Many thanks for the interest and the support
I’ll try to clarify the spirit of my initiative (if my English will be enough).
When I was Young (now I’m 50) I had a band too and I remember that the sound was less powerful but much better than that actually produced by modern systems. In particular, the listener could clearly listen the voice, follow the bass phrasing and identify the sound of individual instruments.
I have listened numerous bands in the last decades and all were penalized by a similar unbearable defect: the power was very high BUT the quality …too low. It was almost impossible to identify the individual instrument or to follow a bass phrasing or to clearly hear a clean voice (…yes I think that “clean” correctly translate the Italian terms “limpida”).
I believe that the reasons for such deterioration are essentially two:
1) The use of solid state devices employing many (too much) components in the signal path.
2) The common output.
I’ll try to explain these points.
1)The old systems were all tube-based ones. They were not powerful nor perfect. However, they were simple in the schematic and warm in the sound (I think this is a peculiar characteristic of tubes).
2) Each instrument had its own amplifier… including the voice/s. This approach led to a better intelligibility of the individual sounds. This concept is easily verifiable listening a guitar being reproduced alone or together with other instruments (also in a musical traces).
In the last 5-6 years, I have strived to teach my sons and their friends to distinguish Music from noise, better … power from quality. In this view, I organized a course of Electronics for children and, together, realized a very good HiFi system (Impianto HiFi | Agorà ) also reported on diyAudio (http://www.diyaudio.com/forums/tubes-valves/73513-intriguing-300b-se-direct-coupled-amplifier.html ).
Now, I have listened several times my son and his band … as well as other bands, and all have the same weaknesses … a feeling of confusion, without the possibility to identify individual components, and with a feeling of discomfort.
That induced me to embark in the mixer project.
As you have noted, the general setting of our project is different from the actual systems. The actual systems share the sounds on the basis of the frequency range, assigning to each group of frequencies specific amplifier apparatus (sub, mid, high). In some cases (many) that separation is even lacking and the system is based in the separation into two-three channels (right, left, sub). In any case, there is always a mix of instruments and voice/s.
In our setting the mixer presents 4 separate and independent sections, each having a own preamplifier, amplifier, and speakers. Specifically … voice, bass, right channel, and left channel. I strongly believe that voice and bass cannot be together in the same circuits and they cannot be reproduced by the same speakers. Both these components must have separate and specific preamplifiers, amplifiers, and speakers.
Finally, I think that at least the role of voltage amplifying (preamplifier and mixer) should be assigned to the tubes (that do this role better that solid state devices).
I have to confess that we have not yet taken a definitive decision about the project. Your considerations will have a relevant heavy on our final decision.
In any case, I (and my son) thank all for the interest and the very useful comments and support.
Ciao
Antonio e Francesco
I’ll try to clarify the spirit of my initiative (if my English will be enough).
When I was Young (now I’m 50) I had a band too and I remember that the sound was less powerful but much better than that actually produced by modern systems. In particular, the listener could clearly listen the voice, follow the bass phrasing and identify the sound of individual instruments.
I have listened numerous bands in the last decades and all were penalized by a similar unbearable defect: the power was very high BUT the quality …too low. It was almost impossible to identify the individual instrument or to follow a bass phrasing or to clearly hear a clean voice (…yes I think that “clean” correctly translate the Italian terms “limpida”).
I believe that the reasons for such deterioration are essentially two:
1) The use of solid state devices employing many (too much) components in the signal path.
2) The common output.
I’ll try to explain these points.
1)The old systems were all tube-based ones. They were not powerful nor perfect. However, they were simple in the schematic and warm in the sound (I think this is a peculiar characteristic of tubes).
2) Each instrument had its own amplifier… including the voice/s. This approach led to a better intelligibility of the individual sounds. This concept is easily verifiable listening a guitar being reproduced alone or together with other instruments (also in a musical traces).
In the last 5-6 years, I have strived to teach my sons and their friends to distinguish Music from noise, better … power from quality. In this view, I organized a course of Electronics for children and, together, realized a very good HiFi system (Impianto HiFi | Agorà ) also reported on diyAudio (http://www.diyaudio.com/forums/tubes-valves/73513-intriguing-300b-se-direct-coupled-amplifier.html ).
Now, I have listened several times my son and his band … as well as other bands, and all have the same weaknesses … a feeling of confusion, without the possibility to identify individual components, and with a feeling of discomfort.
That induced me to embark in the mixer project.
As you have noted, the general setting of our project is different from the actual systems. The actual systems share the sounds on the basis of the frequency range, assigning to each group of frequencies specific amplifier apparatus (sub, mid, high). In some cases (many) that separation is even lacking and the system is based in the separation into two-three channels (right, left, sub). In any case, there is always a mix of instruments and voice/s.
