eTap2HW Emulations

Getting to the point of the ‘final end-game ‘adding the emulations intothe hardware DSP solution it’s time to re-address what’s going to be part of it.

  1. Echomatic I model ‘J’
  2. Echomatic I model ‘F’
  3. Echomatic II
  4. VOX Longtom
  5. PA304
  6. Roland 301 mode ’5′ with variable speed
  7. DOG (dream of glory emu )
  8. T.B.D possible Abbeyroad reverb
The Echomatics are now in calibration mode, the Roland is almost ready for calibrations.  Filtering is presently revisited to assure it can be calibrated with sense, not trial and error!
In all emu’s the tapenoise and W&F are in but ‘zero crossing’ distortion taken out due to space limitations. The first daughterboard loaded with the compiled EMU’s will be eminent and used for final calibrations. Before that, I’ve made arrangements to take my test-rig to Arwati who will play some Shad tunes I can use to fine adjust frequency curves. Remember that patch settings are just position’s  on the dials.
Skeleton code of both Echomatic-I and II are already offered to another ongoing project as a potential core delay model because  automation with pre-set patches of this DSP unit is rather easy..
I’ve also ordered a singleboard computer with LCD display and VU meters to be ready for an automation job try-out when the project ‘at hand’ is ready. This board features a microprocessor that includes also some DSP functionality.
/eof/

Adding the Haas stereo effect ala PK for your mix

A write-up on a possible way of using a reverb technique resembling PeterK’s way of using reverbs and opening up your stereo field for the guitar making it a more transparent experience seems to be correct from a timing point of view as the new eTAP2 B3 is with this feature build-in:

The main idea is to add additional echo’s to the left and right of the instrument (solo guitar) but with two different timings. It’s very much like a pingpong delay. The effect applied is called the Haas Effect as the discoverer of this physical phenomenon regarding the perception of sound by the brain.

Adding those reverb’s is also adding some more body to the instrument as they are all sourced from one source so about 3dB will be added overall. Remember that the resulted sounds are not matching 100% due to the complex processed nature of adding two audio streams so it could be a little bit less.

I’m explaining this as a post processing to show all steps in sequence. Real-time is also possible and advisable to hear directly the sound.

Firstly, the main EQcurve like the smile curve or the inverted smile curve should be applied and then the echo you need for the particular song. Now you should add an extra delay with left timing 300 mS (66% Dry/wet) and a right timing 350 mS (66%dry/wet). Here you use the dry/wet ratio’s to obtain the desired results. (Yes, you could also change the 300mS to another value but make sure the delta between left and right is maintained around the 20-50mS

[Note1] the maximum should be always  < 80mS. this value is based on complex signals but for speech it should stay between 2mS and 50 mS. So, Yes 80mS is allowed but start with the low-band <50mS first!

The delayed track can differ up to +/- 10dB in volume from the original. but start with -10dB

A more advanced way is to duplicate the [source+echo] to another track and firstly apply a high-pass filter to avoid frequencies < 400Hz. Then add the extra delays with left timing 300 mS (100% wet) and a right timing 350 mS (100% wet). Here you use the mix ratio [source+echo] / [additional echo] to obtain the desired results. A good starting point of -10dB for the [additional echo] is highly recommended. This advanced approach should provide a better “wall reflections” as low frequencies are not really reflected from walls so they should stay away to avoid a muddy sound. The best way to establish the ideal setting is to mute the [source+echo] and adjust the [additional echo] to just detectable. Then adjust the [source+echo] to your liking. By using this approach the reverb is never “overdone”

[Note1] The sounds will begin to be heard as distinct when timing exceeds >50mS this is called the Haas Effect Called after Helmut Haas reference: his doctoral dissertation “The Influence of a Single Echo on the Audibility of Speech”

/eof/

eTAP2 Engineer (b3)

Here is the new compiled build3 of the eTAP2 family.

This engineer version is with the PP function. The performance over the eTap2 build1 is improved compatibility, performance and some required upgrades. The eTap2 engineer-pro version will be,after testing also available.

On-line help and support is embedded into the serialized unit.

/eof/

eTAP2 Engineer-pro

It was time to upgrade the eTAP2 family with a new VST version as  computer processors are more sophisticated these days and therefore the loading minimized for this type of VST module I’ve added the following functionality:

  • Reverb
  • Ping-Pong ala Korving (was part of build 02)
  • Input gauge

The unit is now a little bit bigger and compiled with the latest compiler to make it even more compatible. The layout of the knobs is slightly improved

The unit is in Beta mode and I would like to ask for a volunteer to check it out and work with me to test/improve the additions. The actual CPU loading on my machines is less then 5%. One of the ‘machines’ is a virtual ‘Windows XP’ desktop running on my Apple mini!! Let me know per comment if you would like to be a ‘tester’.

