Saturday, December 31, 2016
Is the German GEMA essentially the same as the MCPS? Does a band putting out its own CDs need to register with different people in different countries, or do these organisations cover all situations?
SOS contributor Tom Flint replies: GEMA performs pretty much the same function in Germany as the MCPS (Mechanical Copyright Protection Society) does in the UK. GEMA's full name (Gesellschaft für musikalische Aufführungs- und mechanische Vervielfältigungsrechte) translates as the Society for Musical Performing and Mechanical Reproduction rights. In other words, GEMA help songwriters, lyricists and music publishers obtain their royalties and, just like the MCPS, GEMA acquires these funds by taking a cut of record sales revenue in exchange for granting manufacturing licences to record labels.
In the UK, the MCPS licences usually have to be paid by the record label up front and are set at 8.5 percent of the price the label charges the distributor for each record (known as the PPD or Published Price to Dealer). The 8.5 percent is the writer's cut of the record's sale price, although writers who are signed to a publisher have to split their fee according to their publishing deal. If no dealer or distributor is involved, the figure paid by the record label is rated at 6.5 percent of the retail price, excluding VAT. GEMA operate in a similar way, although they take just over 9 percent of the PPD.
Other countries besides Germany also have their own versions of GEMA. In France, for example, there is SACEM, in Japan JASRAC, and in the US they have the Harry Fox Agency.
Quite whether you will actually need to deal with GEMA, or any other foreign agency depends on your location. According to the MCPS, licensing is not determined by the country of manufacture, but by the country in which the label is based. This means that if you are a UK-registered company it won't be necessary for you to get a licence from GEMA, even if you are using a German manufacturing company to make your CDs. The same is true if you are manufacturing CDs in the UK and exporting them to Germany. Obviously you could strike some sort of deal with a German label and have them release the record on your behalf, but it would then be up to them to obtain the relevant licence from GEMA.
It's worth noting that the MCPS are not the only collection society you need to consider contacting when releasing a record. There is also Phonographic Performance Limited (PPL), which collects licence fees for records played on the radio and TV and in pubs, clubs and other public places, and the Performing Right Society (PRS), which collects royalties from the public performance and broadcast of musical works (both recordings and live performances). Fortunately, both the PPL and PRS gather musical performance royalties from foreign countries on your behalf, so you don't necessarily have to sign up to the equivalent organisation in each and every country.
Friday, December 30, 2016
Thursday, December 29, 2016
By Hugh RobjohnsI recently purchased a second-hand Tandberg reel-to-reel tape machine and I'm having difficulties connecting it to my external hi-fi. I was provided with a lead that has a five-pin socket at one end and phono leads at the other, which I plug into the 'analogue in' socket on my hi-fi. However, when I'm playing tapes the music only comes out of one channel. The back of the Tandberg has two of these five-pin sockets and also three other holes, marked 'p up', 'amp' and 'radio'. Can you tell me how I can get the sound coming from both speakers and not just one? Any help would be most appreciated by this novice reel-to-reel owner!
SOS Forum Post
Technical Editor Hugh Robjohns replies: There are several possibilities here. The most obvious one is that the DIN-phono lead you have is broken. DIN is the Deutsches Insitut für Normung, a German standards-setting organisation, and it specified a range of connectors using a similar body with between three and 14 pins. The three- and five-pin versions were used a lot on hi-fi equipment in the '60s and '70s, before the RCA 'phono' socket became the standard interface, and now the five-pin DIN is most commonly found on MIDI leads. If you have a test meter, check the connections between the phono plugs and DIN pins to see if the cable is faulty.
For some bizarre reason, some manufacturers' implementation of the DIN wiring is exactly the opposite of others, so although I am giving the most common way of wiring them up, bear in mind that this is not always the case. The 5-pin DIN sockets were used to convey stereo unbalanced signals. The DIN pins on a male jack are numbered in the order 1, 4, 2, 5, 3, clockwise from right to left (see diagram). Normally, pins 1 and 4 were used for the left and right inputs, respectively, and 3 and 5 for left and right outputs, with the middle pin of the five (pin 2) serving as the common screen or earth connection for all four signals. If your DIN-phono lead only has two phono connectors on it, the centre pins of the two phonos will either go to 1 and 4, or 3 and 5 — a test meter will help you find out which.
The other possible explanations for why you're only getting output on one channel are broken electronics within the machine itself, or that you are trying to play a quarter-track tape on a half-track machine (or vice versa)...
You can check the latter by looking at the heads or making a test recording to a blank tape. A half-track head uses almost half the tape width for each channel, so you'll see the two head gaps occupying just under half the tape width, with only a small gap (guard band) between them. A quarter-track head uses slightly less than a quarter of the tape width for each track, and the two channels are separated by a quarter-track width, so the two head gaps are separated by the width of another head gap.
As for the 'p up', 'amp' and 'radio' sockets, this suggests that the machine has a built-in record selector and preamp. 'P Up' will be an RIAA phono pickup input, for example. 'Radio' is pretty self-explanatory, and 'Amp' is probably another line-level input — but it could possibly be an output intended to go to a preamp. It would be worth checking anyway!
Wednesday, December 28, 2016
By Tom Flint
I recently bought a Yamaha AW4416 digital multitracker and am in the process of getting to grips with its features. I'm going to be recording some vocals and guitars to the hard drive, but I use a lot of MIDI-sequenced sound modules for backing, so I am considering the option of sync'ing my sequencer with the AW and running the modules in time with the recorded material. Nevertheless, I'm still not sure if it would be more sensible for me to record the outputs of the modules to the hard drive. What are the pros and cons of each method?
