Just how bad is it? Audio mixer with headphone amp.

[paulca]

I have learnt a lot about OpAmps, but I have a lot to learn still.

Be gentle, but just how bad is this first attempt at a schematic?

The idea I'm aiming for is individual line level pre-amps with a gain of 2, each attenuated by a pot before joining together and the output being buffered.  Effectively a 3 channel stereo mixer.

An additional buffer for the Aux output to prevent impedances affecting the power amp.

A parallel pair of op amps as a power (headphones) amplifier.

Thing's I'm not really sure about.
1.  I copied the power amp schematic.  Why are there 100Ks to ground on the + inputs?
2.  How do you choose the capacitor and resistor output values?  Is this to set the impedance seen from the next device along?
3.  Is this even remotely likely to work?
4.  Have I calculated my gain correctly?
5.  Do I need caps and resistors on the AuxOut?
6.  Same question for the inputs.


Note I know my connectors are not wired right yet.

I did attempt to test a few things in LTSpice but it keeps crashing on me today

[Audioguru]

Most of your opamps have no DC input bias voltage reference. Since you have a plus and minus 12V supply then 0V is usually the reference that must be applied with a resistor from 0V to the "+" input of the first opamps then they need an input coupling capacitor to block any DC from the signal source.

A mix opamp is usually inverting then adjusting one or more input levels does not affect the levels of the other inputs like your non-inverting mix circuit does. The input level controls in your circuit load down all the others so the output level from your mix opamps will be fairly low.

Why do you have input opamps? Feed the inputs into level control pots then they feed an inverting mix opamp that has a gain of 2.

You do not make a power amp by paralleling opamps. The 100k resistors to ground give the paralleled opamps their 0V DC input bias voltage reference that I was talking about.

A simple formula (one divided by pi x f x R) is used to calculate a coupling capacitor value.

Like most opamps, your output buffer opamps will probably oscillate when driving the capacitance of a shielded output cable. They need a 47 ohm to 100 ohms series output resistor to feed a cable.

[paulca]

Thanks Audioguru!

So it would just be the first op-amps that experience incoming signals that need to remove DC bias?  Once they have removed it the rest do not need 0V reference?

I was a bit suspicious of the mixdown with 3 stereo pots basically connected together.  At the same time I thought that putting them before the input opamps.  I figured that would be attenuate then amplify, which sounded backwards.  Though I note the headphone amp does just that.

How does an inverting amp not get pulled down with other resistance on the input?  Can you explain how that works?

I choose to have input opamps as an attempt to prevent different input impedance messing around with each other.  When I connect two audio sources to one output I invariably get one being very quiet while the other loud and things like that.  I thought putting a buffer (or preamp with a small gain) on each input would separate the impedances.  Besides every mixer circuit I looked at had input amps.

For the coupling caps, can I just carry on the trend of using 1uF and 47K's?

If I put 48KHz as the frequency and 47K as the resistor it gives me a 140pF. 
If I put 20Hz as the frequency and 47K as the resistor it gives me a 340nF. 
What do I do with that?  Aim high, middle, low?  Sample schematics online seem to just use 1uF.

[Audioguru]

I said that you do not need the input opamps. Feed the input signal to the grounded level control pots through series coupling capacitors then the sliders of the level control pots feed through series coupling capacitors through series resistors to the inverting input of a mix opamp that has its + input biased at Vdd/2. All opamps must have their + input biased by Vdd/2 or by the output of another opamp that has its + input biased at Vdd/2.

When a mix opamp is inverting then the inputs do not affect each other when you change the level  or impedance of them. Because the - input of an opamp that has negative feedback is a "virtual ground" with no measurable voltage signal. An input current through a series resistor to the - input simply causes the same current of the opposite polarity in the feedback resistor, not in the other input resistors. When the feedback resistor has current in it then the output voltage of the opamp changes according to Ohm's Law.

Most mixers have input opamps because they are microphone preamps.
1uF into a 47k load passes audio frequencies down to the earthquake frequency of 3.4Hz. Audio goes down to no lower than 20Hz so a 330nF film capacitor is fine for 10.3Hz since two coupling capacitor at 10.3Hz cause a cutoff of frequencies below 20.6Hz.

A coupling capacitor passes all frequencies above its calculated cutoff frequency. Then if it passes 20Hz it passes all audio frequencies and higher frequencies.
Why do you have an ultrasonic frequency of 48kHz that nobody can hear? Humans can hear no higher than 20kHz but when I was young I heard ultrasonic frequencies from burglar alarms that nobody else could hear. You need coupling capacitors to pass frequencies down to 20Hz then two RC couplings need to be calculated to pass 10Hz.

[hamster_nz]

It is only a minor point, but consider connecting the pots for the power amp to the outputs of the mix op-amps, the other side of the 47k resistors.

Otherwise changing the power amp volume settings will change the loading on "mix buffer" op-amps, reducing headphone volume (slightly) as power amp volume goes up.

With the power amp at min volume the headphone signal is between 47k and 50k, so about 50% of the mix amp output. With the power amp at max volume, it is between 47k and (100k in parallel with 50k, or 33k), so about 36% of the mix amp output.

You most likely wouldn't notice it in practice....

[paulca]

I said that you do not need the input opamps.

The other reason, of course is I want independent positive gain on the inputs.  If one input needs lifted above it's current level without input amps I would have to turn the others down and then raise the output volume.  That's just cheesy.

I have used 'real' mixing desks, line or mic they have input amps and gain on the input channels.

Why do you have an ultrasonic frequency of 48kHz that nobody can hear?

Sorry, was getting confused with the sample rate on the digital side 

You need coupling capacitors to pass frequencies down to 20Hz then two RC couplings need to be calculated to pass 10Hz.

The coupling caps on every amp?

Also, what are the two RCs for 10Hz for?

[BrianHG]

In keeping with your existing topology and op-amp count setup, to simultaneously get rid of the possible input DC bias and provide a visible termination load at the audio inputs, here is what I would change in your schematic:

Move and change the 1uf at the out of the op-amps to 10uf np to the audio input, then move the adjustment volume control right after that and feed the + input of each op-amp.  As for the outputs, just leave the 47k series resistors.  I would lower these to 10k just to lower impedance in the circuit.  This will not affect the op-amps ability to drive the audio output, but, lower the impedance to the next stage.

Now, there are other ways to mix all your inputs without cross-talk eliminating 4 op-amps, but, I would keep the input section as is for understanding and simplicity sake.

Next, remove the 47k series resistors at the output of your mix buffers.  You will also need to add gain on your mix buffers if you want the original sound levels of 1 single input source to 1 output.  I like to add a few db in my mixers, so I would make a gain of 2 over here too.

Once again, there are other ways to do this mix of all the source channels using inverted inputs, but, the way you set it up is very functional and good for basic understanding.

As for all the gain resistors, 100k is a little high.  I personally would use a lower impedance feedback, like 10k or even 1k.

[Audioguru]

Level controls are usually logarithmic so when they are set to halfway their output levels can be dropped to nothing or increased 10 times. Have the gain of the mix opamp set to 10 times or 20 times then set all the input level controls to halfway. Then you can turn one up or down a lot without affecting the others. No input opamps are needed.

The input to each level control needs an input coupling capacitor to block any DC that might come in from a signal source and it needs an output coupling capacitor to feed the input resistor of the mix inverting opamp. The Aux Out opamps can be removed because the output from the mix opamps do the same thing. The outputs of the Mix opamps need output coupling capacitors to block their DC. Then you have three RC lowpass filters made from the coupling capacitors.

[Audioguru]

You need low impedance feedback resistors for high frequency video circuits, not for audio.
Here is an inverting mixer that uses a single polarity supply:

[BrianHG]

Now, for the headphone amp.

I know that doubling the output current of a small op-amp may appear to be good to drive headphones, and, for cheap low volume and unpredictable current limiting points at higher volume with unpredictable headphone loads, ok.  There are op-amps better suited specifically for driving headphones, or, if you don't mind a little extra current on your power supply, I would use 1 of your same op-amp, driving a heavily biased 2N3904/2N3906 emitter follower configuration making a good strong output drive.

[BrianHG]

Question1:  Why are you swamping Left & Right between the pins of the stereo RCA input jacks and the RCA output jacks.

Question2: Why is pin 2 the GND on your headphone jack.  Most data sheets for these connectors has pin 1 as GND.

[Brumby]

Question1:  Why are you swamping Left & Right between the pins of the stereo RCA input jacks and the RCA output jacks.

Question2: Why is pin 2 the GND on your headphone jack.  Most data sheets for these connectors has pin 1 as GND.

Note I know my connectors are not wired right yet.

[paulca]

Question1:  Why are you swamping Left & Right between the pins of the stereo RCA input jacks and the RCA output jacks.

Question2: Why is pin 2 the GND on your headphone jack.  Most data sheets for these connectors has pin 1 as GND.

Note I know my connectors are not wired right yet.

The plan is to find, import or create a proper symbol for a stereo jack and possible an RCA Phono gang.

[paulca]

Now, for the headphone amp.

I know that doubling the output current of a small op-amp may appear to be good to drive headphones, and, for cheap low volume and unpredictable current limiting points at higher volume with unpredictable headphone loads, ok.  There are op-amps better suited specifically for driving headphones, or, if you don't mind a little extra current on your power supply, I would use 1 of your same op-amp, driving a heavily biased 2N3904/2N3906 emitter follower configuration making a good strong output drive.

I know I could use a push pull set of mosfets. 

Here's the thing.  I have that headphone amp already built, it came as a kit and it drives my headphones perfectly.  The only minor annoyance is that if the signal source is low volume, a quiet track or quiet YouTube video I just don't have the gain.  This lead to the idea of adding an adjustable pre-amp to the input.

I know AudioGuru is trying to steer me away from it, but I'm still getting dragged back to pro audio mixers and practices. 

1.  Set your input gain on the channel for 0db normalisation.
2.  Set your filters, EQ, balance (if these are after gain, repeat 1)
3.  Repeat for all your input channels.
4.  Set your send levels.
5.  Set your return levels.
6.  Set your mix levels.
7.  Set your master level.

To do that right, and it is tempting, but possibly OTT, I would have a 0db SIG and 12db CLIP LEDs on the inputs and have pots before and after the input pre-amps.  The LEDs would be after the pre-amp gain.  So I can normalise the channels to approximately 0db (relative to a reference I assume).

I keep hearing "Gain structure" when we consider just putting unnormalised signals into a mix amp, boosting it and then attenuating it down.  It just feels wrong.

I know it will "work", I'm not arguing that you are wrong, but it still feels wrong.  I don't mind making it over complicated either, a lot of this project is about learning stuff.  I could just go out and buy a desktop amp block with 3 inputs and 2 outputs and be done.  Very few will have independent input gain though.

Ill try and update the schematic tonight or tomorrow if I get time.

Thanks again, everyone for helping   I'm learning.

[IanMacdonald]

As mentioned a gain of 2 (6dB) will be only just noticeable on audio, but not much use for boosting low signals. 20dB (10x gain) is probably a more sensible figure to aim for.

The way this works is that 10dB is ten times the sound power, and an apparent doubling of loudness. However, since power is proportional to the square of voltage, that's just over 3x the signal voltage.

The minimum loudness change the human ear can reliably detect is 3dB, which is roughly a doubling of sound power.  Sounds surprising, but it's true.

Doubling the voltage thus gives about twice the minimum change the human ear can detect. it's still not much though.

2x: 6dB (approx)
10X: 20dB (exact)
100x: 40dB
1000x: 60dB

A mic input needs about 60dB gain to cope with relatively quiet talkers. Though I've heard some opera singers claim that they can produce a volt from a microhone.  (Put your glassware away whilst testing this)

[paulca]

Thanks Ian.  So a little more gain on the input amps.

I was searching around and getting annoyed with the usual "All about circuits" type random audio circuits and found this:

https://carvinaudio.com/pages/mixer-schematics

And a single mono input channel.

http://carvinimages.com/schematics/DX%20Series%20Channel.pdf

Seems to be the actual schematic for a single input channel on a mixing desk.  It's a bit hard to calculate the gain as they are doing funky stuff with the XLR input and the Line, though the Line passes through a voltage divider and the XLR doesn't R9/R10.

A lot of it is beyond me, but I can identify stages and components.... mostly because they are labelled LOL.

Another interesting thing is that the variable gains on the filters at least seem to break a supposed rule of not using feedback as gain.  I remember in High school the first opamp circuit I seen had a pot on the feedback loop.  It has always seemed a bit odd to me that people either boost then attenuate or vice versa, instead of just altering the gain of the amp directly.  I'm sure there are reasons, but this channel circuit uses variably amp gain.

I also count a whole 4 dual op amps in the input stage for mono.  To replicate that for a stereo channel that would be 8 dual op amps and they haven't even mixed it down yet.  Granted most of the opamps seem to be to do with active gain filters, LOW, HIGH and a sweep-able MID (not sure f it's fully parametric with Q or just sweeped).

I don't need anything this complicated.  It would be nice to have Low,Mid,High but maybe next time.   I just thought I'd add it to see what real mixing schematics look like. 

[dmills]

The auxout buffer should have 47 ohm or so resistors between the opamp output and the socket to reduce the chance of oscillation with a capacitive load.

100k is a bit on the large side for the 5532 (Which is a bipolar opamp with relatively low noise impedance), 10k or even 2k2 would be more like it, also lower value resistors are generally quieter due to lower intrinsic thermal noise.

If I was doing this, I would dump the input amps and do the gain in the summing amplifiers at the moment you have a per input gain to the output of Av = 2/3 so you actually attenuate.
Make the summing stage inverting and give it some gain, make the pots maybe 5 or 10k (To give yourself space to have the summing input resistors be large enough not to disturb the taper too much) and invert a second time at the output buffers.


Where is the decoupling?

State variable filters (Which are what that mixer input strip uses) consume about 4 opamps per band, so swept mids are expensive on parts count.

Regards, Dan.

[BrianHG]

Don't go too crazy on the gain.  You'll begin to add hiss.  Also, having gain on you mixer stage means the primary input opamps wont have to go all the way to your rails to provide the same 2vrms output since their output is divided by 3 in the mixing.  Unless you go to Audioguru's negative scheme at the mix stage (when done right, it's used quite often in mixers), stick with my recommendation of having a gain of 2, or 4 if you want more at the input amps and the same gain again at your mixing middle amps (unless you choose 3x deliberately to compensate exactly for the mix) & don't forget to shrink those overly large sized 100k feedback & GND gain resistors.

Also remember, the higher you make the gain on your op-amp, the more lousy the high frequency response.  Another good reason to divide the gain between the input amps and you mid mix amp.

[paulca]

If I was doing this, I would dump the input amps and do the gain in the summing amplifiers at the moment you have a per input gain to the output of Av = 2/3 so you actually attenuate.

Av=2/3... how?  A 1:2 divider on the feedback side should equal a gain of 2.

Anyway, as stated, while it may be more complicated than it needs to be, I feel I want normalised inputs.  I have played the gain structure game before and hated it.  To turn one channel up you have to turn the output gain up and then turn down the other channels.

It's all very well when you connect 2 or 3 channels of equal or nearly equal output together, but when you get one channel that is 1/10th the power of the others it gets really irritating pulling all the other channels down to that level and then amplifying all channels along with all channels noise.

As a random example, I was watching a YouTube video on OpAmps and the presenter was quite interesting, but he was also mumbling and the level on the video was barely audible.  I had YouTube at max, PCM at max, master at max and the headphone amp at max and if I had decided to play music at the same time I would have had to wind the music input right back and there still wouldn't have been enough gain on the output amp, if there was it would mostly be amplifying the noise.

Granted this is a for a desktop headphone amp to combine all my sources and outputs into one box, so, no I probably don't 'need' input pre-amps, but I have my mind set.

[BrianHG]

If I was doing this, I would dump the input amps and do the gain in the summing amplifiers at the moment you have a per input gain to the output of Av = 2/3 so you actually attenuate.

Av=2/3... how?  A 1:2 divider on the feedback side should equal a gain of 2.

Anyway, as stated, while it may be more complicated than it needs to be, I feel I want normalised inputs.  I have played the gain structure game before and hated it.  To turn one channel up you have to turn the output gain up and then turn down the other channels.

dmills may be speaking as after you make my recommended changes a few posts up here: https://www.eevblog.com/forum/beginners/just-how-bad-is-it-audio-mixer-with-headphone-amp/msg1424158/#msg1424158
Which solves this problem, as well as a bunch of others...

Same goes for all my subsequent posts, I expect you have updated you schematic to my recommendation.

[paulca]

Don't go too crazy on the gain.  You'll begin to add hiss.  Also, having gain on you mixer stage means the primary input opamps wont have to go all the way to your rails to provide the same 2vrms output since their output is divided by 3 in the mixing.  Unless you go to Audioguru's negative scheme at the mix stage (when done right, it's used quite often in mixers), stick with my recommendation of having a gain of 2, or 4 if you want more at the input amps and the same gain again at your mixing middle amps (unless you choose 3x deliberately to compensate exactly for the mix) & don't forget to shrink those overly large sized 100k feedback & GND gain resistors.