In our setting the mixer presents 4 separate and independent sections, each having a own preamplifier, amplifier, and speakers. Specifically … voice, bass, right channel, and left channel. I strongly believe that voice and bass cannot be together in the same circuits and they cannot be reproduced by the same speakers. Both these components must have separate and specific preamplifiers, amplifiers, and speakers.
Finally, I think that at least the role of voltage amplifying (preamplifier and mixer) should be assigned to the tubes (that do this role better that solid state devices).
I have to confess that we have not yet taken a definitive decision about the project. Your considerations will have a relevant heavy on our final decision.
In any case, I (and my son) thank all for the interest and the very useful comments and support.
Ciao
Antonio e Francesco
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As requested, I include some helpful requirements for the tube mixer
1)Type of system:
- for live music,
- band structure: drums, 2 guitars, bass, keyboards, sax, voices.
- type of music: pop/rock
- power produced: a total of 3-6000 Watt.
2) Subdivision of the channels:
- Voice channel (600-1000 Watt),
- right channel (600-1000 Watt),
- left channel (600-1000 Watt),
- Bass channel (1000-1500 Watt).
3) Inputs:
- Modular input section (a single board for each channel),
- 10 input channels (for the actual necessities),
- possibility to increase the number of channels.
4) Basal philosophy
- General: tube mixer + solid-state power amplifier + passive speakers.
- Tube mixer: it should include only few and essential elements. Specifically:
a) Elements for individual channel: gain, fader, bass, treble, input mic, input line, low-cut filter, phase inversion, vu-meter, channel selection (right-left-bass-voice), balance (between right and left channel), monitor and headphone output, send and return access.
b) Common elements: master control, vu-meter, send and return access, echo.
Status of works
Some choices have already been done and experimentations carried out.
- Each channel has an aikido makeup gain preamp ahead of a passive Baxandall filter.
- Each channel has an insert that bypasses the makeup amp and filter.
- 3 main buses, 2 mono called "voice" and "bass" and a stereo LR with a pan control.
- The monitor is set up as 3 mono sends, 1 headphone send, pre-EQ and pre-fader (despite what's shown on the other diagram...).
The Aikido board has already been tested and the combination ECC85-6DJ8 or 6DJ8-6DJ8 gives a gain of 29 and 14 (respectively) with a frequency-range of 5Hz-75KHz (+/- 1 dB).
See general schematic
1)Type of system:
- for live music,
- band structure: drums, 2 guitars, bass, keyboards, sax, voices.
- type of music: pop/rock
- power produced: a total of 3-6000 Watt.
2) Subdivision of the channels:
- Voice channel (600-1000 Watt),
- right channel (600-1000 Watt),
- left channel (600-1000 Watt),
- Bass channel (1000-1500 Watt).
3) Inputs:
- Modular input section (a single board for each channel),
- 10 input channels (for the actual necessities),
- possibility to increase the number of channels.
4) Basal philosophy
- General: tube mixer + solid-state power amplifier + passive speakers.
- Tube mixer: it should include only few and essential elements. Specifically:
a) Elements for individual channel: gain, fader, bass, treble, input mic, input line, low-cut filter, phase inversion, vu-meter, channel selection (right-left-bass-voice), balance (between right and left channel), monitor and headphone output, send and return access.
b) Common elements: master control, vu-meter, send and return access, echo.
Status of works
Some choices have already been done and experimentations carried out.
- Each channel has an aikido makeup gain preamp ahead of a passive Baxandall filter.
- Each channel has an insert that bypasses the makeup amp and filter.
- 3 main buses, 2 mono called "voice" and "bass" and a stereo LR with a pan control.
- The monitor is set up as 3 mono sends, 1 headphone send, pre-EQ and pre-fader (despite what's shown on the other diagram...).
The Aikido board has already been tested and the combination ECC85-6DJ8 or 6DJ8-6DJ8 gives a gain of 29 and 14 (respectively) with a frequency-range of 5Hz-75KHz (+/- 1 dB).
See general schematic
An externally hosted image should be here but it was not working when we last tested it.
and preamplifier-tone contro sections
An externally hosted image should be here but it was not working when we last tested it.
Because I do not have a software to test it. The simulation software (tonestack) has sveral topologies of tone control (Baxandall included) but not the subtype of that with feedback.
In addition, I do not love feedbacks and, when possible, I avoid it.
Finally, the feedback requires a further tube (one for each channel). It is not possible to implement the tone control between the first and the second section of the actual schematic, but only after the second stage.
In addition, I do not love feedbacks and, when possible, I avoid it.
Finally, the feedback requires a further tube (one for each channel). It is not possible to implement the tone control between the first and the second section of the actual schematic, but only after the second stage.