SOS contributor Tom Flint replies: First, let us deal with the MIDI synchronisation option. External sequencers can be sync'ed to the AW4416 via MTC or MIDI Clock. There are several advantages of working in this way, the main one being that the sequenced part of a composition is always editable, right to the point where everything gets mixed down to stereo. This means that a basic MIDI arrangement can be developed hand-in-hand with recorded audio parts. Indeed, it's often the case that once a vocal or guitar arrangement is added to a composition, some part of the MIDI backing becomes redundant and needs to be removed or changed.
As far as audio routing is concerned, unless you want to use an external mixer, the setup requires the sound modules to have their own set of input channels and sockets in the AW4416, which can get complicated if you are regularly using the inputs to record multiple instruments to the hard drive. If you have a number of MIDI sources you might consider buying an analogue input card so that the modules can be permanently connected without obstructing the inputs you typically use for recording. For example, Yamaha's MY8AD and MY8AD24 YGDAI expansion cards (the latter being a 24-bit version), provide eight balanced TRS jack inputs, allowing four stereo modules to be connected and routed to input channels without troubling any of the standard ins.
The main problem with the MIDI sync approach becomes apparent if you have to take your recorder out of the studio to a session, to record a vocalist or instrumentalist, for example. If none of the sound module parts are recorded, you would have to either take the entire MIDI rig to the session, including sequencer, effects and modules, or you'd need to bounce all the audio sources to a spare couple of tracks in order to create a guide part (this is done by routing all the relevant input channels to a pair of busses, and then assigning them to a pair of record tracks).
One further drawback is that there are more chances for a non-recallable setting to get altered accidentally. For example, you could spend ages carefully setting up EQ, level and processor settings for each mixer channel, only to find the mix balance completely altered by a nudge of a sound module's volume fader, or tweak of a global parameter setting. The problem is likely to be compounded further if you continually have to unplug sound modules from the inputs so that vocals or guitars can be recorded. In such circumstances, the AW4416's preamp pots will need to be reset to their mix position after every recording session. Problems such as this are definitely worth taking into account if you are someone who takes equipment out on the road regularly, if you are likely to adjust the settings of certain modules from one song to another, or if you intend to use the AW for a variety of jobs.
I use an AW4416 sync'ed to a MIDI setup, and at the moment I record my MIDI gear to the AW because most of my songs have MIDI arrangements that were carefully prepared as backing for live gigs. I like the fact that once the sound modules are recorded they can be disconnected, freeing up AW4416's input sockets and channels for other uses, and it's comforting to know that a composition will not be damaged if I adjust a few module faders or preamp levels. I've also found it useful to have been able to take the AW to recording sessions across the country without needing to do any submixing or take MIDI outboard with me.
The recording method's most obvious drawback is that it uses up valuable audio tracks that could otherwise be use for massed backing vocals or other parts, although by making good use of virtual tracks and the bussing structure it is still possible to comp multiple backing parts — see the Yamaha AW4416 User Tips article in SOS June 2005 for more on how to do this.
Another problem arises if the recorder and sequencer have not been synchronised and a MIDI part needs modifying late on in the recording process. This is because the replayed MIDI part is likely to be out of time with the rest of the composition. Fortunately the AW's onboard editing tools make it possible to move audio in time.
The AW4416 User Tips features didn't explain how to do this, so here's an example, assuming that the part about to be replaced is a sequenced performance played from a stereo drum machine.
First off, record the new part to a pair of unused tracks or virtual tracks, but leave the old drums intact. Once recording is complete, the exact relative position of the old and new recordings needs to be established. To do this, stop the recorder on the original drum track's first distinct beat, navigate to the waveform page and use the data wheel to scroll to the beginning of the wave. Identifying individual beats visually is easier when the waveform display is set to a low resolution, and therefore has small vertical (Amp) and horizontal (Time) values. A low Time number also makes scrolling to the correct point considerably faster. Nevertheless, once the position marker is at the start of the beat, a finer resolution needs to be selected for precise positioning. I rarely use a time resolution smaller than 'x2048' for this purpose.
Next, press Locate and note down the exact time value. I like to double-check the position by looking at the waveform of the other track in the stereo pair The two should match, but off-centre panning can alter the point at which the waveform is visible, so it's best to be sure. If all is OK, the same beat-locating procedure can be used to determine the corresponding start point in the newly recorded drum track. Once again, note its position down exactly, making sure that the figure is taken when Time and Amp settings have the same resolution as was used previously.
Next it's time to start editing. For the sake of this example, let's say that the original drum pair began at 05.240 but the new drums start later at 05.365. It's then simply a matter of subtracting 240 from 365 to leave 0.125. Therefore 0.125 needs to be cut from the start of the new tracks. Before deleting anything, it's worth pairing the drum tracks so that any editing work done to one half of a pair is automatically applied to the other. Track, as oppose to channel, pairing is done by clicking on the relevant broken-heart symbol in either the TR Editing screen or the V Track page.
To make the edit, select Part rather than Track or Region, and choose to 'Delete' the relevant track numbers. Set the start point at zero and end point to 0.125. After the edit, check that the position of the waveform matches that of the old audio, and do a listening test to ensure the work has been successful. If you are playing back the new tracks next to the old there should be a degree of phasing and cancellation all the way through, thus proving that the new parts are about as close as they needed to be. The old drums can be deleted as soon as the new parts have been saved to disk.
On other occasions you may have to insert space rather than cut it to align the parts, but process is identical, except that, from the Edit Part menu, Insert needs to be selected rather than Delete.
So as you can see, both methods do have pros, cons and workarounds, the choice just depends on how you prefer to do things.
