High Impedance vs Low Impedance

August 07, 2016 by Fu Yongwei in audio

Are low impedance signals balanced? What is the difference between impedance and balanced / unbalanced transmission? In this post, let's take a look at how different rating of impedances will affect your audio signal integrity in the real world.

Before we go into the topic of impedance, I would like to point out that the impedance level of a transmission level does not dictate the type of transmission (balanced / unbalanced). A very simple scenario would be a low impedance microphone connected to an audio mixer with an XLR - 1/4 TS cable (unbalanced cable), thus creating an unbalanced output. Similarly, if the same low impedance microphone is connected through an XLR - XLR cable (balanced cable), the output of the microphone would be balanced. In this simple explanation, we can easily determine that the type of cable used will affect the type of transmission (balanced / unbalanced). (Refer here for more info)

Common examples of a high impedance equipment would be an electric guitar or a karaoke microphone. These equipment are usually seen in consumer products as they are able to provide a relatively higher output signal level as compared to low impedance equipment, hence, needing less amplification or gain. In contrast, low impedance equipment has a lower output level and often require input transformer before preamplifiers circuits in order to set up or strengthen the signal. This is one of the main contributing reason for most consumer devices to adopt high impedance output as in most cases, they do not require amplification after input (negligible) which in turn cut down on overall hardware cost (e.g. preamplifier circuits).

BAE 1272 Microphone Preamplifier, Credits BAE Audio

The main disadvantage of high impedance signal is that they do not perform well over long distance signal transmission (more than 10 meters). In addition, a high impedance line is adversely affected by the inherent capacitance that is present in the cable. This capacitance combines with the impedance of the source create a "low - pass filter"which progressively cuts higher frequencies (the longer the length of the cable, the lower the cutoff of the frequencies). Furthermore, high impedance lines are also more susceptible to the various forms of interference (e.g. high frequencies noise and radio). Therefore, due to these reasons, long cable distance in high impedance output are not practised in production work. Low impedance output, on the other hand, performs well in these situations, retain a higher rating of signal integrity. This is also why many professional grade audio equipment have adopted low impedance output.

An important note, the equipment impedance connected to the cable will affect the transmission line impedance. For instance, a low impedance microphone will lower the impedance of the entire line connected to it. Similarly, a high impedance microphone will have higher impedance line throughout the whole transmission (microphone to an audio console).

In order to preserve signal level, it is important to drive the input of an equipment (input impedance) with a source signal (output impedance) that is significantly lower. If the input impedance of a device is not significantly lower than the source impedance (output impedance), signal integrity will suffer (e.g. low SNR value).

When a signal needs to be split and sent to more than one routes, the impedance of the destinations provide additional paths for the electrical circuits. As a result, reducing the overall load impedance introduced to the signal. As a general rule of thumb, it is advisable to work on 10 : 1 load impedance ratio to help retain signal integrity. For example, a mixing console output with an impedance of 100 Ω needs to be sent to 4 amplifiers, each with an input impedance of 20 000 Ω, but can the signal be split passively? Note that we should try to keep loading impedance at the ratio of 10 : 1 or above and there are multiple input destinations.

ARX MSX 48 Mic/Line Splitter (Passive), Credits ARX

Hence, the signal could be split passively as the loading impedance ratio is well above 10 : 1. There are manly ways to split signals, one common way would be to use a passive cable split (“Y” cord cable) or to use a passive isolated mic / line splitter.

Passive "Y-Cord" Mic Cable, Credits Amazon/Hosa

YWFU

Live Recording with Yamaha R series interfaces

August 04, 2016 by Fu Yongwei in audio, music

Unlike many other audio interfaces used in music production, the Rio series I/O racks were designed to integrate with pro-audio equipment used in live and broadcast sound. A notable example is the incorporation of these interfaces into the Yamaha CL/QL system architectural. Unlike many audio interfaces that are purposefully designed for music production, the Yamaha R series interfaces do not have the onboard head amplifier (HA) control and these parameters are usually controlled through an audio console, such as the Yamaha CL5.

Yamaha CL and QL consoles, compatible with R series interfaces

On the other hand, parameters of the R series interfaces can also be controlled with R Remote, which is both compatible with Mac and Win platform. This has allowed the R series interfaces to be used in a recording situation without a console. In the post, we will take a look at how the R series interfaces can be setup for recording applications.

Since the R series interfaces are all equipment with Dante (Digital Audio Network Through Ethernet) capabilities, interfacing these boxes with your DAW do not require any additional hardware other than an ethernet cable with Dante Virtual Soundcard (DVS) and Dante Controller (DC) installed on your computer to route your audio signals into your DAW. DVS is a software that it converts your computer existing ethernet port into a Dante device of its own, allowing up to 64 channels of audio (bidirectional) can be utilised with a latency setting of 4ms, 6ms or 10ms.