Hi Antonio,
I just tried right clicking and opening the images in another window and it worked - no idea why
I have a couple of questions and points about the mixer.
On the general schematic, the monitor signals are taken before the fader but in the second one they are taken after the fader. Am I right to assume these are meant to be three separate mixes to be sent back to the musicians (foldback)? in which case pre fader would be best.
How much mic gain do you need.? As it stands there is 42dB gain to the output of the tone stack.
The other thing you have to watch is the maximum input level which with close miked drums kits can be exceptionally high. It might be worth considering having a switchable 20dB pad at the mic input
You do not really need the the cathode follower part of the first Aikido. It is only driving the gain potentiometer and the SRPP first stage will do that. That will save you 10 tubes.
You also do not really need it to drive the tone stack, especially if you scale the resistance values up to reduce the load. That saves another 10 tubes.
However, you do need a proper low impedance source for the insert point after the EQ so I would recommend placing a cathode follower in there.
What signal level are you expecting at the insert point?
Cheers
Ian
I just tried right clicking and opening the images in another window and it worked - no idea why
I have a couple of questions and points about the mixer.
On the general schematic, the monitor signals are taken before the fader but in the second one they are taken after the fader. Am I right to assume these are meant to be three separate mixes to be sent back to the musicians (foldback)? in which case pre fader would be best.
How much mic gain do you need.? As it stands there is 42dB gain to the output of the tone stack.
The other thing you have to watch is the maximum input level which with close miked drums kits can be exceptionally high. It might be worth considering having a switchable 20dB pad at the mic input
You do not really need the the cathode follower part of the first Aikido. It is only driving the gain potentiometer and the SRPP first stage will do that. That will save you 10 tubes.
You also do not really need it to drive the tone stack, especially if you scale the resistance values up to reduce the load. That saves another 10 tubes.
However, you do need a proper low impedance source for the insert point after the EQ so I would recommend placing a cathode follower in there.
What signal level are you expecting at the insert point?
Cheers
Ian
http://www.bestsellers-store.com/audiospeaker/images/stories/Valve/tube tone.png
This is very simple. But the quality can be better with the other half of the tube
Sajti
There are problems with simple single stage negative feedback (NFB) tone controls made using tubes. The only problem you do not get is instability because it is only a single stage.
The first problem is open loop gain. In a single stage using something like a 12AX7, the best open loop gain you can hope for is 34dB (50 times). This is OK for reducing distortion at mid band where you are aiming for unity gain but when you have 20dB of boost you have 20dB less NFB and hence 10 times the distortion. Also, in tubes, distortion is level dependent so when you boost a signal its levels increase, which increases distortion and the NFB reduces which also increases distortion.
Another problem is that high mu tubes have a high anode resistance which means a single stage, even after NFB, has a relatively high output impedance and the low standing current means it has very little drive capability and really needs a following buffer stage. So you really need 2 triodes.
Another problem is that you cannot with an ease dc couple the NFB in tube designs the way you can in transistor designs. This means the gain rises to the open loop value at very low frequencies.
An alternative with 2 triodes is to use passive EQ followed by a low distortion gain make up block with good drive capability. You can choose tubes and topology for low distortion and good drive capability without the need to strive for huge open loop gains. You only get level related distortion and no NFB related distortion and you don't have to concern yourself about a rising low frequency open loop gain.
Cheers
Ian
Consider the “general schematic” as source of the mixer schematic.
By means of S4 it is possible to select the pre-tone and the post-tone (pre fader) type of “send”.
In the first case we have a signal of “send” 2,5V RMS for mic and 1.4V RMS for instr (it is possible to level these two signal by inserting a resistor after the first stage). Such a signal is good (e.i., does not need of further amplification) for any signal-elaborating device. Furthermore, this signal has not been influenced by tone control.
In the second case (after the tone control) the signal of “send” is 10 time less (and can be utilized only by elaborating-devices having an input preamplifier stage).
I have considered a mic with an output of 2 mv RMS (I think that includes most of the microphones).
I supposed that and inserted the 50K pot between first and second stage. Do you think the pot is not appropriate? If so, where to insert a switchable 20dB pad?
I considered the first Aikido stage as mic preamplifier, and the second Aikido stage for the instrument preamplifier. How could I eliminate one of them and having the same gain (or an appropriate gain)?
I have already calculated the new tone control, scaling up it of 5 (50K pots instead of 10K pots). The load is reduced but the loss remains 10 times. In my first tests, I utilized 1M pots … and the loss was the same.
How?
In my schematic, I should have a 140-250 mV (mic and instr) after the tone control. In this way, the further summing stage should have a gain of 8-10 to obtain an appropriate output signal.
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The single tube feedbacked tone control has some drawbacks of course. But the distortion, and the noise will be less, than the non feedback driver stage, handle 10times higher level signal to avoid the attenuation of the passive version.