Tuesday, December 27, 2016
Monday, December 26, 2016
By Hugh Robjohns
I have just got hold of a CAD M9 valve mic and am concerned about the noise level from it. When I switch it on it chuffs and farts a bit, much as my Fender valve amp does, then settles down. That seems OK. But once it has warmed up, the background noise level seems high compared to my other condensers. If I have a vocal take at normal levels being recorded at about -6dB peak, then the noise level is registering at -38dB. The vocal sounds fine, but the noise seems high. Is this what I should expect from a valve mic? Should I try another valve in it?
SOS Forum Post
Technical Editor Hugh Robjohns replies: Valve mics are generally more noisy than solid-state condensers. The M9 is specified with a self-noise figure of 15dBA, which is roughly 8dB higher than the best of the large-diaphragm solid-state designs — the Neumann TLM103 has a self-noise figure of 7dBA, for example.
However, while it is possible that your mic is faulty or requires a new valve, the high noise floor you describe could also be down to poor mic technique.
With vocals peaking at -6dBfs, a noise floor of -38dBfs does seem poor. The question is, how much of that noise floor is due to the mic, and how much is due to the recording environment? Are you recording a low-volume source at a considerable distance, or in a noisy room, or with a poor-quality mic preamp, for example?
If you have access to another large-diaphragm mic, I would suggest you rig that alongside the M9 and adjust the gain to get the signal peaking at the same level for both mics, and then compare the background noise floors. If both mics deliver similar noise levels, then the room or your technique are at fault. If the M9 is more than a few dBs noisier than a solid-state large-diaphragm mic, then the M9 is in need of repair.
It could be that the valve is faulty or worn out, and certainly that's the easiest thing to replace yourself. However, there could also be a problem in the power supply or elsewhere in the mic's output circuitry, which would require a return to the supplier to be fixed.
Saturday, December 24, 2016
Friday, December 23, 2016
By Hugh Robjohns
I never did quite understand the subtle differences between all the different variants of Dolby — A, B, C, HX and SR. Could you explain them to me? Are there any others I've missed? What are Dolby Labs doing these days? I guess they've undergone some 'reduction' themselves...
SOS Forum Post
Technical Editor Hugh Robjohns replies: Dolby A was the first professional noise-reduction system — launched in 1967 if memory serves — and it used four separate frequency processing bands. You can think of them crudely as bass, mid-range, treble and high treble, with the top two overlapping so that the 'hiss region' was processed more heavily than the rest. Avoiding line-up errors between encoding and decoding was crucial, so the infamous Dolby warble tone was used to identify encoded tapes and to allow accurate replay alignment. Dolby A was originally used to get respectable audio performance out of early professional video recorders, but was later adopted for multitrack recording and cinema optical soundtracks.
Dolby B was a very simple domestic system intended to improve the performance of compact cassette recorders. It was also used on some later domestic quarter-inch machines. Dolby B was a single-band system affecting only the high end, with very modest compansion. It had no facility, or indeed any practical need, for replay alignment.
Dolby C was a much more aggressive multi-band version originally intended for small-format professional video-tape systems and narrow-gauge semi-professional studio multitrack recorders. It was very sensitive to mistracking, but was unfortunately designed without any line-up tone facility to calibrate playback levels.
In the professional market, Dolby A was superseded by Dolby SR, which was Dolby's most sophisticated multi-band noise reduction system. This employed 10 bands altogether, some operating at fixed frequencies and others moving automatically to suit the material, and allowed the user to achieve a signal-to-noise ratio of around 90dB from analogue tape. However, although it was a very clever and effective system it arrived just a few years too late and the digital revolution effectively eclipsed it. Dolby SR used a modulated noise signal for identification and replay alignment.
Finally, Dolby S (one you missed off your list) was a last-ditch attempt aimed at semi-pro and domestic recorders, and was a halfway house between Dolby SR and Dolby C. It still had no built-in line-up facility, though. It was used on some semi-pro narrow-gauge multitrackers and the last of the high-end hi-fi cassette recorders.
Dolby HX is not a noise-reduction system at all — it is a clever system to avoid over-biasing on analogue tape machines using high-output tapes. This system was used on some high-end domestic cassette recorders and the last of the professional analogue two-track machines, such as the Studer A807. Dolby HX is a once-only process that needs no decoding. In essence, it reduces the bias level if there is a lot of high-frequency content in the audio signal, thus preventing over-biasing and the noise artefacts and frequency-response errors that go with it.
Dolby Labs still make Dolby SR and A systems for analogue multitrack and cinema applications, and I guess they are still collecting licensing revenues from the other systems when they are used on domestic cassette recorders and the like. However, most of the company's efforts these days are geared towards digital data-reduction systems, which are based entirely on the frequency-masking principles first exploited by Dolby's analogue noise-reduction systems. That is why Dolby AC3 has always been amongst the best of the data-reduction codecs for a given data rate — the company had a major head start on the rest of the field.
Thursday, December 22, 2016
Wednesday, December 21, 2016
Tuesday, December 20, 2016
By Mike Senior
I'm having some problems with MIDI routing. I'm in the process of setting up a Yamaha RM1X sequencer, Alesis Micron and Access Virus synths, a Yamaha TG300 sound module and a Yamaha SU10 sampler. Could you explain how to set all this up so that the RM1X is sequencing the whole lot? Ideally I would like to use the Micron as both the controller keyboard and as a sound source. Any advice would be greatly appreciated.
SOS Forum Post
SOS Reviews Editor Mike Senior replies: The RM1X has just the one MIDI output, so the first thing you'll want to do is invest in a MIDI Thru box so that you can feed this output to all the other devices. You'll need one with at least four Thru outputs. The Kenton FB5 would do the trick — it can route two separate inputs to two sets of four outputs, or a single input to all eight outputs — as would the recently discontinued Philip Rees V4. Connect the Thru box's outputs to the MIDI inputs of the other modules.