/eof/

How does the eTAP2 VST preset manager works?

The preset-manager, part of the eTAP2 VST module, works with JPatch data stored in the module and is therefore read-only. This data is also copied in RAM memory so any change you make to the patches (or new patches designed) are only kept in the RAM being memory that’s freed-up when the module is switched-off again.

The best way to manage this is to use the load and store all programs facility.

Let’s assume you’ve changed patch 1 and would like to move on the the next patch before saving. In this case your first patch is deleted and replaced with the default patch! Here are some simple rules so you can achieve saving them:

Rule1: Any change on a patch should be saved before moving on (switch the locked LED to unlock and just switch the save button below the LED) Then if you go to the next patch and changed it then do this procedure again…

Rule2: When all changes are in: conduct a save all programs before you’re quit using eTAP.

Rule3: When you start a new session with eTAP just load all programs with your file and it’s all there to be used.

It’s also good practice to use a naming convention for each new set of (save all programs) patches and to keep the old text files for backup. I’m using programs120108.txt for a configuration I’ve saved on 12Jan08

 

/eof/

Fingerprinting echo devices

A 3D frequency versus time plot of a sweep frequency as input resulting into an measurable output is a good way of establishing the character of the echo unit under investigation. In this first example I’ve added some supported lines/comments to explain how to ‘read’ these fingerprints.

On the left axis is the frequency band and on the right axis the decay (or delay time in mS). The very first part around the 0 mS is the dry part component on it’s own as there is not yet delay added (replay heads output + dry). The gradation in color of the graph is per dB value so the curve can be translated to e.a. a filter program.

let’s have a look at some basic emulations I have here in my library:

Here, firstly, is the Meazzi model Echomatic 1′ve obtained from an eTAP2 emulation based on SPICE emulations of the original Meazzi unit’s schematics.

and here the Meazzi model Echomatic2

Now, as we’re getting more aquanted with reading the 3D graphs let’s have a look at a real Meazzi tape unit with 12ax7 tubes. I’m sure this is a surprise as the frequency bandwidth of the replay heads/tape are very narrow indeed. It’s possible that this unit was a restored unit and therefore cannot be used as a reference. I’m sure the guy owning this device feels pity for the ‘electronic’ users…should he?

Next an ‘authentic’ Meazzi Trannie. That’s not up to any standard at all! Seems hum is the main contributor. just thought of sharing this also to demonstrate the tremendeous   amount of spread in the existing hardware configurations around.

I have fingerprints of more Meazzi models in the library. both the Meazzi I & II are part of the eTAP2HW version and fingerprinting are part of the sound analysis to assure it sounds correctly. Remember our ear sound-per-frequency curve is getting lower and lower.
/eof/

An advanced W&F design

Thought of sharing this as it’s on the shelf but possible of interest:

Based on the low-cost W&F described in this blog this design is extended to control a PT2399 delay. The unit is based on a set of 8 different oscillators  generated by a single Tiny2313 microprocessor that can be switched per requirement into 4 selected frequencies to specific parameters covering all vintage requirements.The PT2399 delay is controlled by a single transistor and could also replace the first head of a DSP delay chain ( or time subtracted from the delay line). From a cost point of view this design is very affordable while the oscillations are reliable.

The nice daughterboard with the PT2399 is designed by “AnalagCustom” as part of a 6 ‘head’ echo/reverb device here is the link to the documentation.

This design is part of a more extended echo set and kept outside the scope of the eTAP2HW deliverables.

The code used to program the Tiny2313 microprocessor is to be found here. The actual timing is changed according the study results on capstan/rollers or disc behavior as described in this blog.

/eof/

A generic sound flow using digital equipment to obtain a vintage character.

The requirements using equipment based on DSP and BBD devices to bring the quality into the framework of vintage equipment is firstly a 12AX7 tube. In most cases a pre-amp based on this tube is more then adequate to ‘inject’ the harmonics to your sound chain. After this stage the sound chain is divided into a ‘dry’ and a ‘wet’ part as both require specific treatment. The equalizer (qty 2) is to be adjusted per channel to achieve an optimal result.