Also remember, the higher you make the gain on your op-amp, the more lousy the high frequency response.  Another good reason to divide the gain between the input amps and you mid mix amp.

Thanks.  The plan is to maybe up the gain on the input amps a little.  I am still tempted to use feedback gain control there.

Hiss and over-gaining is why I want to avoid this pattern being suggested of attenuating the inputs then boosting them up at mix amp only to attenuate them back down again at the headphone amp.  I want a small amount of normalisation gain on the inputs, then use AudioGuru's inverting mixing amp with unity gain. 

In normal operating conditions with good inputs the input amps should be running in unity.  Another reason I'm considering adding variable gain feedback instead of amplify then attenuate.

Finally enough gain on the output amps to make my headphones clip....  I'll add a third 5532 is I need to   Or a quad of mosfets if I have to.

Ideally I would like the input level pots to be unity gain at centre. x2 or x3 fully clockwise and 0 fully anti-clockwise.  That might be tricky though but a log pot of the right size might work.  I have to review these gain figures in light of the DB gain calculations some placed above.

The mixing circuits seem to all have variable gain on the inputs and then mostly attenuate across the whole desk (except filters which allow positive and negative gain).  Having your channel faders about 0db is a mix smell.  Master faders usually don't even go about 0db.

[paulca]

Same goes for all my subsequent posts, I expect you have updated you schematic to my recommendation.

I'll try and make the updates tonight to clear up confusion.

What are the pros-cons of variable feedback?  I suppose finding the correct pot/resistors might get expensive?

[BrianHG]

Do not use variable feedback.  The opamp feedback and gain resistor on the negative input is the most sensitive part of your circuit.  It responds to any RFI with ease.  Only use a fixed gain with 2 resistors as close as possible between the resistor and the - input of the opamp.

My directions were to place your series 10uf caps and volumes right at the RCA audio inputs.  This simultaneously protects your circuit from an audio source with a DC offset and makes a constant load on the RCA inputs matching the full series resistance, use 50kohm as this is standard, of the volume control.  The wiper goes to the + input of the opamp.  The opamp should have a fixed gain of 2 through 4 depending how much boost you want.

The opamp outputs of each channel should each have a series resistor feeding the center mix opamp which you can set a fixed gain of 3x to normalize the audio.

Doing things Audioguru's way means you need to invert the audio at each stage, and you still need now to further strategically place the mix resistors at the middle mixer stage right at that op-amp's - input to prevent noise.

[Audioguru]

The Carvin mixer uses three opamps at the input to make an instrumentation amplifier instead of buying an IC that does it (they were not available years ago). The input of an instrumentation amplifier is balanced (microphones are balanced) and for lots of gain for a microphone. Your inputs are unbualanced line levels so you do not need the three input opamps for easch of your inputs. The Carvin circuit does have an input gain control that you need for each of your inputs.

The Carvin circuit has normal bass, mid and treble tone controls but does not have the high frequency cut filter that is in your circuit. All three tone controls and the Fader use variable feedback to adjust the gain.

You said that you have a headphones amplifier that you made from a kit but it does not have enough gain. It is simple to replace the feedback resistor with a higher value resistor for more gain.

EDIT: Here is an instrumentation amplifier:

[paulca]

You said that you have a headphones amplifier that you made from a kit but it does not have enough gain. It is simple to replace the feedback resistor with a higher value resistor for more gain.

Yep.  I am borrowing the schematic for it into my design.  I believe the gain calculates as 3.13 but I could be wrong.  Opamps always surprise me.  It is the lower right corner of the schematic.

I can adjust the feedback resistor, but I need to check that I don't bump into the current limit of the parallel pair.  Of course, using 3 is an option or drop the parallel and use an output driver circuit.  I have to bear in mind that my headphones are fairly high impedance, around 34Ohms I believe, but if I connect a lower impedance set to the amps in the future I might bump into the current limiter of the 5532.

On the gain, it currently only has one input and with a good quality input I have enough gain, just about, it won't clip the headphones, but it comes close, distortion becomes present at full level, but barely noticable.  It is when there is a quiet source I don't have the gain.  In my experience in audio, yet not electronics, the place to add this gain is at the input, not the output and that becomes more important when there are multiple inputs.

EDIT:  Also if you note in the Carvin schematic both XLR and Line are amplified at the input.  Mixing desks have a "Gain" knob at the top of each channel.  It has 0db at the 12 o'clock position.  Anti CW attenuates, CW amplifies.  This is before the filters, channel fader etc.  They often have a +48V phantom voltage switch and a line level db selector..

[paulca]

Okay, so backing things down to being a bit more simple.

A single channel input pre-amp.
As per the annotation:  Gain of 0 - 3.128



I did consider messing around with a series resistor to try and get 1.000 gain in the middle of the pot, but I think it's a fools game.

When I can get this right I will move add one mono channels to the mix down amp and see if I can get that right.

I do like the idea of the inverting summing amp for mix down.  I like that the inputs will not interfere with each other, although now I have the volume pot before the pre-amp I'm not sure it matters.

I'll review the other comments before attempting the mix down, as I think my Tesco delivery has arrived

[Bassman59]

Take a look at the schematic for the Rane MLM82S. Everything you want to implement, except for the headphone amp, is there.

One could argue that you should use THAT 124x differential receivers for the line-level inputs instead of the op-amps, and INA163 for mic-level inputs.

For the headphone amp, the OPA551 has enough drive to clean the wax outta your ears.

[Audioguru]

Your idea of using an opamp for each input with a useless gain of only 3.1 makes no sense. You can barely hear a level change of only 3.1 times.
The input level control must be logarithmic, not linear so when it is turned down to half the input level is 1/10th and when it is turned to maximum the input level is maximum.
Since the input level control set to halfway cuts the level to 1/10th then you need gain of 10 following it. The gain of 10 can be in the mix opamp then the input level control can feed directly into the mix opamp through a series resistor that the inverting mixer needs.

Carvin and Rane use 3 opamps or an instrumentation amplifier because they have balanced mic inputs that you don't have. Rane even has balanced line inputs using a differential opamp but you do not have balanced line inputs. Then you do not need input opamps.

The 32 ohms headphones need about 2V RMS to produce 0.125W to be loud. The old NE5532 (about 41 years old) is spec'd to drive 10V RMS into 600 ohms which has a peak current of 23.6mA. With a peak current of 23.6mA the voltage in a 32 ohm earphone is 0.53V RMS producing 88mW which is loud but maybe not loud enough. Your circuit uses two NE5532 opamps each with a 47 ohm output resistor so the output power into a 32 ohm earphone is enough when  the power supply is 6V or more.

   

[BrianHG]

Your idea of using an opamp for each input with a useless gain of only 3.1 makes no sense. You can barely hear a level change of only 3.1 times.
The input level control must be logarithmic, not linear so when it is turned down to half the input level is 1/10th and when it is turned to maximum the input level is maximum.
Since the input level control set to halfway cuts the level to 1/10th then you need gain of 10 following it. The gain of 10 can be in the mix opamp then the input level control can feed directly into the mix opamp through a series resistor that the inverting mixer needs.

Carvin and Rane use 3 opamps or an instrumentation amplifier because they have balanced mic inputs that you don't have. Rane even has balanced line inputs using a differential opamp but you do not have balanced line inputs. Then you do not need input opamps.

The 32 ohms headphones need about 2V RMS to produce 0.125W to be loud. The old NE5532 (about 41 years old) is spec'd to drive 10V RMS into 600 ohms which has a peak current of 23.6mA. With a peak current of 23.6mA the voltage in a 32 ohm earphone is 0.53V RMS producing 88mW which is loud but maybe not loud enough. Your circuit uses two NE5532 opamps each with a 47 ohm output resistor so the output power into a 32 ohm earphone is enough when  the power supply is 6V or more.

 

He is going from line level to line level at the next stage.  He is not amplifying a microphone, he is not looking for serious gain, nor does he want any of his stages to clip by turning up to source volume to the max.  If he wants minimal hiss and the best possible frequency response of the opamps, the minimal gain of 3 at that stage and another gain of 4, approx 10db at the mid stage isn't too terrible a choice.  Maybe to get a gain of 6db after the 3 source mix, he can choose a gain of 6, but going to a gain of 10, with a maximum line level source of +/-1.25v * 10, he hits his +/-12 supply rail, though, he can go to a +/-15v supply.  Placing some gain (maybe targeting to 6 instead of 3)  at the input stage, next the 1/3rd level loss due to the 3 way mixing, then adding some gain in that middle opamp as an additional boost, say x2 or x4, is a strategy where he can use with his 3 way all positive audio design mixer, generating some additional signal gain without any clipping risk anywhere.

Remember, if he was doing a negative mix in the middle feeding the -input, he would not need that extra voltage swing at the first opamp stage outputs.  But now, those opamps will need to invert the signal as well to make positive audio in the end.

[Audioguru]

The original circuit has no overall gain, it has loss.
1) Input opamp gain= 2.
2) Passive mixer gain= no gain, the output is 0.33 times because a 47k mix resistor is loaded with two 47k resistors in parallel from the other two inputs.
3) Mix buffer, no gain but its output is divided to be 0.52 times.
4) Output buffer= no gain.
Then the output level is 2 x 0.33 x 0.52= 0.34 times with an input level control at maximum.

My idea has the mix opamp a true inverting active mixer with a gain of 10. The mix opamp feeds directly to the output with no loss and it also feeds the headphones amplifier.
The logarithmic input levels at halfway produce the same output level as an input level and the level control can boost a low level up to 10 times or reduce it to zero.

If true phase is important then an inverting opamp can be added to each side with no change of gain.

[paulca]

The original circuit was wrong.  It was a best effort with a few bits of critical information missing from my concept.  I'm learning.

The Carvin and Rane as well as the mixers I have used have gain on the inputs to normalize them to the same level before mixing them.  The only exception was a £35 DJ mixer which assumed identical turntable decks.  Even that one (actually a 3 channel with 1 line input) was fairly useless as the line input was too low and needed the mix fader planted to MAX and the turntables turned down and then master raised to get a decent mix.  Bad gain structure = noise. 

If you search on ebay for "Audio mixer" all but the cheapest and dodgy-ist have pre-fader input gain on all channels.  Including line only channels.

I could go for more gain at the input, a mixing desk often has -10db / +60db and will clip.  The channel faders often have gain too, into the mix, though less than the input gain.  The master rarely has gain and usually 0db is the max stop.

I know your way will work AudioGuru, but I would like to aim for sending already gained or attenuated channels into the mix amp.  To be more honest to the mixing desk idea I should have another pot after the input amps.  But it's overkill for the purpose of a desktop audio box.  Though I expect you believe the input gain/attenuation is overkill too.

I accept your point on using log pots.  I also intend to not get caught into audiophoolery but I hope to buy branded 1% resistors and decent caps and log pots.  This is why I'm also trying to limit the number of resistor and cap values.

I'm still unsure of whether I should go with the inverting summing amp, post input opamp or not.

Anyway, I'm going to try and clear a breadboard tonight and start testing these things with the scope and some waveforms.  I only have a cheap signal generator kit thing and a soundcard, but it will do.

The only hurdle is not having a +-12V or +-15V supply.  I'm wondering if I can series my two RD Tech PSUs.  Can't find anything with google, but they pass they are definately not earth referenced as one is running off +- DC and the other off a battery.

[BrianHG]

For an op-amp based mixer, you need nothing more than a wall-wart 15v/12v AC out adapter.  2 1N400x diodes, an optional series 1w resistor to soften the rectification current peaks, 6 caps and a 7815/12 and 7915/12 regulators for your power supply.  This project wont suck more than 200ma total.  The AC out wall wart guarantees transformer isolation from the mains.  No stinking Y capacitor means no looping ground hum and the low power means you pretty much will survive an accidental short-circuit.  That is unless for some reason you use an above 500ma wall-wart.  I wouldn't go above 1 amp.

Makes the device safe to the touch as well, with approved wall-wart of course.
Stay away from switching supplies here.  You just don't need em.

[paulca]

I know I can do that and probably will for the breadboard.  I still think I can series my PSUs and take the mid point as ground to get true +/-15V.

I want to avoid a virtual ground in the real thing.  It tends to cause issues when connecting to other equipment, especially hot plugging.  Suddenly you are raising a piece of equipment ground by 6V, they tend to get upset, especially 5V devices which I found brown out and crash.  Not a bit deal for headphones though, but connecting it to an Arduino to display an LED level meter... brown out crash.  A bigger coupling cap would raise the ground slower, but maybe not slow enough. 

I also crashed the USB Host in the PC doing similar.  It did not like it's ground being ramped up to 6V at all.  "dmesg" output showed the OS considered the host as "dying" and disabled it.  I had to reboot to get it back after 20 minutes of trying to send the reset code to the device /sys

[BrianHG]

I want to avoid a virtual ground in the real thing.

[paulca]

Oh, hang on, I think I follow you now.  You mean to open the AC/DC wall wart and tap the transformer to get 15VAC then filter it to get +/- 15V DC?  EDIT: or more likely +- 6V or 9V or something similar after caps filters the half sines.

I actually bought a 12V wall wart off RS destined for that experiment exactly.

[Audioguru]

When you have too many level adjust pots then one or two of them always end up in the wrong position.

Everything you said about gain and noise applies to a microphone mixer that you are not making. A line level mixer does not need gain here, there and everywhere and makes inaudible noise.

[paulca]

When you have too many level adjust pots then one or two of them always end up in the wrong position.

Everything you said about gain and noise applies to a microphone mixer that you are not making. A line level mixer does not need gain here, there and everywhere and makes inaudible noise.

So every audio mixer is wrong.  Every sound engineer friend I have is wrong. 

Set the master fader to the bottom of the sweet spot region (e.g. -5 dB).
Set the channel faders for each input in the sweet spot, usually at the 0 dB point.
Unmute the inputs and dial up a rough mix using the preamp input gains as if they were faders. Let the preamp gains fall where they may.
Now, forget about the preamp gain knobs and focus on mixing the show using the faders. All the faders are now near zero dB for maximum mixing finesse.

http://www.fohonline.com/current-issue/74-tech-feature/10445-back-to-basics-gain-structure.html

I can find hundreds of references if you want.  Gain is added at the input.  Always is.

With your approach, if one input changes, all inputs need to change to balance the output.  This might be fine in my case where I might have one or two inputs active at a time, but if you have a 24 channel desk and one line feed from a keyboard is low it would be completely ludicrous to alter the mix of the other 23 channels to rebalance the mix.

[paulca]

The other consideration is noise.

Every gain stage amplifies noise.  If you attenuate a signal you do not lower the inherent SNR of the equipment.  Attenuate it enough you will lower it into the noise.  When you then amplify that signal you amplify the noise as well, therefore you have just cut your SNR dramatically.

So the approach of attenuate then amplify is backwards and ruins you signal to noise ratio.  You want to lift your signals away from the noise ONCE then attenuate them to the desired level.  Running a desk with the faders above 0db suggests a bad mix, for that reason.

This master mix output is not gained again until it goes to the power amps or the recording equipment.   In a recording studio the line and mic inputs will be gained only once.  (excluding guitar pre-amps, amp heads, active mics, electronic instrument line level output amps).

[Audioguru]

With your approach, if one input changes, all inputs need to change to balance the output.  This might be fine in my case where I might have one or two inputs active at a time, but if you have a 24 channel desk and one line feed from a keyboard is low it would be completely ludicrous to alter the mix of the other 23 channels to rebalance the mix.

Nope. When the mix opamp has a gain of 10 times then most input level controls are set to halfway on their logarithmic level controls. If one input is low then simply turn up its input level control without touching the other input level controls.
The boost of up to 10 times is 20dB which is plenty.

Audio opamps are low noise and are used with a gain of 200 times or more for microphones. With a gain of only 10 times in the mix opamp then its output noise level is 20 times less (-26dB). Inaudible.

[dmills]

Dangerous Audioguru...

In a virtual earth summing stage the noise gain is proportional to the number of inputs, so while not a problem with a small number of inputs the summing amp on a large desk can very easily become  a noise problem, to the point that actually breaking the desk down into a tree of summing stages can actually result in lower total system noise.

I would note that with even halfway respectable opamps (Which the 5532 easily are), operating in a suitable impedance environment, the noise from the RESISTORS usually exceeds the noise from the opamps.

I highly recommend that the OP obtain a copy of "Small signal audio design" by Douglas Self for more then they ever wanted to know about how to design audio doings.

Regards, Dan.

[Audioguru]

This simple mixer has only 3 inputs, not 24.

[BrianHG]

I think that paulca's design, though not necessarily some of our chosen topology/techniques for a 3 channel line level mixer, with some of the changes already made above, like the volume at the RCA inputs & most of everything else remaining the same will function excellently for his intended use.  The ne5532, even with some gain, will have such a quiet noise floor that he wont hear a thing so long as the PCB is well laid out.

As for the power supply, with an AC out wallwart.  Just use power wire A direct to GND, then the other goes through the optional resistor in series, to the 2 diodes (both tied at that point, just 1 forward, 1 backward).  Add your power filter caps there, I like something like 10000uf 25v each, or higher. (This is already overkill, but I like my audio filter caps huge and you are doing half phase rectification if you power supply is done like this...)  For a 12v AC adapter, you should get around +16v on the forward diode and -16v on the reverse diode.  Put those 2 voltages through your 7812 and 7912 and you will have a clean +/- 12v.