Once all the units are connected to the RM1X's Thru box, you need to connect the MIDI output of the Micron to the MIDI input of the RM1X, allowing the Micron to act as a controller keyboard. You'll need to switch MIDI Local Off in the Micron so that its keyboard doesn't trigger its own internal sound module directly — you only want its sounds to be triggered via the sequencer. Finally, make sure the RM1X is set so that it echoes information at its MIDI input to the relevant channel of its MIDI output according to the RM1X's track selection — this setting will be somewhere in the RM1X's MIDI Setup page in Utility mode.
If you want to hear all the sounds at the same time, you'll need to get yourself a small mixer. If you don't need any channel processing (EQ and so on), then take a look at the compact 1U line mixers available from companies such as MAM, Rolls and Behringer.
Using a MIDI Thru box has an additional advantage. If you don't want to have all your gear switched on all the time, you can switch off what you don't need without having to rearrange your routing. Furthermore, if you ever need to pull a sound module out of your rig temporarily for any reason, it doesn't affect the rest of the setup.
Monday, December 19, 2016
Saturday, December 17, 2016
By Martin Walker
Can you recommend products suitable for the European power grid that can be used to clean up the power signal and ground loops? I am experiencing both ground loops and a generally dodgy power signal. A lot of people recommend that I use some sort of UPS (Uninterruptible Power Supply), but I don't need the functionality they provide, and I would rather spend money on better power conditioning and filtering equipment. Your advice will be greatly appreciated!
Alexander van Rijn
PC music specialist Martin Walker replies: In my opinion it's only worth 'cleaning up the power signal' if it's dirty, and a huge number of background noise problems are caused not by mucky mains, but by audio wiring that results in ground loops. This is the source of lots of unwanted nasties that sneak into your audio signals, and removing them often requires no dedicated products at all. Problems range from straightforward 'hums' (which normally include various levels of the mains harmonics, such as 50Hz, 100Hz, 150Hz, and so on in the UK, or 60Hz and higher multiples in the US), to a wide range of scratches, ticking, buzzing and other digital gremlins that are often associated with computer activities such as graphic redraws, mouse movements, and hard-drive activity.
If you're experiencing any of these ground-loop problems, you won't solve them by installing a power conditioner or an Uninterruptible Power Supply, so before you even think of spending money on either of these options you should examine your basic wiring. Temporarily unplug all the audio cables from your setup, and if you've got gear bolted into a rack, it may also be worth disconnecting the mains cables of this other gear to rule out problems with several metal cases touching each other and causing yet more ground loops.
As tempting as it might seem, short cuts such as leaving the cables plugged in and just switching off the connected gear at the mains won't work, since the mains cables and any resulting ground loops will still be in place. Unplugging one cable can therefore make the background noises better or worse, depending on how this affects the remaining ground loops. Only by removing every audio cable and working through your studio item by item can you totally eradicate ground-loop problems.
You should now hopefully hear silence from your loudspeakers or headphones, apart from a little hiss and possibly a tiny amount of hum or buzz if you turn the amplifier right up and place your ears nearby (be very careful when doing this, since an unexpected signal at this point could damage your ears or blow up your speakers). If there's still more hum than you expect, it might be due to a nearby 'line-lump' power supply, in which case, you should move this as far as possible from audio cables, and at the very least try rotating it to find the 'quietest' position. If you're still unhappy with the levels of hum and noise from your amp/speakers you may need to get them checked out by a technician — remember that hum levels of both solid-state and valve amps can increase over time, due to deteriorating capacitors or valves.
Assuming all is well at this stage, turn down the speaker levels, connect your mixer to the amp, turn up and listen again (if you route all your gear directly to a multi-channel audio interface, this is your 'mixer'). You'll probably hear greater hiss levels from the combined contribution of all the input channels until you pull the master fader right down, but there still shouldn't be any obvious hum or other interference. If there is, it's generally because you've just created an earth loop — the amp/speakers are already earthed via their mains cable, and the mixer is earthed in exactly the same way, so when you connect the two with an audio cable its screen connection completes the loop, causing unwanted earth currents to flow.
If your amp has balanced inputs and your mixer/interface has balanced outputs, the cure is to connect the two via balanced audio cables (twin core plus screen). If not, you may be able to achieve the same results by disconnecting the screen of an unbalanced cable at one end (in the case of soldered cables you can do this inside the plug, normally at the destination end). Similarly, if the amp has a balanced input, but your mixer/interface only provides an unbalanced output, you can make up a pseudo-balanced cable, as I described in 'Computer Audio Problems' in SOS November 2004. Here, one end of the balanced cable is wired to a balanced jack or XLR as normal, while the other end is wired to an unbalanced jack with the screen disconnected or, preferably, connected via a resistor. These cost only a few pence more to make than unbalanced cables, yet provide an ideal solution for connecting any unbalanced source to a balanced destination. I've got such cables wired between all my hardware synths and mixer, and background noise levels are considerably reduced as a result.
Occasionally the only way to cure a ground-loop problem is to install a line-level DI (Direct Injection) box between the mixer and amp, to 'galvanically separate' the two circuits, commonly by using a transformer to transfer the signal — the audio gets through perfectly, but there's no direct connection at all between the input and output cables inside the DI box. This is sometimes the only way to cure some laptop-related ground-loop problems, but in my experience, most others can be dealt with by cable modifications.