After activating your computer ethernet port as a Dante device, Dante Controller will then be used to audio patching purposes. While it is possible for the user to route audio signals without affecting operation, Dante Controller offers users to save different show profiles (presets) and can be used to organise audio patch setting for different bands.

Once the audio signals have been routed to its respective inputs, you would need to patch the Dante I/O into the DAW of your choice. Since I am a pro tools user, I would briefly show the steps on patching the Dante I/O into pro tools. However, the steps are quite similar among other DAW platforms so it can still be applicable for other platform users.

Below is a live sample recorded from the Yamaha Rio 1608 as an audio interface.

No distribution for profit. Credit when sharing.

There is one caveat in my own opinion, although network latency can be configured to be less than 1ms (0.25ms) among Yamaha Pro Audio devices, the R series I/O racks will not be truly a standalone interface without some form of direct monitoring control. As mentioned earlier, DVS latency can be either set at 4ms, 6ms or 10ms, monitoring audio sources from DAW might be unsuitable for the musicians, especially when their performance consist of sixteenth note measures. Although latency can be improved with a Dante Accelerator card (0.15ms) to allow proper monitoring from the DAW, this will also incur additional cost.

The Yamaha R series I/O racks can be easily scalable to any system requirements. With a large array of supported equipment (e.g. Focusrite AM2 or Radial Di-Net), the system design can be very flexible for any type of applications.

YWFU

I finally made a switch with my studio monitors

July 31, 2016 by Jasper Chia in audio

Yamaha had made high-quality monitoring affordable for the masses with the HS series. After listening to a pair of HS-5 for four years back in Ngee Ann's EE department, I decided it was time for an upgrade. Selecting a model was the challenging part since Yamaha offered the HS-8, HS-7 and HS-5. I settled for the HS-5 due to space constraints although the HS-7 would be a better option. Upgrading from the existing MS-101 to these pair of HS-5 proved to be a tremendous step in sound reproduction quality. The first thing that caught my attention was the white woofer cones which were reminiscent of the legendary NS-10. Being a pair of active two-way monitors, the HS series had a clear advantage over my previous MS-101s. For once, you could hear your mix and edits with accurate rendition without much margin for guesswork.

As monitor speakers go, most of them have a very flat frequency response and minimal coloration. This pair is no exception. They do not produce much detectable coloration to the incoming signal and are faithful to the source. I do find them a notch brighter than other European monitors but fortunately, the high end is still smooth to the extent that one will not suffer from the dreaded listening fatigue after long hours of monitoring. The ‘brightness’ seems to reveal the more subtitle details in the soundtracks. That could be attributed to the use of a dome tweeter with a crossover frequency of 2 kHz. Bass is not compromised despite its compact size. I can still hear the kick drum and electric bass guitar distinctively on these speakers. However, their low-frequency roll-off is typical for bookshelf-sized monitors which are around 70Hz at -3dB due to the 5-inch woofer. Their performance will be greatly enhanced with the optional HS-8S subwoofer.

Yamaha HS8S Studio Subwoofer, credits Yamaha

The onboard amplifier seems to be a class AB design judging by the amount of heat coming from the rear heat sink. They are well designed to handle dynamic tracks without any distortion. This is typical for any piece of professional equipment. Floor noise is also kept to the minimal. Yamaha had also made EQ adjustments relatively easy by simplifying a bunch of knobs to two switches. These adjustments are used to compensate for discrepancies in the monitoring environment or listening room. One switch marked room EQ is used to attenuate the bass (500 Hz shelf filter) in steps of -2 dB or -4 dB. Another is labelled High-Trim which is used to adjust the output to the tweeter in steps of -2 dB, 0 dB or + 2 dB. Should these pairs sound too bright, that can be compensated by cutting the high trim by – 2 dB. I found that leaving the switches at 0 dB most suitable for my room as they produce the most balanced sound.

My room has another problem, though; the walls actually produced much exaggerated low-mid emphasis. That will have to be resolved by either shifting my table out of the corner or treating the walls. Inputs wise, these speakers accept both balanced XLR and 1/4” TRS plugs. The volume knob has markings and notches for professional line level at + 4 dBu and consumer line marked at – 10 dBV.

To summarise this review, I will make things simple here:

Pros

Value for money
Aesthetics
Relatively accurate
Minimal sound coloration
Flexible Inputs

Cons

Low-Frequency response could be better (But for this size, it is already good.)
A more neutral signature would be preferred. (A little bright)
Absence of trim pots for High and Low-frequency adjustment

But this is just from my perspective. If you want to simply hear the HS series, simply walk into a Popular bookstore (Serangoon Nex). Most of them have a pair of HS-5 with an HS-8S. However, some are using KRK monitors. So you can get a brief comparison from there. I do not quite prefer the KRK VTX series as they muffle the sound to an extent.

Overall, I will rate these pair of monitors 4.5 stars out of five. I would definitely recommend to anyone getting their first pair of speakers for production or simply music playback in their study.

Jasper Chia