You can easily get better parameters if You use the second half of the ECC83 as direct coupled cathode follower. This will results lower output impedance. And You can get higher gain if You apply bootsrapping from the cathode follower, back to the common cathode stage. With this solution You can get the open loop gain close to 100.
Sajti
You can easily get better parameters if You use the second half of the ECC83 as direct coupled cathode follower. This will results lower output impedance. And You can get higher gain if You apply bootsrapping from the cathode follower, back to the common cathode stage. With this solution You can get the open loop gain close to 100.
Sajti
Checking the HTML, they should be links, not image tags.
Mixer - general scheatic | Agorà
and
Mixer-tube | Agorà
OK, I understand. However, my question was about the monitor (headphone) sends rather than the insert (send/return). On the 'general schematic these (orange lines) are taken after the fader. Is this correct?
The gain of the transformer and the first Aikido stage is about 39dB. So, if the maximum output from the Aikido before serious distortion is 20V rms (+26dBV) then the highest input signal you can handle is 26 - 39 dBV =-13dBV or about 220mV. So the question becomes how likely is a microphone signal level of 220mV? The answer, with close miked drum kits and guitar cabs, is very likely so I would definitely recommend including a switched 20dB pad. The usual place for this is right next to the transformer primary.
The first tube in the Aikido provides all the gain. The second stage is a unity gain buffer/driver. For the fist Aikido stage you do not need the buffer stage (the right hand ECC88) so you can delete it. You can just take the output from the cathode of the top tube of the first stage. The same can be done with the second Aikido.
Yes the loss is the same. The reason for doing it is to reduce the load on the stage that drives the tone stack
Use the cathode follower part of an Aikido - the right hand tube circuit.
I think it would be better to make up the gain of the tone stack before mixing in order to avoid noise problems so I would suggest placing an SRPP (effectively the first stage of the Aikido) after the EQ to bring the level back up to a nominal 2V or so.
Cheers
Ian
In practice you can build a mu follower with a 6CG7 with 26dB gain and less distortion and no more noise than a 12AX7 with NFB.
Unfortunately the low output impedance of the ECC83 cathode follower has no real drive capability simply because the idle current is so low and it is single ended. Bootstrapping will raise the open loop gain to about 80 (and extra 4dB) but raises the cathode follower output impedance and distortion (because it is positive feedback). The increased cathode follower distortion is offset by the reduced distortion of the bootstrapped stage plus the additional NFB but the higher output impedance remains.
A 6CG7 mu follower with 5mA idle current will drive +26dBV into 10K at 0.4% THD and starts to go into grid current at about +34dBV. An ECC83 cathode follower clips at +17dBV into 10K.
Cheers
Ian
Why don't you start from level diagram?
Also, it would be a good idea to use microphone input transformer: you can get wider signal/noise ratio since tubes have much higher input resistance than output resistance of microphones. Second, you can have balanced inputs to eliminate interferences. Third, you can add phantom power for nice microphones.
Speaking of active VS passive tone controls, no big difference: in both cases you need make-up gain, and it is equal.
What I did in one PA mixer design, I split cut and boost between stages: I controlled cut only before faders, and had fixed boost only in mixing amps. Easy and cheap, and sounded great.
Also, it would be a good idea to use microphone input transformer: you can get wider signal/noise ratio since tubes have much higher input resistance than output resistance of microphones. Second, you can have balanced inputs to eliminate interferences. Third, you can add phantom power for nice microphones.
Speaking of active VS passive tone controls, no big difference: in both cases you need make-up gain, and it is equal.
What I did in one PA mixer design, I split cut and boost between stages: I controlled cut only before faders, and had fixed boost only in mixing amps. Easy and cheap, and sounded great.
And why You want to drive 10kohm with the cathode follower? The next stage can be 50kohm, or even more.
Another issue, that this follower is inside the feedback loop, with minimum 20dB feedback (or even with 40dB if the controls set to flat response). This feedback is reduce the distortion, and the noise close to the unmeasurable level.
And, that the driver for the passiv tone control has to drive with 20V for 2V output. The feedbacked version has to put out only 2V, which is 20dB less!!!!
Sajti
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This stage drives the insert which will be plugged into external FX and EQ units. These usually have a nominal 10K input impedance.
It will reduce distortion but unfortunately the distortion in a normal common cathode amplifier starts quite high, even when bootstrapped. This means that even after NFB the distortion is very similar to that of he 6CG7 mu follower. It is OK provided you don't need any serious drive capability. It is certainly measurable.
Negative feedback does NOT reduce noise.
I said the passive EQ really needs a gain make up amplifier after it - equal to the passive EQ loss. Thus the overall gain is unity just as for the NFB case. 2V in gives 2V out.
Cheers
Ian
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