Once your mixer, amp, and speaker chain have an acceptably low level of background noise, plug each remaining item of gear into your mixer in turn and power it up, listening at each stage for unwanted noises. As soon as you hear any, you know you've either got a faulty piece of gear or a ground-loop problem, and can sort it out in exactly the same way as before. If it's rack gear, you may need to temporarily unbolt it from the rack to check that the problem isn't due to its case touching other earthed metalwork and creating a further ground loop (if it is, use nylon rack bolts or 'Humfrees' to isolate it). Alternatively, low-level circuitry such as mic preamps can pick up mains interference from the mains transformer inside a nearby rack unit. This systematic approach is the only way to deal with ground-loop problems. It may be tedious, but you only have to do it once, and the benefits can be enormous!
When you've got all your gear connected, and still have no hums or other nasties, then and only then is the time to consider adding a 'power conditioner' or UPS. A power conditioner will filter the mains signal to remove any radio-frequency interference plus any incoming spikes and other intermittent noises riding piggyback on the mains signal from the outside world. However, most modern electronic gear, including computers, already includes such filtering in its own power supplies, and in general, it's far better to suppress switch-related mains transients from distant devices such as refrigerators and central heating systems at source, as this will be far more effective.
If, after solving your ground-loop problems, you don't hear any other nasties then you probably don't need a power conditioner at all, but they can be very useful bolted into a rack for live use, to cope with unexpected 'incoming' problems due to stage lighting or grotty wiring in unfamiliar venues. However, if your mains power is 'generally dodgy' it may pay you to have an electrician check your house wiring, and contact the local electricity board to have your incoming mains checked for quality. If, for instance, you live in a remote rural location or close to an industrial estate, you may suffer from occasional but unavoidable interference problems that will benefit from a studio-based power conditioner, although I've never personally found the need for one (perhaps I've been lucky).
A UPS will, in addition, cope with 'brownouts' (occasional severe drop in mains voltage, generally for a few seconds only), plus the more severe 'blackouts' (complete loss of mains power), in exactly the same way as a laptop computer carries on running on battery power if you pull its mains plug. Even if you only use the UPS to power your desktop computer rather than the whole studio (generally a far cheaper approach), it can prove invaluable if you have paying clients in your studio, to avoid your computer rebooting in the middle of a session, and can give you a vital few extra minutes to save the current project before the UPS backup power runs out.
Friday, December 16, 2016
Thursday, December 15, 2016
By Hugh Robjohns
I'm looking to acoustically treat my home studio primarily for recording vocals. I've heard that putting acoustic foam tiles in the corner and singing towards the centre of a room is a good way of deadening a room for recording. But what sort of acoustic foam tiles would work best in this situation and how close could you get to the corner of the room without incurring unwanted low-frequency boost?
SOS Forum Post
Technical Editor Hugh Robjohns replies: Most rooms have a reverberation time that is long enough to become audible when recording. Many are also rather overdamped at high frequencies but underdamped at mid-range and low frequencies, simply because of the kind of furniture, floor and wall coverings we tend to use in domestic rooms. Ours ears quickly adjust to or ignore these characteristics, but when listening to recordings made in such an environment they become obvious.
The simplest and most cost-effective way to overcome this is to come up with a way of stopping reflections in the room from entering the front of the microphone, leaving the mic's inherent directional characteristic to reject sound from the sides and rear, so a mic with a narrow and consistent cardioid pattern works best for this.
By placing absorbent material in an arc behind the vocalist or performer, usually around the corner of the room, any sound that would normally be reflected from those walls behind the singer back into the mic gets absorbed instead. Thus less reflected sound reaches the front of the mic and the recording sounds much drier than it otherwise would.
Acoustic foam tiles will work well — especially the thicker ones — but in practice we have found that a high-tog duvet actually works better, both acoustically and practically. A kingsize duvet works out a lot cheaper, it can be easily hung and later removed without damaging the walls, and it generally works over a wider bandwidth than most foam tiles will. The thicker and more absorbent the material, the lower the frequency it will work down to, and by spacing the duvet an inch or two away from the wall it will work to an even lower frequency.
This approach won't do much to sort out the problem of standing waves — you'll need proper bass traps for that — but if you are careful about where you put the mic and roll off some of the bottom end (which you usually don't need when recording vocals and most other acoustic instruments), it is possible to get some very good results indeed.
It is true that you tend to get a bass build-up effect close to the corners of untreated rooms, but in practice, by the time you have hung a duvet, installed a singer facing back out into the room, and placed a mic on a stand in a convenient location, the mic will not pick up any of that bass boost.
This technique, although very simple, generally sounds better than a lot of home-made (and even some commercial) vocal booths too!
Wednesday, December 14, 2016
By Hugh Robjohns
What's the best way to get rid of tom ring (a ringing noise after the tom is hit) picked up by the tom mics on a recording of a drum kit? I did my best to cure the problem at source, but now some is on the recording.
SOS Forum Post
Technical Editor Hugh Robjohns replies: Obviously, it is essential to control excessive ringing from toms, snares, kick drums and whatever else is in your studio using damping before you start to record anything — it is extremely hard to fix what are inherently mechanical problems picked up on a recording using EQ, compression and other kinds of processing later on. It's also important to remember to check the sound of individual drums throughout the session to make sure nothing has changed.
As to the best approach in this situation, if the majority of your drum sound is based on the signals from the overhead mics, with the individual close mics simply adding impact and definition rather than the core sound, then you need to check the overheads by themselves to see how obvious and unacceptable the ringing is there. If the overall sound from the overheads is acceptable, then you can concentrate on treating the signal from the close mics so that the attack of the hit is preserved but the ringing decay is eliminated.
