Hi community,
I'm building a karaoke setup with the components as seen in the image attached.
The problem is that even with the mic preamp the volume of the Mic is about half of the volume of the other channels. I'm looking for a way to get the inputs for the mixer at roughly the same volume.
The datasheet of the preamp can be found here. It says that the preamp has a fixed gain of 18dB which could be increased to 40dB if I understand well. I've removed the pre-soldered resistors and tried a few different values but this didn't increase the gain.
So now my question is:
•Should I be able to increase gain by setting different noise gate and compression values by changing the resistors?
•Is there a(nother) way to increase the gain?
•Assuming the phantom power supply has balanced xlr output, of which I now only use hot and neutral to get mono, could I increase input level to the preamp by different wiring or additional components?
•Any other solutions, like different mic-preamp or a simple way to reduce the input signals of the other channels...etc?
Sorry for the noob questions, I'm learning by building and help is much appreciated.
Thanks in advance and rspct to all audio engineers all over the world!
-Malik
I'm building a karaoke setup with the components as seen in the image attached.
The problem is that even with the mic preamp the volume of the Mic is about half of the volume of the other channels. I'm looking for a way to get the inputs for the mixer at roughly the same volume.
The datasheet of the preamp can be found here. It says that the preamp has a fixed gain of 18dB which could be increased to 40dB if I understand well. I've removed the pre-soldered resistors and tried a few different values but this didn't increase the gain.
So now my question is:
•Should I be able to increase gain by setting different noise gate and compression values by changing the resistors?
•Is there a(nother) way to increase the gain?
•Assuming the phantom power supply has balanced xlr output, of which I now only use hot and neutral to get mono, could I increase input level to the preamp by different wiring or additional components?
•Any other solutions, like different mic-preamp or a simple way to reduce the input signals of the other channels...etc?
Sorry for the noob questions, I'm learning by building and help is much appreciated.
Thanks in advance and rspct to all audio engineers all over the world!
-Malik
Does anything change for the better when you connect XLR pin 3 to ground instead of leaving it floating? It depends on the characteristics of the source whether pin 3 can best be left open or grounded when driving an unbalanced load. I mean pin 3 of the PP2B output.
Do you want a microphone preamplifier or a microphone preamplifier and voice processor? What you have now is a cheap voice processor combined with a mediocre microphone preamplifier. You can easily make a better microphone preamplifier with a few op-amps, resistors, capacitors and diodes, but you won't have the voice processor functionality then.
With your present preamplifier/voice processor, you can try setting a large compression ratio and a low downward expander threshold, for example Rcomp = 180 kohm and Rexp = 4.7 kohm. You will not be able to get the output level above -6 dBV RMS, because the limiter is fixed to that level.
If this should help enough, then you will probably want to increase the downward expander threshold again to reduce the chance of beeping sounds due to acoustic feedback. You then have to experiment to find suitable values. On the other hand, if the sound is too weak even with 180 kohm and 4.7 kohm, then you know this won't solve the issue.
Do you want a microphone preamplifier or a microphone preamplifier and voice processor? What you have now is a cheap voice processor combined with a mediocre microphone preamplifier. You can easily make a better microphone preamplifier with a few op-amps, resistors, capacitors and diodes, but you won't have the voice processor functionality then.
With your present preamplifier/voice processor, you can try setting a large compression ratio and a low downward expander threshold, for example Rcomp = 180 kohm and Rexp = 4.7 kohm. You will not be able to get the output level above -6 dBV RMS, because the limiter is fixed to that level.
If this should help enough, then you will probably want to increase the downward expander threshold again to reduce the chance of beeping sounds due to acoustic feedback. You then have to experiment to find suitable values. On the other hand, if the sound is too weak even with 180 kohm and 4.7 kohm, then you know this won't solve the issue.
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Thnxs for your help 🙂
I'll try grounding pin 3 of the pp2b and see if it makes a difference. I saw some resources on the internet that do this when going from balanced xlr to mono so I guess I'll just give it a try. Intuitively it feels strange that combining this reversed phase signal to ground (assuming its balanced output) makes a good mono signal...