The Echo ‘wet’ sound channel is with required delays with exact amplitudes and followed by a W&F generator based on 3 flangers/ sin oscillators (Q2/Q20 only 2). After this, a tape simulation is added to obtain that specific tape sound and compression.

The ‘Dry’ channel is directly connected to the mixer together with the complete processed ‘wet’ signal to achieve the correct sound ratio. The processed ‘Wet’ signal is, after adjusting the feedback ratio, directed to the ‘Wet’ equalizer. This Equalizer is configured as a shelving filter[*] so a particular frequency band is slightly amplified (1-2dB) generating an ever darkening sound. In most cases both equalizers are put in series so the dB/octave is achieved by minimal components count.

[*] The shelving is achieved by a bandpass and a high pass filter where the HP filter’s slope is optimized for the total centre frequency. This is also a precaution to avoid motor boating of very low frequencies

 /eof/

Recording the Shadows

Part 1 Published with approval Wim (c)2010,2011

Many Shadows fans have tried to reproduce the vintage 60’s Shadows studio sound, many failed and only a few came reasonable close. Why is ‘That Sound’ despite extensive efforts by many so difficult to get?

More than a year ago I was fed up that I could not reproduce the sound I was after. So I decided to start an extensive research on this topic, with the goal to find what the missing link or links are to create ‘That Sound’.

As basis for testing I used the famous song Wonderful land which is very typical for the early sound.

To make things more clear I split the problem in three topics:

  •  The guitar with all its aspects, including Hanks skills. 
  •  The amp and echo unit. 
  •  The recordings at Abbey Road. 

Concerning the first topic, the guitar etc. the Shadows forums are full of tips dealing with topics like: tone wood of the guitar body and neck, electronics, tremolo, strings, playing technique, play position, plectrum etc. Those aspects are all important but in my view each adds only a relatively small contribution to the tone. Of course they can have a very bad influence on your tone if selected wrong.

In my case I concluded that I had to upgrade my Japanese Hank Marvin Strat, by replacing the electronics with standard Strat ones, changing the pickups to Kinmans en replacing the tremolo with a Calaham. These improved the tone and sustain a lot, but concerning the typical vintage sound relatively speaking, just a bit. Because this topic has already dealt with to a great extend by others on many forums /sites, I will not further go in to it.

The next topic, the amp and echo, I initial tried to solve with an amp simulator (POD 2.0) in combination with one of the modern echo units (Magic Stomp). With an amp simulator you can make further steps in the right direction, but I finally concluded that buying a Vox amp (Vox AC30C2X) and recording it with a mic (Shure SM57) would bring me further. But as many probably noticed themselves, the sound becomes better again, but still that typical vintage sound is not there yet.

The echo unit is of course one of, or maybe the most important unit to make the tone. Some people are lucky and still have a vintage unit, but unfortunately most don’t. End of the previous century the interest in shadows music strongly revived due to internet and by the beginning of the present century, many internet sites existed where people posted their own Shadows renditions.

Parallel to this development various people like Charlie, Piet and Jacob created echo patches for various modern echo units. These units did and still do an excellent job in reproducing the echoes from various Shadow songs. The drawback of these echoes however was that it where basically standard units, not specially designed to create ‘That Sound’. So although they produced the right echoes, they have all kind of limitations, especially if it comes to filtering. Some sound better than other due to the fact that more modeling is possible. So the echoes coming out of the units are ok, but they could not fully help us to get the right tone.

Various fans tried to solve this and have created additional equipment, such as filters (Cutting Edge filter, Ariab filter etc) with and without preamps, or used standard EQ’s, which gave in certain situations further improvements.

Only two types of echo units where specially designed for ‘That Sound’: The Atlantis and the more recent TVS units . These units are not only echo units but also have build in filters and other effects, attempting to come as close as possible to ‘That Sound’. Unfortunately the first is quite rare now and the second because of the special design and limited sales, quite pricy. So not too many people own these echo units.

I based my testing on the Magic stomp because this unit is considered quite good and just has the basic echoes (EFTP) required, with some filtering and modulation.

Now we come to the third topic, the recordings at Abbey Road. Very limited information is available on recording the Shadows and many people mention, when asked about it, the statement in the booklet from Roberto Pistolesi that: not much extra was done in the studio.

1 Another look at creating ‘That Sound’ 

I have developed some nice experience in the last years about recording and I never believed that ‘not much extra was done in the studio’ and I think that this whole point has been greatly neglected by most guitarists. This also makes sense because most guitarists, especially some years ago, had very little experience with recording.