[paulca]

Thanks.  I don't have a negative regulator to hand

As per my other thread I have two of my bench supplies in series, but I'm too scared to lift them above 1V.  I don't what they to go bang.

One is powered off a battery, the other off a LED Lighting DC supply.

[paulca]

On the amp design.  I am waiting bread boarding a few things before getting back into the design.

I hope to start with one input amp.  Test it with a scope.  Then try a second.  Then try a mix amp.

I'm sure at that stage I will have messed something up!

[BrianHG]

if you have 1w or 5w approx 12v zener diodes, you can make a second R/C filter stage at your power input caps and parallel the zeners in parallel to limit voltage to 12v on each cap.  That will suffice.

[paulca]

So tonight's mission was successful.

The input amp worked (mono).  Gain looked good.  Response, for as far as I could tell was grand on sine and "fair" on square.  I have waveform caps I can share later.  I tested at 1Khz, 20Khz and for some fun all the way up to 1.2Mhz.  Things go ropey above about 600Khz, but it wasn't that bad.

Square wave at 20Khz has quite a slow rise/fall time and a little overshoot, but I don't think it's that big an issue.

It was also successful in that the two RD Tech power supplies did a grand job of providing +12 0 -12V rails.

Also successful in revealing just how important decoupling caps on the power rails is.  I had huge noise ripples all over the place without them.

Also successful in confirming the cheap chinese kit build waveform generator is crap.  After decoupling the noise off the rails, I was depressed at the noise on the output until I hit "HOLD" a few times on the wave and confirmed 100% the noise was on the input and not being added by the amp.  Little oscillation pulses at a much higher frequency flowing long the waveform.  They existed on both input wave and output wave, so I ignored them.  (Signal gen was running on a completely separate battery, so clean power, though it did share the ground from the PSUs).

Finally, "successful" in finding out my laptop power supply has no damn discharge what-you-ma-call-ems on the main caps.  Bastard thing zapped me after I unplugged it and it hurt.  I do make an effort not to touch the pins when I pull a plug as I have been bitten before, but it got me.  Grrr.  Nothing like a little 120/240V zap to piss you off.  It has a proper UK plug with part insulation, so there is no way to touch the pins while they are connected to the mains, so it was definately the caps on the mains side discharging into my finger

[paulca]

if you have 1w or 5w approx 12v zener diodes, you can make a second R/C filter stage at your power input caps and parallel the zeners in parallel to limit voltage to 12v on each cap.  That will suffice.

Not sure I follow, but... oh, voltage regulation.  It's fine the bench supplies stepped up to the task.

My finger hurts  and it twitches still

[paulca]

So I had another session with the prototype.  Learnt a few more things.

My PC is foul for noise.  There is noise between 300mV and 1V peak to peak up around 40KHz though the scope was having trouble with it.  It's most likely the switch mode power supply or some other EMI generated in there.  I spend about 30 minutes ruling things out until I had to insight to connect my phone to the input.  Wham, no noise.

The PC noise was so intense it was impossible to determine what the amps where actually doing.

That sorted out, using the mobile phone as a source, with 2 of the preamp circuits above and an inverting sum amp it was working.  I was able to mix two channel together.

The gain structure is a bit off though.  What is the natural gain of a inverting amp?  I looked up the basics and it said if R1 = ... = Rn = Rf I would have a gain of 1.  But while I had a gain of 3 on the preamp the gain after the inverting amp was only about 30%.

I also had PSU noise to deal with, an oscillating ripple pulse appearing around 500Hz with the oscillation around 20Khz.  The waveform looked like a bass drum does, if that makes sense.  I expect it's the PSU's output mosfet ringing. 

Filters...

On the HF PC input noise.  I know it won't bother me as it's way outside of human hearing range, but is it worth trying to put a low pass filter in to kill anything over about 22Khz?

On the PSU noise, I had two switch mode bench supplies in series, probably not the best clean audio supply, but even with a 100uF and 100nF across both rails AND 100uF and 100nF across each rail to ground, I could not get rid of it.  Assuming the PSU I get for the project (probably an isolated encapulated PSU) has similar noise, how do I kill it? 

[Audioguru]

Since your pc generates so much noise then it might cause the opamps to amplify it and produce clipping distortion of the audio. Filter out the ultrasonic noise.
The gain of an inverting opamp is Rf/Ri. If the two resistors have the same value then the gain is exactly 1 regardless of how many input resistors there are. The opamp must be able to drive the value of Rf and most opamps have trouble driving less than 2k ohms including the output load. A non-inverting opamp is a passive mixer where the inputs affect each other.

Why aren't you using logarithmic level controls? If a linear control is turned down from max to halfway then the signal level drops only a little, then drops more and more (like an on-off switch) as it is turned down to zero. A log pot drops the level properly and evenly.

[paulca]

Thanks.

I haven't decided on pots yet.  I expect I will end up going for some 1% Alps log pots from RS.  They may not play on the breadboard.

At the moment I am using stuff out of my component stock and as I only have 1 50K pot, I was using a 100K 10 turn trimmer on the second input channel.  It's a bit rough and ready.

Also it's on a breadboard... my feedback resistors are like noise antennas.  I can get all manor of pretty scope pictures by touching them.

That's why I'm not being too fussy.  I just want to prove a few concepts to myself before I consider actually building it properly on possibly strip board.

I expect I need a bit more practice and study on PCB layout to make the end device I want to build.  So I'm spending time doing digital circuits on PCB to learn before I move to an analogue board and all it's sensitivities.

[paulca]

And... on the inverting amp.

I put 47R resistors on all the outputs.  So to aim for a gain of 1 of the inverting mix amp I put a 47R on the feedback.  I gather from what you said this probably isn't going to work well.

Do I need resistors between the amps at all?  If I put the OUT of an opamp directly into the IN- of the inverting amp and hard link it to unity will it still function as a unity gain summer or will the virtual ground on the - input kill the upstream opamps output?

[Audioguru]

If the output of the opamp must have a maximum swing of 7V peak then the output current in a 47 ohm feedback resistor needs 7V/47 ohms= 149mA peak! But most opamps can drive 2000 ohms minimum for a current of 3.5mA including the current in the load. Why the extremely low value of 47 ohms? Why not 47000 ohms instead?

Please learn what an inverting opamp is. It has a series resistor to its - input and another resistor fronm its - input to its output. Its voltage gin is the ratio of the two resistors.If one opamp drives the - input of a second opamp that has a negative feedback resistor then the output of the first opamp is almost shorted and the second opamp has a voltage gain of a few hudred thousand so it will be clipping the signal like crazy. Each input needs its own series input resistor whether it comes from a signal source or from the output of another opamp. Yes, the - input of an inverting opamp is a virtual ground, the series input resistor is very important.

[paulca]

I found myself a copy of Small Signal Audio Design.

Interestingly, Chapter 1, page 6, Gain Structures, describes exactly what I am aiming to do.

1. Don’t amplify then attenuate.
2. Don’t attenuate then amplify.
3. The signal should be raised to the nominal internal level as soon as possible to minimize
contamination with circuit noise.

It then goes on to suggest that 1 and 2 are compromises when you have an attenuating gain control, you have to do one or the other.  1 risks clipping, 2 risks signal to noise loses.

As my input stages will not be getting anywhere close to the amps VPP peak of 20V (a gain of about 30-40) I have plenty of headroom to lift it considerably, gain of 10 or 20.  The idea is to lift the level once (rule 3) and then only buffer it or attenuate it for output level.  Not this attenuate, gain, attenuate, gain series stuff, that just adds noise.

It also mentions that active gain controls should be used ideally, although I have not read the chapter on the pros and cons of their use/design yet.  I'm sure there are dragons there.

[paulca]

Well, had another session with this.  Decided to try something different and implemented one of these:

Source: Small Signal Audio Design

The positives are that it's the most stable outputs I have seen on the scope.  Plenty of gain, although when running my headphones the gain fell to 1, which has me a little perplexed.

The downsides are it's an opamp killer on a breadboard.  Two 5532s died.  The first one was most likely because I forgot the GND reference on the second stage amp.  That chip now does nothing but howl with a 13.5V DC offset.

The second chip died when I had a lose connection on the breadboard.  I was (probably foolishly) running the output into my headphones (34 Ohm IIRC), so a total output load of only 134 Ohms.  I had fissling and was poking various connections to find it then WWOOOOOOOOOO!  The opamp broke into full song howling.

If I swap in another 5532 directly everything comes back to working, I conclude both chips are dead.  They seem to work in isolation as voltage followers, but as soon as they go back into this circuit the each behave incorrectly.

I'm not sure what it is about the circuit that makes it easy to kill opamps, maybe the low value resistors and 5532s are not exactly expensive, but I'd rather not have to order more. 

I was using my new PSU an XP Power +-15V 200mA.  It seemed to be pulling around 40mA idle, 66mA running the circuit which was a single 5532 chip + an LED with a 2.2K limiter across the 30 volt rails.  This does not sound like the efficiency in the datasheet of 84%.

Anyway.  I'm a bit concerned about proceeding to create two inputs and a summer from this if I have other issues.

I'll retest the PSU in isolation to make sure it is functioning and I haven't wired something wrong.  40mA to power an LED which should be drawing around 12mA.

The small signal audio design book claims the resistors are small value to keep down the noise, if I take them up a factor of 10 would that should make the circuit safer against breadboard glitches?

[paulca]

Actually checking the datasheet for the XP PSU, it states 25mA idle current.  I believe the 40mA I was measuring included the LED, so 25+12mA = 37mA and not far off the measured 40mA.

That leaves the rest of the circuit pulling 26mA (20mA account from the PSU efficiency).  The 5532 has a max input current of 10mA, but it's hard for me to work out how much is dropping on the various resistors and not going into the opamp.

[Audioguru]

Why does a volume control need two opamps??

The high stray capacitance of your solderless breadboard probably causes the 10MHz opamps to oscillate and draw extra current.
The linear volume control is an on-off switch that does not match the logarithmic sensitivity of our hearing to loudness.

[paulca]

I believe the Baxandall configuration causes the volume control to be an approximation of logarithmic with a steep drop at lower settings.  It forms a nearly straight line on a log graph.



I'm reading Small Signal Audio Design book and he does not favour log pots very well and while he mentions them, points out their issues, he then avoids using them altogether.   There is a large section on using circuitry around the pot to emulate the log law.  One of which is the Baxandall design. The main problems with them are their tolerances are poor, usually 20% and for stereo pairs they are never matched very well resulting in perceptible balance offsets.

Another reference for this configuration, including a HF filter, that uses 4 op amps is here:
http://www.ti.com/lit/ug/tidu034/tidu034.pdf

However using this configuration with 2 opamps and no filter stage I get very little noise on the scope.  It's down around 70mV which I get anyway no matter what I connect the probes to.

[paulca]

If I get a working module in this prototyping, is it worth perf boarding or strip boarding it and then hooking them together rather than using the breadboard in one mass of mess?

I find it's too easy to nudge a wire or a resistor and upset things... or fry an amp apparently.  I'm fairly sure they are dead.  Definately the one that howls at a 13.5V DC offset.

[Audioguru]

I make all my prototypes soldered together on a carefully planned compact stripboard layout. If a part needs to be changed then unsoldering and replacing it is simple. No stray capacitance to cause oscillation and no messy antenna wires to pickup interference. I cut each copper strip with a drill bit so that each strip is used for a few functions to be compact and a few short jumper wires make the remaining connections. Many of my stripboard prototypes looked good enough to be sold as the finished product when only one at a couple were custom designed by me. 

[paulca]

Thanks.  I think I might made a prototype of a stereo baxandall.  If I create that I can put it inline with my current system and test the extra gain on the pre-amp works as expected.  Then I just need to replicate it twice.

The only thing I don't have is a dual pot.   I suppose that's another RS order then   I can hack in 2 trim pots for now they should match the footprint.

I am going to try and use KiCad for planning the strip board.  I'm also low on strip board, might have a bit big enough, might not.  After that I have IC footprint board (groups of 3) that I could use and if that fails it would be dual sided pad board which works, but is messy creating tracks with fuse wire grid style.

[paulca]

Well the prototype board worked out okay.  In so much that it was laid out okay and soldering went well.  I only did one channel.

On testing it however audio plays through it fine, but it's attenuated, the gain pot does nothing and the opamp pulls 30mA and gets hot after a while.  Sometimes is a-miss somewhere.

I've had 2 beers, so I'm not starting to scope it tonight.  Tomorrow maybe, starting at that gain pot to see whats wrong there.

[paulca]

Got it.  The outer feedback line is coming off the - input, not the output.  That will impress it.  I'll fix it in the morning.

[paulca]

So progress!  At last.  Having fixed the fact that pins 6 and 7 where completely backwards on my perfboard, it now works perfectly.  Still gets a bit warm and still pulls too much current, but at least it's only about 13mA this time for a single channel.

I also made progress with the design, (below).

I'm still a bit baffled with impedance.  For example the LineInputStages all have 100R's on the outputs, which is then followed by a 22K, so that makes it 22.1K and will alter the summing amp gain a little, but more importantly when I split the L/R_MIX to the AUX buffer amps do I need separate resistors on each leg of the split, or is it fine the way I have it?

I haven't done the headphone amp yet and somehow I have to invert the signal in it, due to using another of the Baxandall blocks for the master volume, which inverts phase.  I might change this to make a slightly different master volume block based on the same idea.

A few design issues remain.  1.  The active volume controls do not reduce to zero, I believe due to there being a path through the active gain pot to the output, so the original signal leaks through.  I could be wrong.  This is not a worry on the inputs, but is an issue on the master volume.  I would like "OFF" or "Infinite".  Expensive pots or a proper fader will provide this, but probably ££££.

For the headphone amp, Small Signal Audio Design book actually recommends... wait for it... an array of 14!!!  NE5532s in parallel to drive headphones.  14!  I don't think I need that many, it sounds fine with 2 driving 34Ohm headphones.  I might go to 3, 4 or maybe look at a better op amp for the output.

[Audioguru]

Why do you have so many coupling capacitors? Most of your opamps have a gain of only 1 so their output will have the tiny maximum DC output offset of 0.004V without a coupling capacitor and resistor load to ground. Who cares about an offset of only 0.004V max? The offset voltage might even be nothing.

Why do your coupling capacitors have the huge value of 100uF? Feeding 22k ohms then they pass earthquake frequencies down to 0.07Hz! That is one cycle each 14 seconds. Why not use 1uF then the audio -3dB cutoff frequency is 7Hz then will be flat down to the 20Hz lowest frequency that we can hear. The formula to calculate a coupling capacitor value is very important and is simple.

Can you hear a difference between true phase and reversed phase? Does every radio in your house produce true phase? Do you want to add many other mixers to this mixer?

An audiophool would use 14 NE5532 opamps in parallel to drive headphones through solid gold Monster cables. An audiophile uses only one fairly high power NE5532 opamp through ordinary copper cables to drive 600 ohm Sennheiser headphones.

[paulca]

I'm borrowing building blocks from that book!  Now, granted the book is basically his experience in designing real pro-audio 16/24/32 channel mixing consoles, but... a 100uF cap and 22K block is about 10p, I might as well.  Removing them would then require I understand much more possible have to redo the maths of the various parts.

On the coupling caps, there is a reason for the larger uF caps after the summing mixer.  I only scan read it but basically the summing portion has a LF roll off, so the coupling caps should be kept large to prevent further LF damage.

On the gain.  The Baxandall stages have a gain of around -70db to +17db.  The mix sum amp, I believe has a gain of 2.  The Aux buffer is driving un-known external inputs, so I figured decoupling it would be wise.

On the 5532 headphone amp, re-reading it, he shows THD graphs for a single 5532 driving 5Vrms into 200Ohms and how adding a second parallel 5532 drops that THD to almost nothing.   4 can drive 100Ohms with virtually no distortion.  12 are used for 25Ohms.

I doubt I would notice the distortion talked about with 4, driving a 34Ohm load.  So I'll maybe aim there.

On phase.. you are right.  Unless I ever intend to mix the output with another, phase will not matter.

[BrianHG]

Why do you have so many coupling capacitors? Most of your opamps have a gain of only 1 so their output will have the tiny maximum DC output offset of 0.004V without a coupling capacitor and resistor load to ground. Who cares about an offset of only 0.004V max? The offset voltage might even be nothing.

Why do your coupling capacitors have the huge value of 100uF? Feeding 22k ohms then they pass earthquake frequencies down to 0.07Hz! That is one cycle each 14 seconds. Why not use 1uF then the audio -3dB cutoff frequency is 7Hz then will be flat down to the 20Hz lowest frequency that we can hear. The formula to calculate a coupling capacitor value is very important and is simple.

Can you hear a difference between true phase and reversed phase? Does every radio in your house produce true phase? Do you want to add many other mixers to this mixer?

An audiophool would use 14 NE5532 opamps in parallel to drive headphones through solid gold Monster cables. An audiophile uses only one fairly high power NE5532 opamp through ordinary copper cables to drive 600 ohm Sennheiser headphones.

