHz sampling carries an additional SCA channel, and 216 kHz sampling embodies the entire 99 kHz MPX spectrum.

Usage of IP in broadcast applications is rapidly proliferating. IP based networks provide advantages in both reduced operational cost and flexible interconnection. The IP packetized transport of the MPX signal does add additional overhead for packet headers as well as delay associated with the packetization process. The packetization process needs to accumulate samples in a buffer prior to packet transmission. A higher number of samples in a packet results in lower overhead, lower packet rate and higher associated packetization delay. Conversely, the lower number of samples per packet translates to higher packet rate, higher overhead, and lower delay.

IP networks present impairments such as packet loss, jitter, and loss of network connectivity. Jitter can be mitigated by a receive jitter buffer. MPX streaming uses PCM encoding so lost packets results in missing data for that packet interval. Forward Error Correction (FEC) can be extremely effective for random packet losses, however if the packet losses occur in bursts its effectiveness deteriorates. Redundant streaming over independent networks (network diversity) or redundant time delayed streams (time diversity) mitigate against burst packet losses. FEC and network/time diversity can be combined, and streaming parameters determined, after packet loss pattern analysis.

Junius Kim is an engineer with GatesAir. and part the GatesAir design team responsible for developing Intraplex IP Link MPX, a next-generation MPX codec.