In the sixties Abbey Road was already a famous studio and it is hard to believe that they did not use at least a nice portion of the technical possibilities, the effects, they had for recording the Shadows. Luckily recently a standard work came out called ‘Recording the Beatles’. This book describes the equipment and recording technique in the 60’s at Abbey Road in great detail. Although the Beatles started recording a few years later than the Shadows the whole period of the 60’s including equipment is described.

So I studied all recording techniques at Abbey Road and will describe here the main ones, which either have been used for sure or might have been used for recording the Shadows in this period.

 2 Abbey Road basic recording equipment

At Abbey Road Studios, many different ways to influence the sound where possible. We will now go through the key elements to see what their impact on the sound is or could have been:

2.1 Microphones 

The sound coming from the Vox amps speaker(s) was recorded with a microphone. At Abbey Road Studios, for the Vox amps Neuman U47 and U48 were use and after 1966 the U67. The U47 was switchable from cardioid to omni. The U48 is actually a smaller version of the U47.

Also on the amps sometimes Neuman KM 54 were used, which has a fixed cardioid. In the Echo Chambers Neuman KM 53 was the favorite, because it was a fixed omni. They are all condenser mic’s. For the Beatles each Vox amp was often recorded with another mic . If this also has happened for the Shadows is unknown, but you can just try it, if you record all three guitars, or even put two mic’s on front of your Vox AC30.

The above mics currently have an icon status. They are priceless for hobbyist, but there are cheaper mics close to their sound available.

Now the Shure M57, which is a dynamic mic, is popular with Shadows fans. It’s a cheap mic with a very neutral character. Condenser mics often have a smoother, silkier sound and it is advisable to experiment with condensers to improve your sound.

The U47 was thus both cardioid and omni with the following frequency responses:

 

From the frequency responses can be seen that the U47 had above 2 kHz a response up to about max +5 dB.

The KM 54 as you can see below has a very linear response.

2.2 Amps (Mic and Line)

The microphone signals went to tube amps located in the mixer desk, the REDD 37 desk and after 1964 the REDD 51 desk. The REDD 37 desk came out in 1958 and both desks were 4 channel stereo mixers.

Abbey Road Studios had a very un-orthodox standard for impedances. Standard both the incoming and outgoing signal was 200 ohms. The result was that a lot of line amps were needed. For example the REDD mixing desk 37, needed 31 Siemens V72S valve amps.

For the REDD 51 desk, the EMI REDD 47 valve amp was used. The Siemens V72S valve amps sound was described as more smooth and round, while the EMI REDD 47 valve amp sound was more aggressive and punchier. These amps will certainly have influenced the sound considerably. Valve amps/ valve mic preamps as we call them today for mics, all sound different. Recording engineers state sometimes that a mic makes 20% of the sound and the preamp 80%. This depends ofcourse on the quality of the mic and the preamp.

It is definitely worth, instead of going directly with your mic into your audio interface, to try to use a valve preamp before it. The best is a preamp where you can vary the impedance, because you can adjust your sound a lot with it. Also it is possible to buy a DIY board including parts list for an EMI REDD 47.

See here:

http://www.dripelectronics.com/index.php….&id=6&Itemid=33

2.3 Mixer REDD Desk 

These mixers had four channels and primitive EQ’s. Essentially you could just adjust the low and high frequency side. At the low end, there was a shelving boost or cut of +10 db/ -10dB at 100Hz in steps of 2 dB. On the high side was a shelving EQ, but a very special one. In boosting, the peak was at 5 kHz, but cutting the peak was at 10 kHz, with again the same range of +10dB / -10 dB in 2 dB steps. Also the settings of the low and high side affected each other a lot.

Furthermore there were two types of EQ which could be used in the mixer, the Classic and Pop EQ. They were small hardware plugin devices that could plug into the mixer desk. Through a hole in the mixer you could see what type was inside. The Pop version was obviously used for pop music.

The pictures give an insight into the behavior of the EQ.

 

 From the graph of the bass EQ you can see that here the possibility exists, in relation for instance to the Inverted Smiley curve and the Cutting Edge filter curve, to remove some extra dB’s. My testing has proven this to be needed to get closer to the vintage tone. Also because of the shapes of the curves, this potentially must have had a big effect on the creation of the vintage sound.