Now I know I like low frequencies, but I must agree with you Audioguru.  Paulca has gone off the rails in all respects when his first schematic would have worked fine beyond his ability to hear have any better.

However, Paulca may be doing this just for creating an over electronically glorified mixer, not that all of this may actually improve the audio.  In fact, it will most likely lower the sound quality just having the audio passing through so many transistors in so many op-amps with so many feed-backs and gain stages.  (I can almost guarantee a proper laid out board of the earlier design will have a better sound especially in the high frequency range...)

As for driving headphones, I stand by my earlier suggestion of using 1 op-amp, driving a 2N3904/6 emitter-follower setup.  Too many op-amps will kill sound fidelity.

[paulca]

I'm borrowing from the design of pro-audio mixing desks.  I found it a little odd that I was hearing two different things from two different places.

Yes the original might have worked fine and no I probably wouldn't notice the difference, but I have at least a small part of my background and family working in pro-audio.  Things are done differently.

An example was pre-amp gain.  Everything audio I know prior to electronics told me to add gain at the input.  But all the amatuer audio stuff in EE forums use pots on the input to attenuate then boost which is a pointless noise generator circuit.

A the same time I still need you guys to help me as borrowing blocks from a book can lead to trouble until I learn the details of how to calculate my way out of a pickle.

Out of interest, if you put all the blocks together from the book and built a 24 channel desk with full inserts, parametric EQ, sends, returns, channel groups, PFLs, AFLs etc. etc. it would probably have in the order of 500-1000 op amps.  Do your CDs sound distorted or noisy?

[BrianHG]

On the gain.  The Baxandall stages have a gain of around -70db...

With headphones especially, you wont be able to mute the volume with only -70db.  Better add a mute button to each stage.

Be careful when signal mixing at lower impedance with higher uF caps.  These caps begin to exhibit some funny responses with their internal ESR and using cheap electrolytics or ceramics here may do funny thing to your signals below 500hz.  In really high quality audio, at the line level, we tend to go for 4.7uF or 10uF film type cap, with only a 50k or 100k load to avoid this issue.  On rare occasion, I've seen 22uF used, but, it was in a 24k/12k ohm XLR balanced load.

[paulca]

As for driving headphones, I stand by my earlier suggestion of using 1 op-amp, driving a 2N3904/6 emitter-follower setup.  Too many op-amps will kill sound fidelity.

This is the books suggestion for a cheap, simple headphone amp:

[paulca]

As someone on here said, if an audiophool spends £400 on a set of speaker cables it WILL sound better to him.

Just to clarify.  I do not mean I want to throw money at the thing and hope that it sounds better because of it.  I believe what I am doing is bypassing the marketing and mark up and building something that would cost 10 times what I could afford to buy already built.

Audio gear falls into three categories. 

* Domestic stuff which will have a quoted figure of 1000 when it only delivers 10.
* Audiophool stuff which quotes 1000, delivers 2000, but only 5 was needed at 100th the price.
* Pro-audio stuff which quotes 100, provides 100 and 50 was needed, but you pay a shed load more.

It's emulating the later I am aiming for.

Of course, most of my gains may be lost in my poor board design, crap cheap components and poor source cabling etc. etc. etc.

[BrianHG]

I'm borrowing from the design of pro-audio mixing desks.  I found it a little odd that I was hearing two different things from two different places.

Yes the original might have worked fine and no I probably wouldn't notice the difference, but I have at least a small part of my background and family working in pro-audio.  Things are done differently.

An example was pre-amp gain.  Everything audio I know prior to electronics told me to add gain at the input.  But all the amatuer audio stuff in EE forums use pots on the input to attenuate then boost which is a pointless noise generator circuit.

A the same time I still need you guys to help me as borrowing blocks from a book can lead to trouble until I learn the details of how to calculate my way out of a pickle.

Out of interest, if you put all the blocks together from the book and built a 24 channel desk with full inserts, parametric EQ, sends, returns, channel groups, PFLs, AFLs etc. etc. it would probably have in the order of 500-1000 op amps.  Do your CDs sound distorted or noisy?

No longer, everything today is all digital.

But, back to older analog.  You need to count the op-amps through 1 source to the central mixing point.  Then that mixing point point to the output jacks.  And yes, after all that, there is some loss and change in sound & the EQs had bypass buttons on the better units consoles for true flat.  Remember, they take all the sources & they only want to go through that mixing console once.  Take that output and pass it through another mixing console, and the sound will be affected.  I'm not saying these consoles were junk, far from it, but, there is a reason we have gone all digital today.  Pass a digital out from 1 digital mixing console into the next digital mixing console and remix with flat EQ settings and I bet the sound will go through perfectly clean if the code was written properly.

[paulca]

I think we agree, however do not be fooled into thinking that "everything is digital today".

In the past few years I have probably seen about 20 mixing desks in person,  I have yet to see a digital one (that wasn't a PC or Mac based thing).

Small 8 to 16 channel mixing desks you would use as an audio semi-pro running a few gigs and recording a few demo tapes would be like buying a higher grade oscilloscope.  Priced, today, around £1000.  A fully digital mixer would cost about ten times that.

PC and Mac based solutions are plagued with problems, not least the thing rebooting or installing updates live on stage, but they are used.  Very often you would have multiple programs producing audio on a single laptop, these would be mixed digitally with an external remote pyhsical control unit, but the sum output of those will end up going through an analogue mixing desk.

If you are going to a pro-audio studio and want a fully digital desk, expect to pay £500 a hour+.

[BrianHG]

As someone on here said, if an audiophool spends £400 on a set of speaker cables it WILL sound better to him.

I spent 300$ CDN (around 250$us) on my 10 feet speaker cables 100% pure silver and measured them to copper with my scope.  Mixing a 40hz sine wave with a smaller 10Khz square wave.  With the thick solid core copper, the peaks on the 40hz squished the squareness and amplitude of the smaller 10Khz signal inside.  The silver cable were almost perfect, though, I could not head the difference.  I still prefer that the signal out of my amp is reaching the speaker where a high current low frequency isn't mushing the fidelity above 10Khz.

[BrianHG]

As for driving headphones, I stand by my earlier suggestion of using 1 op-amp, driving a 2N3904/6 emitter-follower setup.  Too many op-amps will kill sound fidelity.

This is the books suggestion for a cheap, simple headphone amp:

Yes, that's the basic design and you will get a whole lot more mileage out of that circuit instead of parallel op-amps....
In fact, that circuit will ROAST the parallel opamps...
Just double C1 (if you keep the 10k load) and C2.  Make C2 a low ESR cap.  Only take it that far if you have 20hz headphones or are using my Subwoff...

[paulca]

Yes, that's the basic design and you will get a whole lot more mileage out of that circuit instead of parallel op-amps....
In fact, that circuit will ROAST the parallel opamps...

I do hate safety nanny limiting and ... the 5532 has a current limiter.

I know I like my headphones and they are the most expensive I have owned, but they have too easy a life.  I haven't heard them clip or distort very often.  Too soft.

So... adding in the real point of all this is to learn and to play with electronics.... maybe I should do a transistor output stage for the experience.

It can't been anywhere near as frustrating as debugging opamp circuits.... oh wait.  It probably can.

[BrianHG]

heard them clip or distort very often.  Too soft.

They wont be soft anymore with all your gain everywhere + a 30v supply range.  Also, sometimes, there is clipping in your source as well.

[paulca]

BTW... the general pattern in the book is to present the basic "default" textbook solution and the discuss it's problems.  Then present known designs by some amazingly intelligent people, sometimes the author which try to "have their cake and eat it" with little draw backs.

From my main background in software engineering, I see this happen a lot.  And... I agree... sledgehammer to crack a nut often actually makes things worse!

Back on topic, the second iteration of design in the book on headphone amps is this, which addresses a few problems with the first (a) posted above.



I haven't read all my way between the sections, but it ends explaining that you might as well just use parallel 5532s.  I think he owns shares in TI.

[paulca]

heard them clip or distort very often.  Too soft.

They wont be soft anymore with all your gain everywhere + a 30v supply range.  Also, sometimes, there is clipping in your source as well.

I have +17db MAX on the input and only if the gain knob is maxed out.  At "half way", it will have 0db gain.  (remember active gain).

I mentioned i have not decided on the master volume and might not use an active gain "block", but if I do and it's at "nominal" it will be 0db gain.

So a full path through the mixer might have absolutely zero gain until the power amp.  Zero gain and zero attenuation mind.  Flat gain structure, so the only noise added is the op amp inherent noise which is tiny.

Of course I can have at least the +17db on the input preamp and ... unless I change it another +17db on the output volume.

Actually, 17+17db is probably not = 34db, but there might be a risk of actually killing headphones with an accidental volume setting which could be an issue.  I want driven hard, not release smoke.

[Audioguru]

Of course opamps and most power amps have a current limiter for protection against idiots who short the output to ground. But the current limiter usually does absolutely nothing when the current is not too high for it. Why do cars have brakes? They are not used all the time (but idiots drive with both feet on the pedals).

[paulca]

Of course opamps and most power amps have a current limiter for protection against idiots who short the output to ground. But the current limiter usually does absolutely nothing when the current is not too high for it. Why do cars have brakes? They are not used all the time (but idiots drive with both feet on the pedals).

Fair enough.  I'm probably mixing two different things.  Somewhere in the early 90s after "Sony Walkman" devices started to put out some serious headphone volume a group of nannies with pitch forks forced some sort of regulation on such devices and Sony and most major brands adapted these... To limit damaging teenagers ears.  It is hard today to find a device without such a limit.

My experiences with the NE5532 have pretty much told me that by using similar chips for headphone amps their inherent current limit almost matches or comes in under those soft regulations on output.  Which is why almost all headphone out devices today are effectively limited to "Nanny state" mode.

I don't want to cook the headphones or my ears, but I do want to know where the ceiling is.  I want to be able to overdrive them, I want to know where the limit is and the headphones clip.   I don't want it imposed upon me.

Which loops back to one of the first design criteria I selected.  When you have a limited output and a quiet signal, you are incapable of addressing that issue.  You need pre-amp gain to lift it to a normal level to do that.

[paulca]

So the pre-amp board got finished.  Works a treat.  Through my existing setup I can push the headphone amp into distortion which is most likely the pair of 5532s in it limiting.  Sounds clean, but I haven't done much scoping for noise.

It is still current hungry which has me a little perlexed.  My first thought was the 100R outputs.  Which should be much higher.  However it is still current hungry when not driving a load.

I need to get a proper measure of the current as I'm only going by the PSU's consumption which is 30mA before connecting the circuit and 76mA when connected.  Driving a load with it barely moves the current consumption, maybe 1 or 2 mA.

The other thing it could be is ultrasonics making the opamps work hard, no idea.

[BrianHG]

Any scope shots?  From input to output... Feed a 20Khz square wave at different gains and different source amplitudes to verify flat response.

[paulca]

So nothing I have will generate anything but a slightly mangled sqaure wave over about 5kHz even at 5K it's barely square.  I tried both my phone and my PC.  The mickey mouse chinese sig gen kit I didn't even bother with as it's square output is unattenutated and about 10V peak to peak with a 5V virtual ground.

The main issue I seem to have is noise from the PSU.  Something around 80kHz but the scope is having trouble with locking onto it at times.

Images here some are worth a laugh, like my phone trying to produce 20kHz:
https://imgur.com/a/owSEF

Sample of PSU noise, straight off the negative power rail:


This was after putting 470uF caps and 100nF caps from both rails to common.

So as this appears on all op amp output I need to get rid of it somehow.  It's at such a high frequency it shouldn't bother the audible range, but it does bother me and it probably isn't helping!

Max gain, 2k sine:


Min gain, 2k sine:


Just noise on the output.  EDIT:  I fixed the none zero output, was a miss connection.  It now mutes at minimum gain... or near enough.

EDIT2:  Yes, it's annoying that the scope does not capture the right hand side with the time division read out and sample rate

[BrianHG]

What do you have for your PC?  What's the sample rate set to in the 'Speaker Properties' on the 'Advanced Tab'.  Also, in the 'Enhancements Tab',  make sue the 'Disable all enhancements' is selected.

By default, if you never set this, usually it is set to the crummiest weakest setting.

[paulca]

What do you have for your PC?  What's the sample rate set to in the 'Speaker Properties' on the 'Advanced Tab'.  Also, in the 'Enhancements Tab',  make sue the 'Disable all enhancements' is selected.

By default, if you never set this, usually it is set to the crummiest weakest setting.

The PC output is a Texas Instruments PCM2707 USB DAC, but I was trying to use a YouTube video    I might install audacity again at some point.  But I have a signal gen in the post, though it's an FY6600 so it remains to be seen if it will need modified before it's safe to use.

However the phone didn't do a very good job either.

On the PSU noise.  A quick google suggests that meerly decoupling caps will not really help and I need to "bypass" with LC circuits.  I'll do some more reading.

[paulca]

I do have the 1Khz square scope calibration signal...  It might at least be square.

[paulca]

Also it's slightly odd that the amps do not seem to be amplifying the noise much, they seem to just be modulating their output with it.  Maybe that makes sense as the noise is not on the inputs, but the power rail.

[BrianHG]

The PC output is a Texas Instruments PCM2707 USB DAC, but I was trying to use a YouTube video    I might install audacity again at some point.

Arrrggg, 48khz 16bit as a reference.  Even bottom of the line motherboard/laptop audio chipsets usually are better quality as long as you have a good grounding on you cable.  These bottom end USB Dacs are only good if you need another sound card source.  There do exist high quality USB dacs, but, the PCM2707 isn't one of them.

Here is 2 samples, 24khz square which will come out as sine and 12khz square sampled at 48khz.

[paulca]

Scoped them from the audio out and they both show as sine waves.  Slight wobbly, but I was holding the probe to the jack plug.

Come to think... you can't really do a 20Khz square wave with a 48Khz digital source, not very well anyway, it's going to filter down to a sine anyway isn't it?

Hang on a moment.  I doubted my little chinese friend.  I hooked it up and got these:


and...  pushing it a bit.



The output is not attenuatable though and it's a bit much to be putting into the pre-amp.  I can try tomorrow evening, can't tonight, I also need to use the little generator up to a jack plug.

[paulca]

Even bottom of the line motherboard/laptop audio chipsets usually are better quality as long as you have a good grounding on you cable.  These bottom end USB Dacs are only good if you need another sound card source.  There do exist high quality USB dacs, but, the PCM2707 isn't one of them.

That PCM2707  sounds a thousand times better than my Intel HDA on my motherboard.  Maybe not in it's sound reproduction, but in the fact that the PC onboard sound is completely saturated in video card noise.  Was sending me nuts until I happened to hook up that DAC which is a £3 board from Alice on ebay.  No more noise.

My my audio project, which this thread's amp is the heart of I hope to use a PCM2902 or similar.  I'm open to suggestions.

[BrianHG]

Scoped them from the audio out and they both show as sine waves.  Slight wobbly, but I was holding the probe to the jack plug.

Come to think... you can't really do a 20Khz square wave with a 48Khz digital source, not very well anyway, it's going to filter down to a sine anyway isn't it?

If my samples are wobbly, your sound card must be set to 44.1Khz, not 48Khz.
Give these 2 a try, they have a native 44100hz sample rate:

[BrianHG]

That PCM2707  sounds a thousand times better than my Intel HDA on my motherboard.  Maybe not in it's sound reproduction, but in the fact that the PC onboard sound is completely saturated in video card noise.  Was sending me nuts until I happened to hook up that DAC which is a £3 board from Alice on ebay.  No more noise.

My my audio project, which this thread's amp is the heart of I hope to use a PCM2902 or similar.  I'm open to suggestions.

I must have gotten lucky with my Asus motherboard and HP laptop.  Full flat to 96Khz completely noise free even wired direct to a 500 watt amp, using the desktop volume settings.  Note I use the plugs in the back of my motherboard and never wire those front panel unshielded audio jacks.  Also read reviews on the audio performance of the PC motherboard I chose.

[paulca]

Try a pair of headphones on max gain, don't play any audio.  Now move the mouse around.  If you can't hear the buzzt-t-t-t-tbuzzzt-t-t-t you are lucky.

Now bear in mind the video card in my PC, which regrettably sits smack bang on top of the chipset more or less and right beside the back plane connectors, not 2 cm from the audio jacks, has dual 10A 12V rails, dedicated to it alone.  NVideo GTX 980.  (It's so big it doesn't quite fit in the case and pushes one of the HD bays out the front.

[Bassman59]

Back on topic, the second iteration of design in the book on headphone amps is this, which addresses a few problems with the first (a) posted above.



I haven't read all my way between the sections, but it ends explaining that you might as well just use parallel 5532s.  I think he owns shares in TI.

Here's the thing -- that book (which I have) was written a long time ago. Not that the principles have changed since it was published, but the devices available for our use have improved greatly. Back then, the jellybean op-amps couldn't drive headphones, hence the designs using discrete transistor buffers or specialist buffer amplifiers. Nothing wrong with using them. But you can save yourself some aggravation and simplify your design by using something like the OPA551 op-amp. Low distortion and enough drive to clean the wax outta your ears.