You can accomplish this using a gate or, at a stretch, the Strip Silence function found in nearly all DAW software, set with a higher threshold than normal. For preference however, I would use expanders on the close mics to chase down the ringing and subjectively tighten everything up. An expander is essentially a compressor in reverse — it decreases the level of low-level signals and increases the level of high-level signals, thus increasing the dynamic range of the signal. The SPL Transient Designer is a very good tool for this kind of job, too.
If the overheads are mainly delivering the sound of the cymbals, with the bulk of the drum sound coming from the close mics, then the situation is more tricky, as simply cutting off the toms' decay will sound obvious and unnatural. First, high-pass filter the overheads to remove as much of the drums (as opposed to the cymbals) as possible. Next, you are in for a lot of difficult work tidying up the individual drum sounds.
Assuming you have each drum on a separate track on your computer DAW or multitracker, you can manually edit each hit to replicate the action of an expander in reducing the level of the ringing, but with the manual precision that such treatment really needs.
Begin by editing the section between the start of each drum decay and the next drum hit, and reduce the level by 12dB or so (experiment with the degree of attenuation until you achieve the desired effect).
At the first edit point, straight after the drum hit, you'll need to set a crossfade time and shape that makes it sound like the drum is well damped, but has a natural-sounding decay curve. On the second edit point immediately prior to the next drum hit, you'll want a very short crossfade — possibly even a butt join if the spill isn't too bad.
With a little care and a lot of patience you should be able to fix it, and you certainly won't forget to check the sound of the kit before you start recording next time!
Tuesday, December 13, 2016
Monday, December 12, 2016
By Hugh Robjohns
I've just finished mixing a couple of tracks and I'm checking out how they sound on different systems. So far I've managed to get them to sound OK (well, acceptable anyway) on my Genelec 1030s, different sets of hi-fi speakers, a boombox, TV speakers and so on. I thought I was on my way to engineering stardom until I tried the tracks on a friend's system which has a subwoofer. Even with the subwoofer turned down a bit, the amount of boomy low bass was incredible and pushed the other frequencies to the back. Do you have any tips on how to prevent this from happening? I guess EQing out the sub-bass frequencies is the trick, but how do you know what you're doing if all these other sets of speakers don't give you a clue as to what's going on?
SOS Forum Post
Technical Editor Hugh Robjohns replies: Do not despair! I strongly suspect the problem is with your friend's monitoring system rather than your mix, especially as it sounds OK everywhere else. Genelec 1030s are pretty good at telling you what is going on at the bottom end, even if you are doing daft things with subsonic rumbles, and if nothing stands out as silly on them, then it tends to point to a bad subwoofer setup.
So, assuming your track doesn't contain stupid amounts of sub-40Hz bass rumbles, you need to help your friend to sort out the standing-wave problems in his room, and to set up his subwoofer properly.
If you can, check your mix on a real-time audio analyser (as found in Steinberg's Wavelab, above) to see what is going on below 80Hz or so. Make sure there are no excessive peaks in the low frequency area — compare your mix against commercial tracks in a similar music style to get a feel for what is 'normal'. I suspect you'll find your track is fine and it is the combination of the room and the subwoofer setup that is the real problem.
A handy way to check for standing waves and the resulting uneven bass response in a given listening setup is to record a simple sine-wave signal from a synth or sound module playing each note in turn over the bottom two octaves. Make sure that all the notes have the same velocity, and ideally, make each note a 'ping' rather than a constant drone.
Play that back over each system in turn and you'll be able to assess their ability to reproduce low bass evenly, and the effect of the room's standing waves. I expect that in your friend's room you'll find some notes set off huge resonant peaks and others disappear completely.
Standing waves are a common problem, and one we come across all the time on our Studio SOS visits, in letters and emails to Q&A and on the SOS Forum. The solution is to set up some proper acoustic treatment in the room and to adjust the positioning of the subwoofer. As a starting point, I suggest you read 'Monitoring & Acoustic Treatment' and 'Choosing & Installing a Subwoofer' by Mallory Nicholls. You also might want to read through some previous Studio SOS articles for some practical examples of room treatment, and searching the SOS web site for terms like 'acoustic treatment' and 'subwoofer' will turn up lots of useful information.
Saturday, December 10, 2016
Friday, December 9, 2016
By Hugh Robjohns
There are many decisions to be made when choosing a monitoring system. Infinite baffle, reflex, or transmission line? Active, powered, or passive? Bi-wired or bi-amped? We help you find the answers you need.
Choosing a monitoring system can be a difficult and confusing task, not least because of the enormous number of models and designs on offer. For a start, there are three basic classes of monitoring loudspeaker: infinite baffle (sealed box), reflex (ported), and the less common transmission line. Some monitors use a single wide-band driver, but most are two-way or three-way, while others use four or more drivers. There are also systems which require a separate subwoofer. And finally, three different amplifier arrangements are widespread: passive, powered, and active, along with bi-wiring options.
So let's have a look at the pros and cons of each of these designs.
Sadly, it's not quite that easy. Clearly, the rear of the cone is working against a much smaller volume of air than the front of the cone, and that volume of air is fixed. Consequently the loudspeaker cone feels a different degree of resistance when moving inwards than it does when moving outwards, which affects the distortion characteristics of the system as a whole. Internal resonances and standing waves can also be created within the cabinet, despite the use of lots of absorbent material, and this can produce various audible colorations in the sound.
Finally, the bass response of this kind of cabinet is relatively limited compared to that of other arrangements, for a given cabinet size, the low-frequency roll-off starting at a relatively high frequency. On the plus side, though, the phase response is very smooth, with relatively little phase shift, and the slope is also quite shallow, averaging 6dB/octave. Indeed, because of the shallow slope, even small infinite-baffle speakers can produce audible bass at surprisingly low frequencies.