This EQ can be bought as a separate unit including an amp to recover the volume. See here:

http://phaedrus-audio.com/phab_phame_phi.htm

The EQ matches in all options the famous Pultec EQP-1A equalizer. For this EQ you can also buy a DIY board including a part list:

[url]http://www.dripelectronics.com/index.php?option=com_content&view=article&id=10&Itemid=36[/url]

This Pultec EQ has also been modeled in various plugins:

  • Nomad factory Pulse-Tec
  • Waves JJP PuigTec EQP-1A
  • Bomb Factory EQP-1A (mac only)
  • UAD Pultec EQP-1A

2.4 Echo Chamber/ Plate Reverbs

Hank obviously made use of his tape echo unit, but in order to make the sound fuller/thicker, Echo Chamber 2 and EMT 140 plate reverbs could be used. (The name Echo Chamber is not really correct, because it is essentially a reverb chamber). 

The Echo Chamber 2 at Abbey Road is essentially a small room of only 6.4m to 3.7m. The mono signal from the mixer was send to a speaker in the Echo Chamber, where they recorded the sound with one (in mono) or two mic’s (in stereo). The returning signal to the mixer was thus mono or stereo.

The reverberations of such a small room are quite small, with a short tail, but with technical tricks, it sounded for the listener as if it was much larger, more like a hall. To create a reverberation time (reverb time) as long as possible, the room was covered with tiles and the concrete was painted. Furthermore, there were sewer pipes upright in order to get maximum reflection and thus further enlarging the reverb time. The resulting reverb-time of Echo Chamber 2 was approximately 2-3 seconds, depending on the climatic conditions.

The plates were mainly used for mastering, but nothing prevents us from trying out plates also.

2.5 Single Tape Echo/ Drum Delay 

From the mixer, the recorded sound was passed on to the Echo Chamber 2 as a Send. The sound of the Send was delayed by a tape recorder, the BRT 2, see 2.7. Standard a delay of 120ms was used. A reverb with a Predelay of 120 ms is perceived by our ears as the reverberation of a fairly large hall. After 1965 there was also Feedback possible, see 2.8.

Let me just explain this:

If you’re in a room of this size and send a sound wave through a speaker at a short distance from a wall, facing the wall and record the first sound coming back after bouncing of the walls with mic (s) close to the opposite wall, then the sound based on the set up at Abbey Road and taking into account the reflection, has at least traveled about 10m. See picture.

 The speed of sound is 0.34m/ms. The sound wave requires thus 10/0.34 = 29.4ms to bridge that distance. This is the minimum Predelay of this room with this speaker / microphone(s) setup.

Add to this as discussed above an additional delay of 120ms and you will have a Predelay of 149.4ms. If you now calculate how big the room appears to the listener, this is (149.4 x 0.34) = 51m, so a large hall.

The reverb is quite different than an actual hall. You hear the fullness created by the Predelay of a hall, but the reverb tail is different and more like a large room. So you have a huge full sound but it is still very clear / transparent and not drowned in the tail, so a very special reverb.

According to the Beatles book at Abbey Road Studios this effect was in the sixties always on. Also the placement of the speaker and mics in the Echo Chamber 2 was optimized and was always kept the same.

Also for the EMT 140 plates, the incoming signal was delayed by 120 ms, but with a specially designed drum delay.

Based on the information above, you can, with a good reverb plugin, simulate the echo room reasonably easy. Eventually you will have to adjust the details, by ear.

2.6 RS 106 Echo Control unit-Band Pass Filter 

In order to make a great reverb sound you have to filter the sound beforehand. At Abbey Road as filter the RS 106 Filter was used, which could remove both highs and lows. This was a passive unit with the following settings.

 

 

There was also a dial to set reduction, ranging from 0 to -50dB.

The RS 106 Filters were sometimes also used as a replacement for the more limited EQ on the mixer.

The most commonly used frequency setting as Send was Bass Cut 600 Hz and Top Cut 10 kHz, but that does not mean that this was used for Shadows recordings.

With this information you can however try the same frequency steps in your DAW.

2.7 Tape recorders 

As tape recorders the BRT 2 were at Abbey Road Studios the most common used units. The first unit was from 1953 and they were used by EMI more than 20 years. They were mono units and at the end of the fifties, Abbey Road Studios had 53 units. In 1959 the BRT 3 was released which was a 4 channel stereo tape recorder. This machine became the standard for Abbey Road Studios for classic and pop music during the whole of the sixties.

The BRT 3 also had an EQ which was standard for tape recorders in these times. There were more standards but the BRT 3 used the CCIR standard with a tape speed of 15 ips (inches per second). This EQ gives by recording a Cut at 4500Hz and by playback a boost at 10 kHz. For the CCIR standard see pictures:

 

The reason for this EQ was to avoid tape hiss. Naturally this also had an effect on the sound on the high side.