[Bassman59]

I think we agree, however do not be fooled into thinking that "everything is digital today".

In the past few years I have probably seen about 20 mixing desks in person,  I have yet to see a digital one (that wasn't a PC or Mac based thing).

Small 8 to 16 channel mixing desks you would use as an audio semi-pro running a few gigs and recording a few demo tapes would be like buying a higher grade oscilloscope.  Priced, today, around £1000.  A fully digital mixer would cost about ten times that.

I'm a mostly-retired soundguy, but several times a year I still get on the bus to mix front-of-house on tour for old friends who are a well-regarded college/indie rock band. We play venues like the 9:30 Club in Washington, DC (the best club in the country), the Sinclair in Cambridge, MA, Rough Trade and Williamsburg Hall Of Music in Brooklyn, NY, World Cafe Live and Union Transfer in Philadephia, PA. Plus we do the occasional festival and outdoor gig (Central Park Summerstage).

Over all of the shows in the last, oh, ten years, I can think of exactly one venue that still has an analog console: the Met in Pawtucket, RI, and that piece of crap needs to get gone. Everywhere else has digital. 9:30 has DiGiCo (which I love), Rough Trade and WCL have Midas Pro 2, Sinclair has Avid SC-48, and the festivals bring out Avid Profile or SC-48 because everyone who'll mix a band on those shows can walk up to it and get right to work.

The big theater here in my city has a Yamaha M7CL. The 400-cap club across the street from it has a Soundcraft Si Performer 3, and the other 400-cap club up the street has a Midas M32. The Behringer X-32 has taken over the small-club market. A friend mixes at a hall in NJ on a Mackie DL32R, which sounds really good but I hate iPad mixing (I've tried, and I can't do it, I need tactile feel.).

Analog consoles for live sound have gone the way of the dinosaur. The last time I mixed on a Midas H3000 was nearly ten years ago. I did see one last summer at Pitchfork in Chicago; PJ Harvey headlined my stage and her FOH guy had one. But he was the exception, and her monitor guy used a Yamaha PM-5D. The reasons are obvious: racks of outboard gear (effects, dynamics, EQs) are unnecessary, it's all at your fingers, and it saves truck pack space. FOH area can be smaller, reducing seat-kill for theaters. Recall of settings makes switchover between acts as simple as a button press. Wireless control means you can stand at a vocal position and ring out a monitor wedge and make adjustments right there, instead of yelling over to the person at the desk.

Anyway, back to the design.

[BrianHG]

and enough drive to clean the wax outta your ears.

 

[paulca]

Apparently you can buy cheap digital 8 channel desks for £600, but most of the stuff I have seen is pubs, clubs, bands and bedroom musicians are using things like Mackie's and Soundcraft spirits.  So maybe they are getting cheap enough for smaller venues, but the only digital mixing I have seen has been using PC/Mac sound card style solutions with USB remotes.  Those are just the electronic musicians though.

Anywhere big enough to have an installed sound system and proper desk/console is a lot larger than the places I have been into. 

Besides I think my info is a few years out of date.  I forget how quickly time passes the older you get   What seemed like a few years ago turns out to be 5 or 10.

[BrianHG]

By the way, Bassman59's suggestion of using an OPA551 op-amp is an excellent idea.  Buy a few in the DIP package so you may swap them if you burn one by accident...  Use gold machine screw alloy IC sockets for this one in your design...

Damn, I now have an app for the little 1/2 watt buggers....

Now, it's time to get serious, try the TCA0372BDP1G at 0.48$ each, that's 1/5th the price of the OP551.   1 Amp out.  Driving an 8 ohm speaker at +/-8v, you will get 8 watts out of this opamp.  And it's a dual, so 1 8 pin device for stereo...

Wow, an 8 watt bugger for 48 cents, stereo.  With headphones, this guy will remove your eardrum from your ears!
You can drive small bookshelf speakers with your pre-amp directly through the headphone output.

[paulca]

Light weights.  I had a TRIO amplifier.  I'm fairly sure they just connected the speaker outs to the headphone jack

I should have had little headphone stickers on that amp with an X through them.  It would have been an Ace in WWII.

[Bassman59]

Now, it's time to get serious, try the TCA0372BDP1G at 0.48$ each, that's 1/5th the price of the OP551.   1 Amp out.  Driving an 8 ohm speaker at +/-8v, you will get 8 watts out of this opamp.  And it's a dual, so 1 8 pin device for stereo...

Wow, an 8 watt bugger for 48 cents, stereo.  With headphones, this guy will remove your eardrum from your ears!
You can drive small bookshelf speakers with your pre-amp directly through the headphone output.

Wow, I had no idea that guy even existed. The data sheet advertises “No deadband crossover distortion provides better performance for driving coils,” and of course headphones are coils. Sounds (pun intended) good to me. Gonna order a few and test.

[Bassman59]

Apparently you can buy cheap digital 8 channel desks for £600, but most of the stuff I have seen is pubs, clubs, bands and bedroom musicians are using things like Mackie's and Soundcraft spirits.  So maybe they are getting cheap enough for smaller venues, but the only digital mixing I have seen has been using PC/Mac sound card style solutions with USB remotes.  Those are just the electronic musicians though.

Right, it’s the EDM kids “mixing” on laptops, but doing that is completely unworkable for mixing a live band with even a modest number of inputs. Your standard two-guitars/bass/drums/vox rock band fits into 16 channels, I’ve been using 26 of late, but remember, there are people playing the instruments, and even the most well-rehearsed band varies their performances. And with that, the person out front behind the faders has to, well, mix the show! And that’s why you won’t see computer-based mixing for live shows, because we need a user interface.

And as the Mackie 24*8 and the Soundcraft Spirit and Allen and Heath GL-series desks wear out and become unreliable, they’re being replaced by Behringer X-32 and the Soundcraft Si and other small digital mixers. Hell, the X-32 costs US$2000 and it has 40 inputs, 22 outputs, USB for recording and playback, and it sounds better than it should. (I am no fan of Behringer.) You’d be stupid to use an analog console instead.

[paulca]

Theory question...

Why does increasing the load ohms on the output effect the opamp gain?  So if for example I am measuring 4.5V peak to peak on the output and then connect my headphones the output voltage drops to around 800mVpp.  The amp would be driving  136 Ohms.  Is this likely to be current limiting?

I think I answered my own question here.  I = 4.5V / 136Ohm = 0.033A.  The output current limit is 10mA.  I = 800mV / 136Ohm = 0.007A

When driving the scope which will have at least a 1M impedance it doesn't current limit as the current is minuscule.   I might just be getting some small understand of impedance.

So given these are pre-amps and I shouldn't really be driving headphones off them anyway, I should test them driving a line level input instead.

When connected together onto the "MIX_BUS" I intend to use 2.2Ks on each which will allow them to run to max Vpp without current limiting.

[BrianHG]

Buy 1 x 8 pin dip 1 amp x2 ON-Semiconductors op-amp for £1 to drive your headphones.  Run it in voltage follower mode for the best response.  Use a series 8 ohm resistor to protect output short.  Get rid of the series caps if you trust your circuit to have 0v average DC out.  Careful, it has a non-standard pinout.

I thinking of making a cheap 30 watt stereo amp, 1hz to 100Khz using 2 of them with an IC heat-sink ontop of them with a 4amp AC out wallwart, not that Amp power ICs dont already exist, it's just going to look cute getting linear 30 watts stereo, ie 60 watts out of 2x 8 pin dip devices for under 5$ total circuit including PCB...

[Audioguru]

try the TCA0372BDP1G at 0.48$ each

Wow, I had no idea that guy even existed. The data sheet advertises “No deadband crossover distortion provides better performance for driving coils,” and of course headphones are coils. Sounds (pun intended) good to me. Gonna order a few and test.

It is fairly noisy (5 times as much as an audio opamp) and has its distortion listed at only 0.5V, 50 ohms like two ordinary opamps I'm parallel. Its slew rate is very low like an old 741 or LM324 so it might not produce the top octave of audio.

[paulca]

I do like the idea of the OPA551 for the output stage.  How does it thermally handle the higher current?  Not that I could drive it's full output into headphones without killing them, but even at lower currents will it not get hot?

That would allow me to finish the design of the amplifier section and get back to the DAC and BT stages.

Still need to prototype two pre-amps to mix amp and... I still need to get get proper square wave response scope shots.

[BrianHG]

Go for the OPA552, costs a bit less and has 12MHz bandwidth, use DIP package.

Your headphones will consume maximum peaks at 100 ma, according to the datasheet, SOA at 85 degrees, if you use a 32 ohm series resistor for the headphone jack, meaning at a 6.5v output at the op-amp pin, you hit 100ma.  (3.25v at the headphone pin, 2x as much as your normal consumer grade headphone output level)  Your power supply is 15v-6.5v= 8.5v, according to the graph on page 19, figure 35, at 25 degrees, at 100ma, just over 10v is allowed, so, 8.5v is safe.  So, we know the IC will be hotter than this.  At 85 degrees, the SOA at 100ma voltage from supply to output pin is 6v.  So, if you use a +/-12v supply instead of +/-15, you will be in the safe zone.  As well, your entire design will run cooler as well at +/-12v instead of 15.

If you want to keep +/-15v,  use the DDPAK-7, you will be in the SOA up to 125 degrees on the copper of your PCB according to figure 37.

[Audioguru]

You can buy the OPA551 in ordinary 8 pins through holes low power opamp package, a tiny surface mount low power package or a small surface mount package that has a metal tab for welding (?) onto a heatsink.

[BrianHG]

You can buy the OPA551 in ordinary 8 pins through holes low power opamp package, a tiny surface mount low power package or a small surface mount package that has a metal tab for welding (?) onto a heatsink.

Everything about Paulca's project is driven to the extreme if you haven't already realized this by now.  Using the DDPAK-7, basic SMD soldered to his PCB with some PCB copper GND plane and vias to the under plane at the heatsink will allow him to max everything out by over 300% in an un-ventilated 50 degree Celsius crammed pre-amp box without any worry of the device going into thermal/current limitation mode.  PCBs with printed copper do some remove heat and it is what is called for in the data sheet.  I've used DPAKs with nothing but the PCB and a rectangle of printed copper with vias to connect the top and bottom with just more copper and they've worked great in apps where I had to dissipate around 4 watts, more than twice of what Paulca needs if he over drives to 300%.

It's what he's going for.

Agreed, if you will only be driving your headphones to 40ma, the 8 pin dip will work fine, especially with a +/-12v supply.

[Bassman59]

I do like the idea of the OPA551 for the output stage.  How does it thermally handle the higher current?  Not that I could drive it's full output into headphones without killing them, but even at lower currents will it not get hot?

I used the DIP flavor in a studio-monitor controller design. Under load at "reasonable" listening volumes with four different pairs of headphones (Sony 7506, Sennheiser HD-280, Grado SR125e, Pioneer something-or-other) they don't get hot. It uses ±15 V rails but the output voltage swing is barely a third of that.

[Bassman59]

Go for the OPA552, costs a bit less and has 12MHz bandwidth, use DIP package.

Watch it, it requires a minimum closed-loop gain of 5, and I found it to be twitchy. 551 was rock-solid stable. And that open-loop bandwidth gets dropped down by the closed-loop gain requirement, so I think it's a wash. (Kinda like OP27 vs OP37, if you remember them.)

[paulca]

So keeping with the flat gain structure idea, where all baxandalls could potentially be at 0db unity gain, would it not make sense to set the headphone amp up to be at "moderately" full volume at that input level?

So that would be an input of normal line level 500mV-1V peak to peak.

Not sure if that idea has legs, but after choosing the active gain stages to prevent attenuate, gain, attenuate, gain all over the place, it would be a shame to stop it there.

[BrianHG]

So keeping with the flat gain structure idea, where all baxandalls could potentially be at 0db unity gain, would it not make sense to set the headphone amp up to be at "moderately" full volume at that input level?

So that would be an input of normal line level 500mV-1V peak to peak.

Not sure if that idea has legs, but after choosing the active gain stages to prevent attenuate, gain, attenuate, gain all over the place, it would be a shame to stop it there.

Full input line level is 1V rms.  Just above 2.5vp-p.
This is also a good level for your headphones max volume, but, be careful.  If you are using the series 32 ohm resistor instead of direct drive, you will need to double the output at the op-amp pin, ie just above 5Vp-p.  This should match a high end consumer DAC unit.

What voltage is line level?
While the nominal levels themselves vary, only two reference voltages are common: decibel volts (dBV) for consumer applications, and decibels unloaded (dBu) for professional applications. The decibel volt reference voltage is 1 VRMS = 0 dBV.

https://en.wikipedia.org/wiki/Line_level

[Bassman59]

So keeping with the flat gain structure idea, where all baxandalls could potentially be at 0db unity gain, would it not make sense to set the headphone amp up to be at "moderately" full volume at that input level?

So that would be an input of normal line level 500mV-1V peak to peak.

Line level for professional audio is defined as +4 dBu, which is 1.23 Vrms or 3.4 Vpk-pk. Depending on the headphones, that might be uncomfortably loud. But just because there's a "nominal" +4 dBu doesn't mean signals coming in are actually at that level. That's why there are input gain trims and output levels on a mixer. I've seen plenty of mixer headphone amp circuits with a gain of ten on the driver amplifier.

[Audioguru]

Is the maximum level for your incoming audio already squashed with a limiter or are faint sounds already boosted with automatic-gain-control? Is so then you do not need much level adjustment.
But for live sounds you should have enough up and down adjustment range of probably plus 20dB down to zero. A logarithmic volume control will cut -20dB when set to halfway. Then what will protect your hearing and headphones if the headphones amplifier has enough power to blow your head off?

[BrianHG]

Is the maximum level for your incoming audio already squashed with a limiter or are faint sounds already boosted with automatic-gain-control? Is so then you do not need much level adjustment.
But for live sounds you should have enough up and down adjustment range of probably plus 20dB down to zero. A logarithmic volume control will cut -20dB when set to halfway. Then what will protect your hearing and headphones if the headphones amplifier has enough power to blow your head off?

He could use my green led clip trick in his pre-amp.  In fact, to protect the next device in even with his line level output, I would strongly recommend it.  Feeding a 30vp-p into an audio amp can also destroy the audio amp as well.

Actually, depending on where it is in his design, he may have to go with 2 red LEDs for the lower voltage.

[paulca]

Hang on.  The pre-amps have active gain.  That does not mean they are putting out 30vpp.  In fact the NE5532 would clip at around 27vpp.  The values used are not my choice I borrowed them from a known design.  They can provide (from memory) -100db to +17db.  A sinewave input set to 500mV peak to peak at max gain comes out at something like 10Vpp.  I will redo the gain tests with the scope at the weekend to get actual figures, but nothing is pushing it anywhere near 30Vpp.

The +17db will only be used when the input signal is too low and I want to lift it to an acceptable level so it can be mixed correctly with another stronger signal.

The plan is (quite 'how you doing') to scope the preamps and mark unity on the pots.  Then I might mark -6, -3, +3, +6 db.  If I'm feeling board I might do more.  They will nominally remain at 0db.

I hope to select values for the summer/mix amp to produce 0db unity.

What I am hoping to achieve is that the final master volume control (either a dual pot or another active gain control) will also sit at 0db.  The headphone amp will have fixed gain so I need to pick values that when everything else is at 0db the headphones will be "comfortably loud".  If I go for a passive master volume it should be "very loud" so I do not need to lift the inputs to satisfy output level.  That would be the master pot at the stops.  If I go for an active gain on the master I will have the ability to "move the fader above 0db" to drive the output to +17db.  In that case the 0db headphone level should be the middle ground, comfortable listening level and the extra gain used for higher listening levels.

Does that make sense?  It's hard to explain.

It's not quite a desk layout as it would have 0db max on the master faders (usually) and an independent headphone level pot which may have positive gain.  In my case there is only a headphone output and so the master and headphone level are the same thing.

As to then connecting the headphone output to another device's line in, I obviously need to consider impedance and not push the headphone output too far as to drive too much current into that receiving device.  Aren't line level inputs "meant" to have an impedance of 100K?

[Bassman59]

Then what will protect your hearing and headphones if the headphones amplifier has enough power to blow your head off?

Reduced supply rails for the output driver would work, but if the amplifier has to drive some esoteric high-ish impedance headphones, that won't really work. Nor, I suspect, would any other passively-implemented trick (the diode, whatever). Paul might wish to consider using one of the THAT VCA/detector chips to put a limiter in his amplifier.

That said: no analog mixing console I can think of has ever come with a built-in limiter on the headphone amp circuit. You could take the console's control-room/cue outputs, run them through a stereo limiter which then drives a headphone amplifier. Glue the headphone amp's volume knob in place, so the only volume control is the console's cue level. This prevents turning up the level after the limiter. And then be prepared to futz with the limiter so it truly does clamp only what's "too loud." (Might as well glue in place the limiter's makeup gain knob.) Adding a limiter feature to a digital console is a "small matter of software," but I can't remember if I've run across a console with such a feature.

Professional in-ear monitor systems have built-in limiters in the belt pack. Of course they have to be set appropriately.