For many, the infinite-baffle design is the most highly regarded and least compromised solution to loudspeaker monitoring. It is also interesting to note that the most widely used mixing references — the Auratone and the Yamaha NS10 — are both infinite-baffle designs. One of the most revered high-quality infinite-baffle designs was the infamous LS3/5A — a BBC in-house design dating back to the early '70s.
A couple of more modern and high-tech examples of the infinite-baffle loudspeaker are the K+H O 300D monitor, most AVI monitors, and the smaller ATC monitors. These speakers demonstrate the characteristically smooth, natural-sounding bottom end and associated mid-range clarity of the closed-box design very well. To many, what the infinite-baffle approach lacks in raw low-end volume, it more than makes up for in quality and transparency.
The vent may be located on the front baffle, it may be on the rear, and it may take the form of one or more round holes or slots. Most usually, the vent is connected to a tube extending back into the cabinet, the diameter and length of which are carefully calculated to achieve the required frequency response. Across a specific frequency range determined by the various parameters of the port opening, the sound from the rear of the loudspeaker cone is allowed to resonate through this port, emerging in the same polarity as the frontal sound to bolster the low-frequency response of the system as a whole.
The advantage of this approach is that it allows a much greater acoustic output at lower frequencies than the infinite-baffle design — you get a far more impressive bass response and overall volume level for the size of the box. However, there are a few disadvantages, one being that any resonant system smears transient signals over time. This can most clearly be seen on the waterfall response charts beloved of hi-fi magazine reviews, where one or more long resonant tails can usually be seen at low frequencies.
In monitoring terms, this inherent time-smearing and resonant behaviour can obscure small dynamic changes in the signal being auditioned, and may also reduce the transparency of the mid-range. In practical terms, a poorly designed reflex system can make it extremely hard to judge the relative levels of bass instruments properly, because their energy is stretched over time.
Another issue is the frequency- and phase-response characteristics of the port resonance. While the low-frequency roll-off point can be extended to a significantly lower frequency using a reflex design than with an equivalently sized infinite-baffle cabinet, the slope is far steeper, and the phase shifts far greater. Thus the level of bass output is greater down to the roll-off point, but then falls away much quicker, and a reflex cabinet is likely to reproduce very low frequencies at a far lower level than an infinite-baffle speaker. The inherently large phase shifts of this design also reduce (or at least affect) the naturalness of the bass end — although not everyone appears to be sensitive to this aspect of sound reproduction.
The extent and impact of these inherent disadvantages depends enormously on the competence of the reflex cabinet's design and what the designer was trying to achieve. There are many excellent reflex designs around, including the larger ATC monitors, all the Genelec models, various Dynaudios, Mackies, and Tannoys, and many others.
The Mackie monitors are an interesting sub-class of reflex design, though, because the port is covered by a passive radiator — in essence an unpowered speaker cone that reacts to the sound pressure inside the cabinet. This is a more complex arrangement again, sharing some characteristics with both infinite baffle and reflex designs — although it falls most comfortably into the latter camp.
- Blue Sky Media Desk: January 2005
- Blue Sky Pro Desk: July 2003
- K+H O 300D & Pro C28: October 2004
- M&K CR2401 & CR480: August 2003
- MJ Acoustics Pro Cinema 1 & Pro 50: December 2002
- NHT Pro M00 & S00: March 2005
- Triple P Pyramid: March 2004
Across most of the low and middle frequency range the transmission line is so well damped that all of the sound energy from the rear of the driver cone is completely absorbed and none of it reaches the outside of the cabinet. In that regard, it operates like a true infinite-baffle design — none of the rear sound reaches the listener. At very low frequencies, though, the line absorption becomes less effective and some very low-frequency sound reaches the end of the transmission line, much like the sound leaving a ported speaker. This allows a near flat response which extends down to at least an octave below any similarly sized reflex cabinet. One other advantage is that the overall frequency response varies very little with monitoring volume — the balance stays more or less constant regardless of listening level — which I personally find very useful.
Getting two drivers to match each other in terms of level, phase, and dispersion at the crossover point is far from trivial, and the on-axis frequency response of a loudspeaker is only one aspect of its performance that must be right. The relative phase through the crossover region is just as important, and the smoothness of the off-axis responses arguably more so — after all, most of the sound energy we hear in a room is reflected off-axis sound rather than direct sound. This is often what differentiates a really good monitor from a less good one.
Loudspeaker monitors have polar responses just like microphones or acoustic instruments. Some designers argue that a loudspeaker should have an omnidirectional polar response, and there are commercial designs built to do that — but in most typical studio situations a directional speaker works far better with typical acoustic treatment designs. At very low frequencies, speakers tend to radiate omnidirectionally because the wavelengths of low-frequency sound are generally far larger than the speaker cabinet. As the frequency rises, the cabinet starts to influence the dispersion of sound, and so the polar response starts to narrow into a more directional lobe. At higher frequencies still, the size of the driver itself starts to influence the dispersion, and the sound lobe reduces to something more like a beam.
Through the crossover region, the sound will be generated by both mid-range/woofer and tweeter, but given the relative size of the two drivers in relation to the wavelengths being produced, the woofer's polar response is likely to be very 'beamy', while the tweeter will have a much broader dispersion. Such a disparity in dispersion angles will cause a huge step in the off-axis frequency response, and consequently a very coloured off-axis sound. This is one reason why a speaker can sound very different when placed in a highly damped room than it does in a more lively, reflective room. It's only in the last twenty years or so that the importance of the off-axis sound and the careful matching of dispersion has been realised. So the width of the front baffle, the relative size of the drivers, and their crossover frequencies and filter responses are all chosen very carefully to optimise the response of the complete system.