The tape recorders were both used for recording as for echo effects (Single Tape Echo).

 2.8 BTR2 S.T.E.E.D Effect

STEED stands for Send Tape Echo/ Echo Delay. This is a further development from 1965 with the BRT tape recorders. This addition creates both the Single Tape Echo, so the 120ms, but also has a feedback, so you got a multiple echo delay and the sound was even fuller/thicker. The feedback loop had also an EQ, where you could set the frequency to remove the lows and highs. As an extra also two EQ, the EQ RS 92 and RS 127 EQ could be put in the loop.

For Beatles recordings after 1965, this effect was always used.

3 Additional equipment 

3.1 RS 127 Presence Box/Brilliance Control 

As already described the REDD mixers had a very limited EQ. In order to have more EQ possibilities in 1962 the RS Presence Box 127, called originally Brilliance Control was put into use. This was an EQ box with 3 possible frequencies of 2.7, 3.5 and 10 kHz and a range of +10dB / -10 dB in 2 dB steps. There were two versions (Rack and Box) of which one had a handle for ease of pulling it from a rack and the other could just be put on a table.

These EQ’s were widely used for recording of instruments, in the mixer, but also in the Sends to Echo Chamber2 or the Plates. Sometimes even two were put in series. The setting of +10 db at 10 kHz was quite popular to create air.

With the above frequencies and volume steps you can experiment a lot in your DAW. If you want the authentic sound you can use the plugins Abbey Road made for this recently.

 3.2 6MB 3.5 Audio filter 

This was an additional EQ that could be used in the send to the Echo Chamber 2, but now with a fixed frequency of 5 kHz and 6 dB. Not known is how often it was used and it may well be that it was used often.

So this you can try also in your DAW.

3.3 ADT 

ADT stands for Artificial Double Tracking. In Abbey Road it was customary to improve/thicken the sound, by double tracking it and using both tracks in de mix. Also Hank did this often, however in 1966, the ADT effect was created to do this automatically. This was done by a clever use of two existing tape recorders. Originally it was only used for vocals, but very quickly for almost everything. Essentially, it means that starting with a mono track, two tracks were made with about 40ms difference. This created the full sound of doubling a track.

By varying the speed of the tape recorders slightly in time and also because of the fact that they did not quite turn stable anyway, additional small variation in the time between the units were created, which further contributed to the sound.

 3.4 RS 92 Neumann Mic Equalizer 3.4 RS 92 Neumann Mic Equalizer 

This was actually an equalizer for the U47, but was also used for other purposes.

The EQ had three settings: Steep Bass Cut, Bass Roll-Off and Cut Extreme Bass. There was also a button with Boost and Extreme Flat Top. Boost Extreme Top did basically nothing and resulted in case the EQ was located after the U47, in the natural sound of the mic.

 The most common use was on the Return of the Echo Room 2 and the plates. Since in this Return the low-frequencies were already removed, the RS 92 at this location was probably only used in the Flat position.

3.5 RS 124 Altec Compressor 

The RS 124 Altec was originally the Altec 436B compressor from 1959, but was not suited to the Abbey Road Studios way of working and was subsequently adapted countless times. The compressor was used on everything, the guitars, in the mixer, as bus compressor and mastering. Rather subtle compression with a threshold of -3dB was often done with it, but sometimes extremes of -20 to -30dB were also common. The compressor gave a very clear coloration of the sound especially when it was used for high compressions. For the sound of the Beatles it was a vital part, so for the Shadows it could have been important too. For this important unit you can also buy a DIY board including parts list.

http://www.dripelectronics.com/index.php….&id=3&Itemid=29

If you want the authentic sound you can use the plugins Abbey Road made for this recently. There are three plugin variations of the RS 124.

3.6 RS 114 limiter 

The RS 114 limiter from 1956, was at Abbey Road Studios often used as an insert in the 4 channels of the mixer. For all the Beatles recordings in the period 1962-1964 it was used. Then it was replaced for vocals and drums by the Fairchild 660 limiter.

 4 Summary 

So all above equipment and describe applications can also have been used on Shadows recordings. Unfortunately there are no records of this on the web. But from above info you know what was possible at Abbey Roads and in your DAW many can be easily tried out.

In a next posting I will shows you the results of testing some of the above mentioned effects out on Wonderful land and can you hear the proof (hopefully) of the importance of certain effects to create the vintage sound.

Part2

Thanks everybody for their nice responses.