The reality is that a live-sound or studio mix person learns really early on that before you hit the solo/PFL button, your hand is on the cue volume knob and it's turned down. It's really the only way to prevent damage to ears, headphones and studio monitors. I know a guy who used a stereo volume pedal when mixing monitors -- it let him keep his hands on the channels he was adjusting while giving him quick access to his cue-wedge volume.

[BrianHG]

Hmmmm, 3 inputs.  0db gain through 1 input if you set volume to max.  That's 2.6vp-p.  Say all 3 inputs hit a bass drum at full volume, since you are mixing, that's now 7.8vp-p.  Let's say you want a little gain at each input, like many Windows apps have low volume, like a lot of youtube recordings, say you allow for +6db gain peak on each input.  That's potential occasional 15vp-p on your line out.  Now, consumer amps, which have 50k input impedance, are expecting 2.6vp-p, maybe they can withstand 5vp-p before you might damage their input devices, lets be even more generous and say 10vp-p.  Sending them 15vp-p might blow their input, and it only takes a single pulse.  Of, if said amp feeds that through, amplified to the speakers, what will happen there.

Better be safe than sorry.

Same with the potential of blowing a pair of 'expensive' headphones which were designed only to receive 2.6vp-p, or, maybe even as far as a deafening 5vp-p for short durations.

[paulca]

The reality is that a live-sound or studio mix person learns really early on that before you hit the solo/PFL button, your hand is on the cue volume knob and it's turned down. It's really the only way to prevent damage to ears, headphones and studio monitors. I know a guy who used a stereo volume pedal when mixing monitors -- it let him keep his hands on the channels he was adjusting while giving him quick access to his cue-wedge volume.

My brother's old spirit folio had a 1Khz test tone on the master.  You can imagine how much fun that was if you hit it accidentally though a 2K PA.  It wasn't much fun on headphones either.

I believe in the end he super glued it OFF.

[Bassman59]

Hang on.  The pre-amps have active gain.  That does not mean they are putting out 30vpp.  In fact the NE5532 would clip at around 27vpp.  The values used are not my choice I borrowed them from a known design.

Clip level and gain are unrelated. You clip when your output signal bumps up against the supply rails (or depending on the op-amp, a volt or two below the rail). Input clipping level depends on the attenuation presented to the input signal before the amplifier.

As to then connecting the headphone output to another device's line in, I obviously need to consider impedance and not push the headphone output too far as to drive too much current into that receiving device.  Aren't line level inputs "meant" to have an impedance of 100K?

Standard line-level input impedance is 10k or thereabouts. It's high enough to be a bridging impedance to a low-Z driver; most output stages put a small build-out resistor (say, less than 100 ohms) between the driver and the jack for line-drive stability and output-short protection. This means you can also parallel several inputs on one output without loading down the driver. Any higher input impedance incurs a noise penalty.

Driving a line-level input from a headphone amplifier output is fine. You can't destroy the input unless the driver-side power-supply rails are higher than the input side and the input receiver device doesn't have internal clamps.

[paulca]

Hmmmm, 3 inputs.  0db gain through 1 input if you set volume to max.  That's 2.6vp-p.  Say all 3 inputs hit a bass drum at full volume, since you are mixing, that's now 7.8vp-p.  Let's say you want a little gain at each input, like many Windows apps have low volume, like a lot of youtube recordings, say you allow for +6db gain peak on each input.  That's potential occasional 15vp-p on your line out.  Now, consumer amps, which have 50k input impedance, are expecting 2.6vp-p, maybe they can withstand 5vp-p before you might damage their input devices, lets be even more generous and say 10vp-p.  Sending them 15vp-p might blow their input, and it only takes a single pulse.  Of, if said amp feeds that through, amplified to the speakers, what will happen there.

There are a lot of "says" there.  Which could all be countered by "why".  Why would you add 6db gain to an already loud signal and why would you add 6db to all channels?

Anyway, assuming it was by mistake, the input gains where all turned up full and all 3 inputs kicked off at the same time.

Maybe, taking another page from (very) basic mixers, a "clip" light on the output of the mix amp.  A basic red LED which comes on at some fixed Vpp.  It could just be a mosfet or similar with it's gate circuit set to trigger the LED at, say 5Vpp.  When that starts flashing with the beat I know my mix is just a little hot, if it lights up permanently it's too hot.

Of course that could be complimented with similar lights on each input.  I could even go as far as having two lights.  A green set to come on at (whatever amounts to 0db line level) and a red set to come on at "too much".

At some point I'm going to have to bring Moscow out.  Must, Should, Could, Won't this time.

[Bassman59]

The reality is that a live-sound or studio mix person learns really early on that before you hit the solo/PFL button, your hand is on the cue volume knob and it's turned down. It's really the only way to prevent damage to ears, headphones and studio monitors. I know a guy who used a stereo volume pedal when mixing monitors -- it let him keep his hands on the channels he was adjusting while giving him quick access to his cue-wedge volume.

My brother's old spirit folio had a 1Khz test tone on the master.  You can imagine how much fun that was if you hit it accidentally though a 2K PA.  It wasn't much fun on headphones either.

I believe in the end he super glued it OFF.

My iPod had a series of 0 dBFS test tones on it, until the time the random shuffle sent a tone through the PA. That was hilarious.

Actually, the worst control on a mixing console is the "Solo-In-Place" button. It's so dangerous that on some high-end consoles it was put under a shield that you had to lift before you could press the button, kinda like the button that launches the nukes or a History Eraser Button.

What does it do, you ask?

Normally, your headphone/cue bus gets its signal from the main mix bus. (Some consoles let you select the default source.) When you press the Solo/Cue/PFL ("pre-fader listen") button, the cue bus gets its signal from the selected channel, so you can listen to that signal in isolation. It's incredibly useful. You can cue up effects before you put them into the mix, you can listen to one instrument while the whole band is going, all that.

When you enable the Solo-In-Place mode, the cue bus drives the main outputs, replacing your mix with whatever you've cued up. Imagine that during a show, you go to cue up the kick drum in the cans, and SIP is enabled, and now all your audience hears is kick drum (and stage wash).  Yikes!

So the button gets a special cover, and the cover gets taped down, and even though it's a useful feature during sound check, nobody ever uses SIP.

[BrianHG]

No, I don't mean using the leds as indicators....  I mean using the LEDs as zener diodes.  Take the output of you line out op-amp, feed it through the series 100 ohm resistor to prevent potential oscillation from the line out cable.  And tie 2 red 2.1v type LEDs in parallel, 1 forward, 1 backward, to GND.  The led window up to around 1.5v, or 3vp-p, has a current in the picoamp level and capacitance under 10pf.  This roasts most zener diodes, as their reverse bias current is in the high microamps and at low voltages, they have capacitance upward of 500pf.  Then as the voltage goes up, the red led slowly begins to illuminate and the current softly rises to around 10ma at around 1.8v, 3.6vp-p, around 2v, 4vp-p at 20ma, past the drive range of your op-amp.

Your aren't supposed to see these, they should be SMD Leds hidden on you pcb just before the output jack.  They will give you a clean unhindered 3vp-p output, and for louder signals, they will smoothly curve limit the output to 4vp-p.  This is a cheap clean mechanism to voltage limit your output.  Normal clipping diodes wont have the rounding effect, if you string enough signal diodes in series to limit you output to 3vp-p, it will flatten there just like an op-amp reaching it's supply rail limit.  If you use 2 zener diodes in series, their capacitance changes with voltage across each diode and the low voltage ones have high capacitance, though, at 20khz, I doubt they will affect the sound much.  As for their current curve as they approach their zener voltage, they will sit somewhere inbetween the sharp small silicon diodes and the soft leds.

You can use Green Leds for even a softer regulation knee and you get a limiter which begins to cap at around 5vp-p.  Good for +4dbu support, with still a little extra range support.  (I use Green Leds on my Subwoofer project to prevent shooting your woofer out of it's case...)

[Bassman59]

You can use Green Leds for even a softer regulation knee and you get a limiter which begins to cap at around 5vp-p.  Good for +4dbu support, with still a little extra range support.  (I use Green Leds on my Subwoofer project to prevent shooting your woofer out of it's case...)

It’s more fun when the subwoofers catch on fire!

[paulca]

So I spent an hour with LTSpice and hacked out a two channel version of the amp.

It all goes well until I get to the split for MASTER and AUX outputs.

The both output the signal okay, but no matter what I do with the feedback resistor on the master op amp it reads exactly the same as the Aux output.  I've tried with and with ground references on the + input.  I've tried with a single resistor on the mix amp output and individual resistors on the output amps. 

Something fishy is going on.

Any ideas?



EDIT: Never mind, it was the master amp that was broken, R17 (Rf) is useless without a resistor to ground to actually divide the feedback.  Doh!

[Audioguru]

Why does each input use two opamps and many parts to have a fairly low input impedance, a gain of only 1.32 times and pass earthquake frequencies down to 0.17Hz?
I simplified it but it still has a fairly low input impedance, a gain of only 1.32 times but it cuts frequencies below 14Hz.
I would probably simply use a piece of wire. Here it is simplified: 

[paulca]

On headphone output levels.

The headphone spec says, the max power is 1.6W at 1Khz.  Is 1Khz likely to be the worst case?

Anyway directly driving the 38 Ohm headphones 7Vrms gives me 1.3W output with 184mA.  Sounds awfully high and requires 20Vpp through 38Ohms.

I found this test report for the headphones:
http://reference-audio-analyzer.pro/en/report/hp/audio-technica-ath-m50x.php

Which gives their sensitivity as 114.6db / V and their Vrms needs for 96db as 0.09Vrms.  ~2.6Vpp

I had a read through this as well, on headphone outputs:
http://www.apexhifi.com/specs.html

I'd like have at least 110db, concert level, but 130db is the maximum.

Using the spreadsheet in the last link, for a sensitivity of 114.6db / Vrms, into 38 Ohms, aiming for 130db, it gives me 912mW, 5.89Vrms and 155mA around 16 Vpp!

Trying something a little lower gives:
110db gives 0.59Vrms - less than 2Vpp
120db gives 5.9Vrms (yes I see the pattern) - ~17Vpp

So it seems due to the log10 nature of things small increases above 110db result in large changes in voltage. So if I go too low I have quiet headphones, if I go to far I have a dangerous (to other equipment) potential voltage on my headphone out.

As BrianHG said, the line level 2.5-2.7Vpp sounds reasonable.... but do I want reasonable?  5Vpp sounds better, giving close to 120db.

Thus, if I keep things around 2.5Vpp from input gain, summer, the headphone amp needs a gain of 2 and the ability to drive at least 50mA.  That would be 5 5532s in parallel to be audiophile, 3 would probably work.  Or more likely one of those OPAs.

[paulca]

Why does each input use two opamps and many parts to have a fairly low input impedance, a gain of only 1.32 times and pass earthquake frequencies down to 0.17Hz?

Because nearly every circuit you find with a search on "Active volume control" will have a buffer amp.  To answer that further I would have to go back and read the whole chapter on volume controls again.  The answer is most likely because someone far more knowledgeable about these things than I proposed it.

Also, on the gain, remember that R2 and R7 are the volume control!  They will go to 0R and 10k, or 10K and 0R.

Reading this however:
http://objectivesounds.co.uk/articles/active-volume-controls/

Suggests the circuit was designed to have high input impedance was due to accomodate valves.  I'm not sure of this, the link is an only source with 1 amp for an active volume control.

Their simplification to 1 amp does not resemble yours however.

[BrianHG]

On headphone output levels.

As BrianHG said, the line level 2.5-2.7Vpp sounds reasonable.... but do I want reasonable?  5Vpp sounds better, giving close to 120db.

This was for line level.  Using this level out of my PC sound card, my headphones burn my ears, though, 5vp-p would sound better.  Again, if you have a series 32ohm resistor in the line, then. double this to 10vp-p with no load, or at the op-amp output.  Note that the damping factor at low frequencies in the headphones improves without that series resistance.  If you need to overshoot a club/dance scene, like a DJ listening preping up the next track to mix, double this voltage again giving you the +10db out.  This can potentially burn out some consumer grade headphones and make you go death.

Anyways, stick with OPA551 for the premo-grade quality headphone amp output.  5x NE5532s in parallel will just consume a ton of extra current on you power supply and be lower quality sound due to slight timing differences and routing between each parallel IC.  In fact, if you want to drive a killer line level out, add another 2 there.  Use 100 ohm in series from each spare auxiliary cloned parallel output to isolate each one instead of using multiple ones.

[paulca]

The series out resistors, what are the pros and cons? 

I gather if I put a 40R or similar in series with the output I can just do the sums again for 40+38 Ohm load... ?

For the buffered line out, I should aim a bit higher for the series resistor, say a 2.2K since I have used them elsewhere... ?

I have no intention of making myself deaf... or death   I just don't like people telling me how loud my headphones are allowed to go.  I want them to not be at 100% volume.  I want 100% volume to be too much.

[Audioguru]

Paul, your simulation does not say and does not show a volume control but now you say it is TI's "active" volume control circuit.
Medical experts say that sound levels over 85dB cause hearing loss and you are talking about 120dB that is going to destroy your hearing.
If your headphones survive 120dB, their power is 35dB more that the 85dB that begins to cause hearing loss. 35dB is about 4000 times the power. Hit yourself with a hammer that causes a small wound. Then hit yourself 4000 times harder and see what happens.

I protected my hearing with earplugs when in loud sounds but my hearing is normal for my age (72). The high frequencies are reduced to almost nothing so I got hearing aids which create normal hearing for me up to 8kHz. The missing top octave cannot be boosted anymore without destroying the remains of my high frequency hearing and/or there will be acoustical feedback whistling if there is more boost.

[paulca]

So, taking a break from the opamps for a few days, I looked at the USB DAC.

I'm currently thinking of the PCM2906C.  The typical application circuit doesn't look too challenging to implement.  There are a few design decisions to make, but I'm currently thinking of just tying the HID inputs low as I don't want/need buttons.

The issues I am pondering is power.  Obviously TI are making these "Bus powered" as 5V is available off the USB.  There is an extended schematic "For high performance" which appears to use a linear 3.3V regulator, but that's about the only change.  I'm not sure what advantage one gets from running the chip on 3.3V instead of 5V I might do some digging there.

My issue with power is that it creates a route for EMI noise from my PC.  The DAC being digital will not have noise on the audio, but it will have on the power.  Now obviously I could stick a dummy load, like a 100K resistor on the VCC of the USB port and run the DAC off a 5V regulator off the +15V rail of the project (I know there are caveats, unbalanced load on the rails and I have not thought that through yet). 

The problem is then the ground.  I can't leave it off as it is needed for the D+ D- (I believe).  But once connected to the board it will then potentially pollute the rest of the board, including the analogue amplifier grounds with PC PSU and GFX Card noise.  I can still hear it on my current USB DAC, although very quiet, however it IS bus powered.

How do you filter a ground?  After a quick googling, this is as I suspected non-nonsensical.  However I did figure out why the "High performance" circuit runs off 3.3V.  The LDO and associated filtering caps will cut most of the noise out of the USB power. I might take that option. 

This has also made me wonder if I would get rid of the noise on the power rails from the XP Power +-15V PSU if I dropped them to 12V with LDOs after ferrite and cap filters.  I probably don't need the 15V rails.

[BrianHG]

With analog DAC
https://www.ebay.com/itm/CM6631A-digital-interface-USB-to-I2S-SPDIF-coaxial-32-24Bit-192K-sound-card/132272012272?epid=887353946&hash=item1ecc06bbf0:g:wH4AAOSw3ZNZdgsT

https://www.ebay.com/itm/DAC-384KHZ-32bit-Decoding-Module-XMOS-PCM5102-TDA1308-USB-Decoder-Board/232547127833?hash=item3624e39219:g:hbMAAOSw09xZ-sV~

Without Analog DAC, just I2S interface.  (you can opto isolate (use a high speed 8 channel digital galvanic isolator) these and place your DAC IC on the analog side of your preamp, no more connection to your PC's GND)
https://www.ebay.com/itm/CM6631A-USB-2-0-to-SPDIF-Coaxial-Optical-24bit-192kHz-Sound-Card-DAC-Board/322769627550?epid=1339946835&hash=item4b2691a99e:g:omsAAOSwkklZpQ6B

https://www.ebay.com/itm/XMOS-USB-Digital-Interface-I2S-SPDIF-Output-Board-Module-DSD256-32bit-384KHz/272891906642?epid=17007417855&hash=item3f89a00a52:g:mP8AAOSwevdZ5srx

https://www.ebay.com/itm/Italy-Amanero-USB-IIS-Combo384-module-USB-to-IIS-adapter-DSD512-PCM384K/272563718243?_trkparms=aid%3D222007%26algo%3DSIM.MBE%26ao%3D2%26asc%3D50073%26meid%3D61b938851ef9422b8621e9c6035a28b0%26pid%3D100005%26rk%3D1%26rkt%3D6%26sd%3D272563722080%26itm%3D272563718243&_trksid=p2047675.c100005.m1851 (new chipset with full list of OS support)

Both I2S and DAC with optional case:
https://www.aliexpress.com/item/HIFI-CM6631A-Digital-interface-32-24Bit-192K-Sound-Card-USB-to-I2S-SPDIF-Coaxial-Output/32855527872.html?spm=2114.search0104.3.231.6e5f4d70Qt73Ho&ws_ab_test=searchweb0_0%2Csearchweb201602_4_10152_10151_10065_10344_10068_10130_10342_10547_10325_10343_10546_10340_10548_10341_10084_10083_10618_10139_10307_5711211_10313_10059_10534_100031_10103_10627_10626_10624_10623_10622_10621_10620_5711311_5722415%2Csearchweb201603_2%2CppcSwitch_5&algo_expid=c9a89ce6-7582-41c6-b9ec-6810a48f3dd7-36&algo_pvid=c9a89ce6-7582-41c6-b9ec-6810a48f3dd7&priceBeautifyAB=0

[paulca]

Interesting boards, but I was kind of hoping to get everything onto one board.  Some of those board smell of audiophool too.  32bit?