Many systems these days employ waveguides around the tweeter to help control dispersion and sometimes to create different polar responses in the horizontal and vertical planes. This is usually to reduce early reflections from console and ceiling, and it's also one reason why turning a nearfield monitor on its side is not a good idea!
Designing a two-way speaker is hard enough, but most designers agree that a three-way system offers the best overall performance. Although there are two crossover regions to perfect, the disparity in size from woofer to mid-range driver to tweeter is much smaller, so the dispersion matching between adjacent drivers is easier. Each driver also has to operate over a much narrower frequency range, which enables each to deliver far better performance. Indeed, the improvement in the mid-range resolution and clarity of a good three-way system compared with a two-way system is very significant. Systems with additional drivers — four-way systems and systems with multiple tweeters, bass units, and so on — become a lot more complicated, and often the advantages are outweighed or at least balanced by the disadvantages.
- Mackie HR624: May 2002
- Mackie HR626: May 2004
- PMC DB1S: January 2003
- PMC TB2SA & DB1SA: May 2005
A passive loudspeaker is powered from a separate amplifier, typically installed some distance from the speaker and connected via a two-wire cable. Given the need to transfer power from the amp to the speaker and the relatively low impedance of the speaker itself, this cable has to have very low resistance and be able to carry large current pulses. Bell wire is not recommended, but any relatively substantial two-core cable will do. Two-core lawn-mower mains cable is ideal in most circumstances, and very cost-effective — far more so than the esoteric cables promoted in hi-fi shops.
Assuming good clean and tight connections and a competent amplifier, a passive speaker connected with respectable cable will perform very well. However, there are potential quality gains to made, if the speaker's innate resolution warrants it, by doing what is known as 'bi-wiring', where the tweeter and woofer are connected to the amplifier by separate cables. The passive crossover must be designed for bi-wired operation, and must provide separate pairs of terminals for each driver (normally linked with bars or brackets which must be removed for bi-wiring). This allows the amplifier to control the damping of each driver more effectively, as this bi-wiring separates the large sustained current flows to the bass driver from the smaller high-frequency signals. However, the crossover filter for each driver is still placed at the end of a long piece of connecting cable.
A related configuration is called 'bi-amping'. Here separate power amplifiers are used to drive the bass driver and tweeter. Each amp is fed with a 'Y'-cord so that it is amplifying the same signal through both channels, and is then connected to the relevant terminals of the speaker. The idea is to remove the interaction between bass and treble signals completely, but the practical disadvantages of this approach usually outweigh any performance gains.
Bolting the amplifier directly to the back of the speaker cabinet reduces the length of the speaker cable considerably, and thus also improves performance. The result of this approach is known as a 'powered speaker', but it is important to remember that the crossover circuitry is still passive.
By performing the crossover separation ahead of the amplification, it is necessary to power each driver with its own amplifier. This affords the second advantage, which is that the amplifiers' responses and power ratings can be tailored precisely to the speakers they are driving and driver protection systems can be easily built in. Most active systems intended for home studios are fully integrated solutions, so the cabling between the amplifier and driver is very short, although larger high-end active systems usually employ separate amplifiers with rackmounting active crossover units.
There are also some disadvantages with active designs, the most obvious one being that the number of amplifiers required has doubled or trebled compared to a passive design, and good amplifiers are inherently expensive. There are cost savings to be made in translating a passive crossover into an active one, but nothing like the amount needed to fund the kind of amplifier usually employed with good passive speakers.
In order to make active speakers for the budget end of the studio market, most manufacturers have to employ cost-effective 'chip amplifiers' — fully integrated designs — rather than traditional discrete circuits. Alternatively, many have chosen to use very efficient Class-D digital switching amplifiers, but in both cases the signal quality is often compromised in comparison with a good rackmounting amplifier. Whether these potential amplification losses are outweighed by the filtering and connection gains and careful system optimisation within any specific system is largely open to debate. However, overall I would say that, on pure resolution and transparency grounds, the losses generally outweigh the gains on most active speakers costing less than about £250 each in the UK, and true monitor resolution doesn't start to emerge until at least double that figure.
- ADAM ANF10: November 2004
- ADAM P33A: April 2005
- ADAM S2.5A: June 2003
- ADAM S3A: March 2004
- Alesis M1 Active MkII: August 2002
- Alesis Prolinear 720DSP: January 2004
- B&W DM602 S3: July 2002
- Behringer B2030A Truth: September 2004
- Dynaudio Acoustics Air Series: September 2002
- Dynaudio BM5A: June 2005
- Earthworks Sigma 6.2: April 2003
- EMES Black TV & Amber: September 2003
- EMES Pink TV Active: October 2002
- Event ASP8: April 2004
- Event TR5 & TR8: December 2003
- FAR OBS: February 2005
- Fostex PM1: November 2003
- Fujitsu Ten Eclipse TD512 & A502: June 2002
- Genelec 8040A & 7060A: December 2004
- JBL LSR6328 & LSR6312: May 2005
- K+H O 100 & O 800: November 2002
- KRK Rokit 5 & Rokit 8: August 2004
- KRK V Series 2: March 2005
- KRK V4: April 2002
- M Audio BX5: December 2003
- Mackie Tapco S5: February 2004
- Mission Pro SM6A: July 2004
- Mission Pro SM6P: September 2003
- Samson Resolv 80A: October 2003
- Tannoy Ellipse 10 IDP & TS212 IDP: June 2004
- Tannoy Ellipse: March 2003
- Wharfedale Pro Diamond 8.1 Pro Active: October 2004
- Yamaha MSP10 Studio: May 2003
- Yamaha MSP3: March 2002