Based on the previous posting about Abbey Road equipment I did some test with the solo of Wonderful land.

1 Setup. 

My Strat goes direct into the Magicstomp and from the Magic stomp I go into an ART Pro VLAII compressor just for a bit of gentle compression and boosting of the volume. From the compressor the signal goes to a Samson D-3500 EQ, which is set to the inverted smiley curve of the Echomatic II. From the EQ I go straight into the VOX AC30C2X into the low input of the normal channel.

The Vox is set at Cut at 75-80%, Normal volume 12 past half, Master volume 10 past 12. Mic is at cap edge at a distance of 32cm. Mic goes direct in to the EMU 0404 USB audio interface, into Cubase 5.5.

In Cubase I had the original from 1961 and the BT made from the original, by removing the solo. So I could compare the original to the mix of my solo and the BT one to one.

On the solo I tried various plugins to try to get as close to the original sound as possible. For the reverb I used a FX track.

Because Wonderful land was already recorded in the spring of 1960 so certain equipment I described in the previous posting where not there yet, so this made the numbers of options for this song smaller.

2 Methodology. 

Every time I inserted certain plugins, I compared the sound by ear and when it improved I also compared the frequency response with a spectrum analyzer. This way I could quickly make progress once I understood where the differences came from.

Result. Download here:

http://www.4shared.com/audio/cpWO8QWM/WFL_LexHall_rev1.html

3 Plugins used.

The following 4 plugins were used to come to this result:

A compressor

 An EQ as amp

Two Filters 

 A Reverb 

There are three types of the RS124 compressor plugin. I have used type 60070B because this was used mainly for tracking. I used a threshold of about -20dB, what was in the usual range for tracking.

Because this compressor takes away a lot of volume you need an amp after it to recover the volume. As mentioned before the impedance at Abbey Road at the input of a unit was the same as the output and amplifiers were used to recover this volume loss. I use the SonEQ for this with the EQ in the neutral position. To simulate the vintage nature of the Abbey Road amplifiers I have put the input and output drive so that you create some extra saturation, so when playing the meter is between -3 and +1. This is called Soft clipping.

Next came the VPS Philta filter. The filter is a hi and lo pass filter which I used to create a similar effect as the Abbey Road shelving filters of the REDD mixer. Unfortunately I do not have one of the plugins, I mentioned in the previous posting yet, which are the same design as the REDD mixer EQ. However by ear I got very good results and for the time being the VPS Philta filter gave a very good improvement of the sound.

After this I compared the frequency response of the original with my mix and noticed that in the 3.5 KHz region my curve was higher than the original. So I used the RS127 EQ rack unit at 3.5 KHz with a 6dB cut.

I cannot directly explain the reason for the difference. The RS127 EQ did not exist yet at the time of recording wonderful land but because it has a vintage sound, it did the job.

After that I send the Solo to an FX channel on which I tried various reverbs. From the previous posting you remember that the predelay time of the Echo chamber 2 was determined at about 29,4ms. Previously I had a discussion with Charlie if Hank had doubled wonderful land. He told me that it was his opinion that this was not the case because in the whole recording a doubling signal with a fixed delay of 29.5ms was present. This you can clearly see especially in the palm muted part. I also measured the delay of those peaks and confirmed Charlie’s conclusions. However the question remained at that time how this was done.

Because now it is clear that the predelay of Echo Chamber 2 is the same as this fixed delay in wonderful land, the conclusion comes to mind that this must have been done with the Echo Chamber.

However if you use a reverb with this predelay and a reverb time of 2-3 sec the reverb sounds less full than on the original.

As you could read from my previous posting, it was quite common, to add an extra predelay of 120ms with the BRT2 tape recorders, making the total predelay say at 150ms (rounded off). So I added this too. Now the reverb comes in line with the original, even with all reverbs I tried, despite the sometimes different way of defining the main criteria on the GUI.

So it looks like they used the Echo Chamber during tracking without the 120ms extra predelay and later during mixing they used the Echo Chamber again, but now with the extra predelay of 120ms.

The ratio wet dry I determined by ear and by looking at the frequencies responses. If you add too much wet you see the curve coming up on the high frequency site. Now the only problem remaining was that the reverb still was to dark. I noticed from the frequency response curves that in my mix around 2000Hz, two peaks where quite lower than in the original.

A good chance was that this was caused by the settings of the RS106, the EQ in the Send to the Echo Chamber Looking at the frequency options of the RS106, I decided to use as bass cut 1600Hz and as top cut 2000Hz. Now by increasing the wet signal slightly the frequency responses became nearly equal and the reverb sound also.