Further, what are the chances of these boards being class compliant USB audio devices that work without specialist drivers/firmware?

But the idea of dividing the DAC into two stages is an interesting one, but don't opto-couplers for USB cost $$$$?  Due to the higher data rates and timing?  Is SPDIF easier to opto couple?  Does I2S not need a ground connection?

The other option is an actual optical, but I haven't checked how much they cost.

[BrianHG]

Interesting boards, but I was kind of hoping to get everything onto one board.  Some of those board smell of audiophool too.  32bit?

Further, what are the chances of these boards being class compliant USB audio devices that work without specialist drivers/firmware?

But the idea of dividing the DAC into two stages is an interesting one, but don't opto-couplers for USB cost $$$$?  Due to the higher data rates and timing?  Is SPDIF easier to opto couple?  Does I2S not need a ground connection?

The other option is an actual optical, but I haven't checked how much they cost.

Look at the SPDIF, it is already coupled.  That black cube isolation transformer beside the SPDIF jack.  SPDIF is difficult to interface compared to I2S.  I2S is on every audio DAC you can buy today.  Example: http://www.ti.com/audio-ic/converters/dac/products.html#p1021=2;2&p84=32;32&p1339=2;2&p348max=384;384

These 32bit 384khz stereo DAC start at 1$.  High end ones at 3$, differential output ones at 4.25$.  The top ones have over 114db SNR.  This is not audiophool.  What if in windows, you turn down the volume to -50db.  With this, you still get you source 16 bits going right through at low volume because of the depth of 32 bits.  As for sample rate, what if you are playing DVD audio at 48Khz, or CD audio at 44.1Khz, or, HD-DVDA/SACD/TrueHD Bluray at 96/192Khz, 24 bit audio.  Keeping your system at 384Khz, or even slowing it down to the standard 192KHz makes for a non-wobbly output like when you tried to play my .wav files which played perfectly clean like a function generator on my old 192Khz 24 bit PC motherboard.  Also, the differential/balanced DAC versions outputs a killer 4.2vrms levels.  No noise here if in your preamp you wire that input channel to receive the differential audio.  Compare that to your 1vrms single-ended old 48khz 16bit usb dac.  Run TI's PCM5242 at 384Khz in differential out, you will get 114db, for 4$.

If you want separation, just use the one of the simple I2S boards.  Mount one sound board in your preamp isolated from the rest of everything.  Those sound boards are self powered from the USB power.  Next, get one of these: https://www.digikey.com/product-detail/en/maxim-integrated/MAX14930CAWE/MAX14930CAWE-ND/5396463  Yes, this is basically a 4 input, 4 output 150MHz optocoupler with 2ns rise and fall time.

I2S only needs 4 digital outputs.  On you preamp, one side of the isolator is powered by the 3.3v output on the USB sound board, & gnd & data fed.  On the other side, you power from your preamp the 3.3v & gnd & the 4 I2S data lines go to your 3$ ti DAC.  The output of the TI dac goes to an input channel on your preamp.  Keep the isolator at the edge of you pre-amp PCB and do not GND or VCC fill under it.  Make the connector off the PCB going to the USB sound board as far to the edge as safely possible.

[BrianHG]

Maybe you should use TI's balanced input high fidelity amplifier design in the PCM5242's example schematic in it's data sheet.  It goes to the volumes that you want and is faster than the OPA551 and it is stereo.  However, to get that full quality, you will need to go all balanced audio for that part of your circuit.  3X cheaper, and more power (1 watt for +/-15v supply), higher frequency than the OPA551.  However, to compare the 2, you would need to convert % distortion for the OPA551 to distortion measured in db for the tpa6120a2.  In fact, the tpa6120a2 roasts your NE5532s to such an absurd degree, so, maybe you don't need such a high quality headphone amp on a mixing console where the sound previously goes through multiple NE5532s.  (In audio applications & driving headphones/line level)

[paulca]

Confession of a blonde moment.

I was concerned about the high current draw.  That current being measured by the INPUT from the bench supply to the dual rail PSU.

Well, of course I forgot to factor the voltage difference.

Highest current pull of the PSU was 70mA.  25mA is the stated idle quiescent input current, but I was reading more like 30mA.  Even if we take the 25mA and subtract that from the 70mA seen...

70-25mA = 45mA
45mA * 12V = 0.540W
0.540W / 30V = 18mA

The opamps are rated for +-10mA, there are two ICs on the circuit + a small amount dropped in the various passives, so they are not being exceeded.  They may be higher than expected, which might be the HF noise I'm dealing with in another thread, I hope, but it's all good.

[BrianHG]

This one was made just for you: https://www.ebay.com/itm/Douk-Audio-PCM1794A-DAC-Decoder-Balanced-HiFi-Parallel-Board-192K-Gold-plated/272982323436?epid=28011370612&hash=item3f8f03b0ec:g:ZKIAAOSwBkRaMjTA

Take a look at the op-amps.
This board has a SNR of 132db since it uses each dac in balanced mode out & is converted to unbalanced using your NE5532s.
Though, you are limited to 24 bit, 192Khz sample rate.  But, this is plenty enough.

In mono mode, that DAC outputs 9v RMS.

[paulca]

I thought I was the overkill suspect?

You seem as overkill on the DACs as I am on active volume controls. 

I think a TI PCM290x will do me just fine.  All I need to do is minimize noise pick up from the PC ground.

[BrianHG]

You seem as overkill on the DACs as I am on active volume controls. 

That's because if you checked the data sheet on the 32bit 4$ DACs, you would see they have a built in pre-amp volume control which can be set through I2C.  No sketchy wired pots in my design.

[paulca]

How do these interface on USB?  Obviously the DAC that produces the i2s, but I mean regarding drivers and sound card emulation.

[BrianHG]

HD Audio drivers for amanero & older windows https://amanero.com/

Xmos has native support HDaudio in Win10, but if you want HDaudio in older windows, you need a a driver.

Same for the CM6631a.

All the other OS, ie Mac & Linux natively support USB Class 2.0 audio.

If you don't have drivers for older windows older than Win10, the boards will run in USB Class 1.0 48KHz audio mode.

I'm  thinking of doing a DAC board with amanero and my Win7 PCs.

[paulca]

So here is an LTSpice file for an example of where I'm at for two mono channels.

This is without a baxandall active volume control on the master, just a pot on the headphone amp input.  The only downside to this is that the pot interacts with the "aux out" amp (top right).

"Input Gain" (x2) and "Master Gain" show pots. Both inputs are set for slight attenuation, master is set to MIN in the up load.  Increase R19 and reduce R20 to turn it up.

C2 and C8 can be removed, I know, along with R7 and R14.

[Audioguru]

What is chopping the waveform?

[paulca]

What is chopping the waveform?

Is that just a Spice artefact maybe?  I'm sure it has to have a resolution it simulates at.  Maybe "minimum step size" in the simulation command might change that.

Anyway I ran through some normal cases and some worst cases.  With a quiet 0.3vpp input gained on the pre-amp I can still get 8vpp and 100mA on the headphone output.  Similarly with a loud 1.7Vpp input and slight attenuation on the pre-amp I can get sensible output levels.

Yes, if I max the input gain on a loud signal and put the master volume full I get close to 24Vpp on the output which will not impress things.  However that would be a mistake.

I don't like that the master volume pot attenuates the aux out though.  Putting another buffer before the volume pot might help.

The next thing I think I need to look at are signal and peak lights, set at sensible points roughly related to dB.  Just not sure what those levels will be, 0db and +10db?  Or the best way to do them.  Zener or voltage divider and a known Vf diode.

[paulca]

So I'm closing in on the rightly.

I have a board in manu for a prototype pre-amp.  Think I have the circuit for the mixer amp and output amp sorted. Also making progress on filtering the DCDC converter power supply.

The open items are:

Master Volume
Limiting the mixer amp output to something non-destructive in the event of a mistake in gain.

The THAT4301 Chip mentioned, by Bassman I believe a number of pages back.  I like the idea behind this.  They are hard to find though.  However I want to explore this idea.

Is there such a thing as a voltage controlled volume / compressor / limiter?  Preferably both in one IC.  ie.  A voltage from a pot selects volume output, which will never exceed an RMS voltage limit set by another pot.  This would ideal, but I would settle with putting the volume control separate (before) the limiter.  High bonus points if it provided indication output when it was limiting, so I can light a "SAFETY" LED or similar.

A quick search on Farnell makes it seem finding one of these chips could be quite difficult.

[Audioguru]

Why is the maximum output level so high that a protection compressor circuit is needed to prevent destruction of the headphones and your hearing??

[paulca]

So I can preamp up quiet signals.

[Audioguru]

The voltage gain allows you to increase the loudness of quiet sounds. The supply voltage and load resistance determine the maximum loudness. Your supply voltage is too high for your load resistance, causing the output power to damage the headphones and your hearing.

[paulca]

I read a few guides on limiters and the first thing proposed and very quickly dismissed was limiting the supply voltage.

Changing the supply voltage to limit the output power just sounds wrong.  All that will achieve is clipping the amps.  That is much more damaging to both ears and particularly speakers and headphones.  Stacking enough resistance onto the output to make the maximum achievable voltage by the mix amp, given a mistake, not be too loud will ruin the gain structure and require running the amps at 20Vpp to get full volume.

Having enough headroom for any output and then limiting the ultimate peak voltage to prevent damaging headphones or downstream equipment seems to be to be a much more professional way of handling things.

Besides, outside of nanny state consumer audio headphone amps will typically melt a pair of headphones if not treated right. 

Further, the limiter will be on the output of the mixer amp, which is effectively my master bus.  It has an aux output.  If I have that connected to something, it doesn't matter if I limit the headphone amp supply rail as it will still fry other equipment unless I limit it's voltage too.

Limiting supply voltage is not limiting it's clipping.  I'd like to explore the option of using an actual limiter/compressor IC.

[BrianHG]

So to be sure, you want to lower the dynamic resolution of your music, not protect you amp and ears.

The cleanest noise free way, with optional independent channel limiting is to use 2 vactrols, or, a stereo vactrol.

Any IC you use will most likely introduce noise and distortion, + you will need +5v, or 12v, + such ICs for some reason like a DC offset you might have to deal with.  Otherwise, you will need to go with a full 2x high quality 4 quadrant multiplier IC from analog devices, 8$ each (You need 2 unless you have gone mono), + a additional set of opamps to drive the gain of each of them.  But, in the audio band, the vactrols are an old, but kinda known around to be a perfect voltage controlled resistor exceedingly high quality for audio apps.

Now thanks to the loudness wars, and the music demo you left to me on another thread, note that unless you redesign these limiter circuits to interpret how the human ears responds to loudness, any circuit you try will still make some recordings sound louder than others since the source already has been clipped and will as one volume already, but be louder than that because of the embedded square wave nature.  So, to solve this problem, once again add a few more op-amps into the gain control circuitry if you wan to fix this.

My green leds at the inputs of your final stage driving opamps controls your upper limit, yet, isn't quite as harsh as lowering your power supply stripping the dynamic functionality of the opamps & the rest of your stages.  And all it requires is 2led + 1 resistor / channel.  Around 20 cents compared to 16$ of analog multiplier, or vactrols, and supply filter caps and around 6 additional opamps + signal diodes+ 4 trimpots, 2 for max signal level, 2 for recovery speed, + diodes or transistors & more caps.

[paulca]

Thanks Brian.

I am keeping the LED solution as a fall back, but there are dedicated stereo audio limiter ICs out there.  Bassman linked the THAT devices which are an option, but I wanted to explore if there are others out there.  Basically they are audio processor ICs which provide limiting, compression, companding or expanding.  So they are, one would assume sensitive to the dynamic range and physco-acoustics of audio.

The THAT ones seem hard to find are expensive etc.  So I was asking if anyone had experience of different chips or even what kind of parametric search I should be running to find them.  I tried looking on Farnell and RS but didn't find anything that looked right, so I'm probably not searching for the right thing.

On the LED clamp circuits, I have found quite a few variants, some even designed to progressively limit as the overlimit increases.  Do the LEDs actually light up when they clip due to the voltage exceeding their Vf?  I would assume so, which would be cool.  It gives me that clip light.

[BrianHG]

Thanks Brian.

I am keeping the LED solution as a fall back, but there are dedicated stereo audio limiter ICs out there.  Bassman linked the THAT devices which are an option, but I wanted to explore if there are others out there.  Basically they are audio processor ICs which provide limiting, compression, companding or expanding.  So they are, one would assume sensitive to the dynamic range and physco-acoustics of audio.

The THAT ones seem hard to find are expensive etc.  So I was asking if anyone had experience of different chips or even what kind of parametric search I should be running to find them.  I tried looking on Farnell and RS but didn't find anything that looked right, so I'm probably not searching for the right thing.

On the LED clamp circuits, I have found quite a few variants, some even designed to progressively limit as the overlimit increases.  Do the LEDs actually light up when they clip due to the voltage exceeding their Vf?  I would assume so, which would be cool.  It gives me that clip light.

Yes the LEDs light up.  On my Subwoff board, since the LEDs are only treating the signals below 25hz, especially the sub 10hz-2hz signals because of the massive VLF gain, it has a cool effect of showing you the breeze, or wind direction, or, impact direction when watching movies.  Since 1 LED handles the positive blow, then the other LED handles the negative.
I should video it since it is slow enough to cleanly capture at 60fps.

The analog devices part number is an AD633.  In the data sheet, they have example schematics on audio volume limiting.

Vactrol compressors: (You can change the Vactrol with a opto-photo-fet H11F3SR2M, you can also drive this guy right into your active volume control.)
https://www.google.ca/search?q=vactrol+compressor+schematic&client=firefox-b&sa=X&dcr=0&tbm=isch&tbo=u&source=univ&ved=0ahUKEwj_9qaY147aAhUE74MKHbcRD28QsAQIKA&biw=1317&bih=714

[BrianHG]

I was wrong about the price of vactrols, they are dirt cheap:
https://www.aliexpress.com/item/5PCS-VTL5C1-DIP4-VTL5C/32714837501.html?spm=2114.search0104.3.2.1234651743yxny&ws_ab_test=searchweb0_0,searchweb201602_3_10152_10151_10065_10344_10130_10068_10547_10342_10343_10340_10548_10341_10696_10084_10083_10618_10139_10307_5711215_10313_10059_10534_100031_10103_10624_10623_10622_10621_10620,searchweb201603_50,ppcSwitch_5&algo_expid=125acb79-4ed6-4b4c-8f1a-1a4c693c72fb-0&algo_pvid=125acb79-4ed6-4b4c-8f1a-1a4c693c72fb&priceBeautifyAB=0

Easy to breadboard with...

[Audioguru]

I hate the way an audio compressor causes the level of the background music to pump up and down with the beat of the music or with the speech of the DJ. Radio stations replaced the sound man's hand on the volume control with a compressor so that their average and peak sounds can be louder than other radio stations.

[paulca]

I hate the way an audio compressor causes the level of the background music to pump up and down with the beat of the music or with the speech of the DJ. Radio stations replaced the sound man's hand on the volume control with a compressor so that their average and peak sounds can be louder than other radio stations.

Not all compressors are created equal.

If you are listening to anything recorded in the last 30 or 40 years it has been compressed.  All commercially recorded digital music is compressed.  Heavily compressed.  Simply because you cannot clip it ever.  Your choices are to run the volume very low to give you plenty of headroom and thus quiet on domestic audio equipment or compress it. Most is multi-band compressed.

What you are describing is a simple compressor with too short a release time. 

If you put Radio1 on and put it into a level meter you will find the meter barely moves.  This is extreme compression due to trying to absolutely maximise the dynamic range of the medium.

[paulca]

Actually maybe "dynamic range" was the wrong word in the last statement.

Anyway.  A few questions...

These LED limiters.  I like the fact the light lights up, so I got to thinking.  How can I split off a signal for the pure purpose of lighting an LED at a set peak voltage without altering the signal?  Would I need to buffer the signal off with another op amp?  Is there a simpler way?   I'm thinking of an LED to show "Signal" and the limiter LED showing "Clip", so I can put them on the preamps and just the clip on the master as a limiter.  This will allow to tune the inputs visually with signal being tuned to be lit at 0db line level and clip at about +10db or something I'll calculate later.

On ICs I can find companders, but it looks like they are "all or nothing".  What I want is a compressor with a threshold.

When I did digital DJ mixing to CD with an absolute hard limit of 0dB I used two compressors in series at the final stage of my master filter chain.

Compressor one was set to -6dB and had a short rise and long fall time with limited gain.  So it was soft and non-intrusive and basically compressed the signal over -6dB lightly.  This allows loud music to compress up into the top volume area.  Compressor two had a -1dB threshold a minimal rise and short fall time and infinite gain authority.  These were the default settings for the "Hard limiter" settings.  The reason it was set to -1dB was, under test it still breached the 0dB limit occasionally.

While I realise this would be very complex, expensive and probably noisey to do in analogue electronics it is what I'm after.  Signals well below the limit should be left alone.  Signals approaching the limit should be compressed, signals above the limit should be suppressed, ideally by attenuation rather than clipping.

However I'm getting the feeling this will be tricky to achieve and I'll end up with some form of LED clamp limiter.

[Audioguru]

AliExpress is Chinese like ebay and Amazon. Don't they also sell cheap Chinese fakes and manufacturer's rejects?
Years ago they made slow compressors with an incandescent light bulb eventually lighting an LDR that took a long time to react.

Recently I made an LED blinker that alternates lighting groups of LEDs quickly then slowly then quickly then slowly. instead of abrupt switching I used an LED lighting up an LDR and when it switches to make the alternating go slowly you can see that it takes some time to change its resistance.

[Bassman59]

Limiting supply voltage is not limiting it's clipping.  I'd like to explore the option of using an actual limiter/compressor IC.

The THAT 4301 I mentioned has everything you need to implement a limiter. Add resistors/pots for threshold, output, ratio and you're set. Read THAT's app notes, it's all there.

You will find this to be a frustrating challenge, taking into account varying source input levels and different headphones with different impedances and sensitivities. There is no one-size-fits-all solution.

But here is the deal. A peak limiter is not really all that different from a clipper. They sound bad.  My Genelec 1031A studio monitors have vactrol-based limiters. Detectors monitor the levels going to the treble and bass amplifiers (after the internal active crossover) and when they exceed the fixed threshold the vactrol turns on and reduces the gain of the input buffer circuit. They bark at you when you hit them. That means "turn it down."

A perhaps better approach is a compressor with a high ratio, a fast attack time, a long hold time and a long slow release time. A peak is hit, the gain is quickly reduced and then the gain stays down for a while. The long hold time helps keep modulation (pumping and breathing) to a minimum. As long as the input level goes above threshold during the hold time, the gain shouldn't change. "Modern" low-dynamic-range-music will probably guarantee this. But, invariably, the input will go down, and the gain will go up.(Or, more correctly, the attenuation reduces.) But this will still sound different from the uncompressed source, so the gain-reduction indicator should tell the user, "turn it down!" You don't want to actually have the thing working in gain-reduction mode.

Or, you could do what most headphone amp designs do, and skip the limiter entirely. Be aware of your levels and what you're listening to.

[paulca]

Yes, I know I keep getting accused of trying to drive a Marshall stack through my headphones, melt the drivers and perforate my ear drums that is not my intention.

However I kind of want that option, if I choose to.  I never want the volume control to get to 11.  10 should be painful for me and the headphones and probably never used, but 11 should be available.

The limiter is more for mistakes.  Say I'm listening to a really rubbish rip of an audio book and the gain is way up, the media player files are out of order and a digitally remastered Slayer track jumps on giving me 20Vpp on the outputs.  Not the volume at 11, but more the volume at 27!  POP goes my headphone drivers and the input amp on my desktop speakers, plus extreme pain in the ears.

This scenario has happened to me, listening to a really quietly ripped audio book at max gain and the media player selected a really loud modern recording next.  Nearly fell out of my seat.

Similar funny story... listening to music on headphones drunk one night at about 4am, I leaned back in the office chair, promptly unbalanced it, fell over backwards spilling beer all over my face and to make matters worse the headphones got pulled out of the amp which promptly lit up the speakers at full tilt and caused a glass lamp to fall off the corner unit and smash into a flower pot.  Instant devastation, total comedy sketch... and the neighbours asked me what the **** I thought I was doing the next morning.

I prefer switched amps where you can have both headphone and speaker or one or other at your choice.  This little box gives me that... if I add a mute on the aux output to the speakers.  Nothing more embarrassing that listening to something dubious on headphones only to realise the speakers are still on!

On the later... how do opamps take to having their outputs switched to floating?  Is it a better idea to switch it to ground through a 1meg?

The THAT ...  how much "compressor" like control does it yield?  You mentioned "threshold" does that limit the actual intervention point or will it compress the output even at lower levels?  Does it provide attack and sustain control?  I did scan read the datasheet in work today, but not in details.  You are leading me to think I should.

[Bassman59]

Yes, I know I keep getting accused of trying to drive a Marshall stack through my headphones, melt the drivers and perforate my ear drums that is not my intention.

However I kind of want that option, if I choose to.  I never want the volume control to get to 11.  10 should be painful for me and the headphones and probably never used, but 11 should be available.

Well, then, what you want seems to be at cross-purposes with what you want.

The limiter is more for mistakes.  Say I'm listening to a really rubbish rip of an audio book and the gain is way up, the media player files are out of order and a digitally remastered Slayer track jumps on giving me 20Vpp on the outputs.  Not the volume at 11, but more the volume at 27!  POP goes my headphone drivers and the input amp on my desktop speakers, plus extreme pain in the ears.

This scenario has happened to me, listening to a really quietly ripped audio book at max gain and the media player selected a really loud modern recording next.  Nearly fell out of my seat.

iTunes has a feature called "Sound Check." When you import an audio file into it, it scans the file and looks for basically its "loudness," and it keeps track of that. On playback, the output volume is adjusted so that all songs you play are at the same "loudness," to counteract exactly what happened to you. It's not compression, it's just level normalization. The level is never changed dynamically during a song. Instead of you adjusting the volume for each song (to compensate for loudness differences), iTunes does that for you.

It works well, usually. There's not much it can do with music that has high dynamic range, such as classical, which might start a piece at piano and then go to fortissimo. iTunes will normalize against the fortissimo, but when you listen you might still have to reach for the volume knob during the piano parts.

I prefer switched amps where you can have both headphone and speaker or one or other at your choice.  This little box gives me that... if I add a mute on the aux output to the speakers.  Nothing more embarrassing that listening to something dubious on headphones only to realise the speakers are still on!

My studio monitor controller has such a feature. When you switch to a new output, the others are muted. It has a "override latch" feature, which lets you have more than one of the speaker outputs on at once.

On the later... how do opamps take to having their outputs switched to floating?  Is it a better idea to switch it to ground through a 1meg?

Short answer: op-amp outputs are never truly floating.

Longer: Look at any op-amp schematic. For the non-inverting configuration, the output connects to ground through the feedback network.

For the inverting configuration, the output is connected to the input via the feedback network. [img]https://www.electronics-tutorials.ws/wp-content/uploads/2013/08/opamp6.gif[/url]

The THAT ...  how much "compressor" like control does it yield?  You mentioned "threshold" does that limit the actual intervention point or will it compress the output even at lower levels?  Does it provide attack and sustain control?  I did scan read the datasheet in work today, but not in details.  You are leading me to think I should.

Yes, you must RTFDS. Remember, THAT is a company that specializes in audio circuits, so their parts are intended to be use in the standard dynamics-control devices (compressors, limiters, gates, expanders).

Threshold sets the point above which attenuation occurs. It does not attenuate when the signal is below the threshold. (If it did, it would be a downward expander.)

It does provide attack, hold and release control. (Not sustain, you are thinking of analog synthesis.) Well, you have to design in the pots to set those controls, which affect the detector operation. It is the output of the detector which drives the VCA's gain-control input.

[BrianHG]

Just use Microsoft Windows built in multi-band loudness equalization feature in your 'Speakers' enhancements.  And highight the filter and click on settings, select 'Short'.  Everything will come out from your PC at 1 volume to your ear, set by the system master volume.  Great for all those stupid Youtube and system audio notifications.  Everything will have a sane volume output.

To replicate the effect of this particular very useful digital loudness equalizer, you need a multi-band limiter in your IC.  It would need to be sensitive and adjusting the individual volume of multiple frequency ranges separately based on each range's volume, not just use the voice band as a level reference and adjust the entire audio band based on that selective level.

[Audioguru]

You never disconnect the load on a vacuum tubes amplifier because the inductance of the output transformer will make huge voltage spikes. A solid state amplifier does not have that problem because it does not have an output transformer.

Some solid state amplifiers have a single positive supply and a capacitor coupling the speaker and blocking the DC. They need a load to charge the capacitor when the speaker is disconnected when powering up to avoid a huge POP when the speaker is connected to the amplifier.

[BrianHG]

Limiting supply voltage is not limiting it's clipping.  I'd like to explore the option of using an actual limiter/compressor IC.

The THAT 4301 I mentioned has everything you need to implement a limiter. Add resistors/pots for threshold, output, ratio and you're set. Read THAT's app notes, it's all there.

That THAT IC's data sheet has some nice specs in the data sheet.  3 things I would have like to have addressed in the data sheet would have been scope shot, IC-IC gain matching and a better way of addressing stereo, both with a common focused ALC, or independent channel rather than having so many separate controls.

Example, with 2 ICs, and the voltage control input, each IC has a gain of around -95db at +0.6v in and a gain of 0db at 0v in and a gain of 30db at -0.17v in.  How precisely matched is this voltage gain setting from IC to IC?  The spec of +/-15mv isn't too convincing since that can be around +/-3db difference per IC.  The data sheet also says these figures are affected by temperature since the spec has a listed temp-coefficient.  If you want a perfect clone stereo volume control, do you need to match ICs + heat-sink them together?

(Wish I new about this chip 18 years ago for my CATV converter, would have been great for the mono volume control...)

[paulca]

Yes That4301 looks like the right bit of kit.

Here's the crunch however.  To do a stereo master volume control with it, I need two of them.  I would need to buy 2 in DIP package to breadboard it out, then buy another two in SSOP for the actual PCB.  That's £60 all in, plus a few quid for the correct trimmers and pots.

So the question comes down to, is it worth £60 for the output to be compressed limited or do I just go for the LED clipper.

I'm leaning towards the LED clipper.  I know it will sound harsh, but maybe that's good.  I should be set to a level beyond which the circuit should never be running, so I shouldn't care what it sounds like.

[paulca]

So I have to make some decisions.  Too many things in the project, now I am considering metering and the headphone amp stage etc. all require me to know my maximum peak voltage.

A few secondary design limitations steer me towards 5Vpk, giving 10Vpp.  These would be notably using an ADC in an MCU on a 5V rail for metering.

I would like to translate that 5Vpk to the SPL of my headphones to determine if it's too much, too little or Goldielocks.  That then allows me to pick output series resistance and the final h/phone amp gain.  I also allows me to validate if the "nominal" level of 0.615Vrms will work.

It will also allow me to design and test LED/Diode clip circuits to enforce the absolute maximum peak voltages to protect the lower voltage parts of the circuit, such as the MCU and downstream connected equipment, such as my desktop powered speakers.  It will also allow me to consider dropping the rails.  If I can get below +-8V I can probably go for some form of linear rail spilt solution and still get my 9-15VDC input range.  Although I still believe that requires a switch mode voltage inverter IC.

I've made a spreadsheet showing various voltages rms, pk, vpp against dB based off a reference "nominal" internal voltage.  The bit I can't seem to get the spread sheet to calculate is the effective SPL at the headphones based on their sensitivity.

Of course I can find calculations based on dB per mW, but I have specifications only for dB/V.  I tried using the same equation and it gave me non-sense.

Can anyone help with that?

The equation I was using was:

sdBV+10*LOG10(Vpk)

sdBV is Sensitivity dB / V.  Which has the figure 114.6.  Straight off I can see that 1Vpk is going to be quite loud.  But is 5Vpk too loud?  Is it even db/Vpk or db/Vrms?

[paulca]

I'm wondering if it was as simple as changing the 10x to 20x in the equations.  It gives me much more realistic figures.

It also now confers with this test report:
http://reference-audio-analyzer.pro/en/report/hp/audio-technica-ath-m50x.php

It gives 96dB as 0.093Vrms and 0.233mW

My spreadsheet (at -17.39db) gives 95.999dbSPL @ 0.083Vrms and 0.182mW.

These are internal amp levels assuming the master and headphone volume are to FULL.

To listen at a modest 96dB the headphone output (or master volume) would need to come back to -18db for a signal at 0dB internal.

But it "looks" like it's all starting to check out.  If it's correct I can have unity gain on the h/phone output amp, making it a voltage follower config.  I will also be able to meter the 4.9V max peak with a 5V rail-to-rail peak detect circuit and a 5V ADC.

[paulca]

So this arrived today.  Only took 6 days for manufacturer and shipping from AllPCB.

The only bemish is my own doing.  It's too easy to get distracted by the zoom functionality and forget how small the board actually is when you plop a graphic on it.

The top black graphic is suppose to say "Protect from sunlight"

A Baxandall active volume control/pre-amp.

[paulca]

And Tada!

It works.  Slightly more an anti-climax this one, compared to the DAC as I can't really put this into proper use on the desk as it still requires the +-15V supply which isn't really practical yet.

However the point was simply to check the circuit works.  By building small prototypes of each major component I get to iron out any anomolies, rather than try to debug them all at once on a large board with all of them on it.  At potentially considerable overall cost, however spread over months.

I still have to scope it and see how it responds to different torture, but it functions from a quick ear test.



The next board already in the post is the PSU with filter.

Following that is the headphone amp prototype, which I home to actually put into a case and make generally useful.

[paulca]

So a little update on progress.  Things are coming together.  Or... rather they aren't yet.  They things are starting to exist.

Linking in a few cross threads on sub projects:

Headphone (power) amp - created as a standalone.
https://www.eevblog.com/forum/beginners/circuit-review/msg1471160/#msg1471160

Ideas and prototypes for meters.
https://www.eevblog.com/forum/beginners/audio-bar-graph-small!/msg1465125/#msg1465125

The DAC:
https://www.eevblog.com/forum/beginners/audio-dac-output-low-pass-filter/msg1450490/#msg1450490

So right now the DAC and headphone amp are in service and working fine, with caveats.

I added gain to the DAC (swapping 12K input resistors for 3.9K) and while it doesn't upset my gear it upsets my brothers amp which clips horribly.  <shrug>  I gained to to see if it would run headphones, it did, but weakly.  However it drives as a preamp absolutely fine, it drove my existing headphone amp to where I no longer needed full volume for loud listening.  In the full circuit I will put the gain back to what it was as it has a premp stage after it.
Issues: The L and R output are swapped, doh!

The headphone amp.  As it's standalone I couldn't build it for my internal chosen levels, so I gave it a whack of gain, 7.666 actually.  Powered it from a voltage divider rail split with two caps.  It performs excellently.  I "can" get it to full throttle with my headphones, but I can't tolerate it for very long.  Face melting.  I've used it all weekend and on a dial that goes from 1-10, 3 is comfortably loud, 4 is kicking loud, 11 is only tolerable for a short period.  But, this achieves my goal, I don't need to run my volume controls to MAX all the time it's back in my control.  The downside of high gain is it amplifies noise, so even touching the bare metal pot make audible noise... and angers the voltage divider noise below.

Issues: The voltage divider virtual ground is highly sensitive to power rail noise.  Also the input biasing to that virtual ground is sensitive to cap matching.  When I was using 220uF input caps to minimise low end roll off, they were unmatched by about 20% and the right hand channel ended up biased about 2V lower than the feedback reference, so... no output.  22uF are easier for the 22K bias resistors to pull, but I'm tempted to lower them and increase the cap size again.  If I lower the 22k bias resistors (biasing the amp side to the virtual ground) it will form an RC filter with the cap and I might put them back up to 220uF.  Alternatively, I could buy higher tolerance caps.

The voltage divider noise, in the short term with the board I have, I might lower the 4k7's to 2k2's or even 1ks to make it a bit more powerful to avoid uA range pick up.  Burning a few mA is worth it. 

I'm thinking of making a revision two of the amp with these improvements and ... maybe a opamp virtual ground buffer and a chain of power rail filtering caps.

What I'm learning is that when you don't build things to a price, but to a spec they sound better.... or maybe that's just confirmation bias and pride.

Still on the list:
Prototype the bluetooth module.  Probably breaking it out with mod wire on a perf board.  It needs a differential to single ended amp stage and it would be nice to combine that with the preamp, but I doubt I have the experience to do that... or it might not work.
Prototype a few bar graphs for the meter module.
Start trying to layout a full board with everything on it.
Start thinking about how I house this thing.  I'm already thinking it's going to end up in a case that's 4" or 5" sqaure with the meter board off the back, angled up, console style.

[paulca]

The thought of EQ is torturing me now.  Bass and treble controls at least, but what I'd probably prefer is a mid-range attenuator, ideally parametric...  I'll explore.

[BrianHG]

You should have just done an all digital mixer.  Included everything in the DSP, volume, vu meter, mixing, auto volume limiting, full graphic eq, built in MP3 player, WiFi, separate headphone and line out cross channels, and a few more capabilities...