This EQ setting of the RS106 is quite interesting and might have been used more frequently on shadows recordings.

4 Some food for thought.

As you may already know, the highest peak at Shadows numbers typically lays between 800-1000Hz, which is usually the second harmonic. So if your set your hi and low filter in the send to the reverb (or in the gui of the reverb) at 1600-2000Hz, what you essentially do is that you set the peak of the reverb sound on the fourth harmonic.

Maybe this was a standard technique because the fourth harmonic which is called a ‘Perfect Fourth’ is a very beautiful sounding harmonic.

I am going to try this out in future recordings.

5 Final conclusions. 

In my view I am quite close now on the main parts of Wonderful land.

Further improvements will of course still be possible but must come from some of the many details, like string size etc. as mentioned in the beginning of the first posting. The intro is the exception and sounds still a bit more different. This is probably because it was processed even more. My suspicion is that a more narrow filtering was used, possibly also by using the Echo Chamber more often. Later I will try one of the Pultec plugins to see if that does the trick.

Finally, I hope that I have delivered some evidence that it is important when aiming for the vintage 60´s sound to use a RS124 compressor, some proper filtering in line with the REDD mixer and the parameters of the Echo Chamber 2 for the reverb. I started the first posting by saying that I hoped to find the missing link for creating That Sound. From the results it looks like that the REDD mixer EQ might be the main missing link.

So further testing with one of the Pultec EQ plugins might be useful to support this conclusion.

(c) 2010,2011 Wim aka Fenderwim

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Sample rates, introduction to digital signal processing

When we discussed limiters it became clear that there are problems understanding the basics of digitizing sound including the post processing towards the red book CD quality. There are plenty of sources on the web with in-depth explanations [wiki: samplerate]. This write-up is just my form of formalization so I can understand it better myself,because I wrote it,within the reach of our hobby in mind.So it possible could also help you understanding the greater picture.

The actual transfering from a analoq wave to a digitized form was investigated at the Bell Labs by a guy called Harry Nyquist who discovered that to capture the entire analog waveform, rather samples of the wave could be taken at various points. He also found that in order to have enough information in the sample pool to flawlessly reconstruct the original waveform, the sampling rate must be at least twice the signal bandwidth.

Humans can hear sounds with frequencies from 20 Hz to 20000 Hz theoretically, although a large percentage cannot hear much above 16000-17000 KHz. For practical purposes it can be said that the bandwidth of signals perceivable by man is 20000 Hz (20 KHz). For audio recording, additional headroom has been added, and the bandwidth measures 22050 Hz.

Based on this Nyquist law of sampling, we need a sampling frequency of 2 x 22050 = 44100 Hz in order to reproduce sounds/music perceivable by man faithfully. That’s where the 44,1 KHz value comes from.
This is regarded as the CD standard. If you also include the wordlenght of 16-bits per sample the signal/noise level would be ~96db of the total dynamic range you have the total specification here!

Sample rates of 96 and 192 KHz are starting to become more common, particularly in DVD-Audio, but many people honestly can’t hear the difference.

Higher sample sizes allow for more dynamic range so louder sound louder and softer softs. If you are familiar with the decibel (dB) scale, the dynamic range on an audio CD is theoretically ~96db, but realistically signals that are -24 dB or more in volume are greatly reduced in quality.

Most (semi) professional sequencers like Cubase supports two additional sample sizes: 24-bit, which is commonly used in digital recording, and 32-bit float, which has almost infinite dynamic range, and only takes up twice as much storage as 16-bit samples.

Just a sidestep, If we use lower sampling rates, for example, 12kHz, we can’t represent a sound whose frequency is above 6KHz. In fact, if we try, we’ll get usually undesirable artifacts, called foldover or aliasing, in the signal. This approach would be unacceptable for producing high quality sounds but could be used to deform the waveform to an extend that it could be a form of distortion/overdrive. Obviously, as we would use 12kHz sampling the solo guitar it’s result would be sounding rather acceptable due to it’s bandwidth.It is normal to apply a lowpass filter to this signal to assure that the frequencies being higher then12Khz/2 are removed so artifacts outside the range of 6Khz are removed. You see, it’s rather easy to understand as there are no more tricks left anymore!

To complete, to bring a digital signal that’s say 24bit@44100 (or higher) or 32bits@44200(or higher) back to 16bits@44100 the dithering technique is mandatory to assure your digital signal is a perfect fit for the CD.

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