As for the voice compressor I think it's not needed. As for the way it's sounding now there's a bit of lag sometimes and it's quite noisy. Just a pre-amp would be fine I guess. Probably it needs quite a bit of gain since the mic output seems low.
Meanwhile I ordered this pre amp kit which is on its way now. Do you think it's decent or would I be better of building something else?
As for the current amp-processer module that I'm using I will try out your suggestions to see if it makes a good enough improvement.
Thnxs again!
I'll try grounding pin 3 of the pp2b and see if it makes a difference. I saw some resources on the internet that do this when going from balanced xlr to mono so I guess I'll just give it a try. Intuitively it feels strange that combining this reversed phase signal to ground (assuming its balanced output) makes a good mono signal...
As for the voice compressor I think it's not needed. As for the way it's sounding now there's a bit of lag sometimes and it's quite noisy. Just a pre-amp would be fine I guess. Probably it needs quite a bit of gain since the mic output seems low.
Meanwhile I ordered this pre amp kit which is on its way now. Do you think it's decent or would I be better of building something else?
As for the current amp-processer module that I'm using I will try out your suggestions to see if it makes a good enough improvement.
Thnxs again!
thnxs! much appreciated.
I'm planning to build 2 karaoke sets which require 4 mic pre-amps total, so next to this kit I'll have to make 3 additional pieces anyway.
So if it's easier to come up with a schematic that isn't a modification of the kit that would be fine too.
I'm planning to build 2 karaoke sets which require 4 mic pre-amps total, so next to this kit I'll have to make 3 additional pieces anyway.
So if it's easier to come up with a schematic that isn't a modification of the kit that would be fine too.
I think it will improve a lot with these changes:
R2 = 2.2 kohm
R5 = 10 kohm
IC1 = NE5532
Decoupling to be added between pins 8 and 4, for example 22 uF, 50 V electrolytic.
It still won't be the world's best microphone amplifier, but at least its noise will be well below the noise of the microphone and you won't get the crossover distortion of the original LM358.
(For anyone who wants to check my statements about noise: according to http://www.samsontech.com/site_media/legacy_docs/C02_ownman_v1s.pdf , the microphone has a sensitivity of 10 mV/Pa (-40 dBV at 1 pA) and an equivalent noise level of 22 dB(A). That boils down to an electrical noise of 2.5 uV A-weighted at the microphone output. As the noise bandwidth of A-weighting is about 13 kHz, that corresponds to an A-weighted average noise voltage density of approximately 22 nV/sqrt(Hz). If I did the calculation correctly, the modified amplifier will do something like 9 nV/sqrt(Hz), well below 22 nV/sqrt(Hz). I assume that the microphone will produce the full voltage on pin 2 when its pin 3 is AC grounded, otherwise that 22 becomes 11 nV/sqrt(Hz).)
R2 = 2.2 kohm
R5 = 10 kohm
IC1 = NE5532
Decoupling to be added between pins 8 and 4, for example 22 uF, 50 V electrolytic.
It still won't be the world's best microphone amplifier, but at least its noise will be well below the noise of the microphone and you won't get the crossover distortion of the original LM358.
(For anyone who wants to check my statements about noise: according to http://www.samsontech.com/site_media/legacy_docs/C02_ownman_v1s.pdf , the microphone has a sensitivity of 10 mV/Pa (-40 dBV at 1 pA) and an equivalent noise level of 22 dB(A). That boils down to an electrical noise of 2.5 uV A-weighted at the microphone output. As the noise bandwidth of A-weighting is about 13 kHz, that corresponds to an A-weighted average noise voltage density of approximately 22 nV/sqrt(Hz). If I did the calculation correctly, the modified amplifier will do something like 9 nV/sqrt(Hz), well below 22 nV/sqrt(Hz). I assume that the microphone will produce the full voltage on pin 2 when its pin 3 is AC grounded, otherwise that 22 becomes 11 nV/sqrt(Hz).)
If it doesn't need to fit on your PCB, there are many ways to build a microphone preamplifier - and it then makes sense to give it a balanced input. For example, the standard application circuits of microphone preamplifier chips like the SSM2019 or the THAT1510 are pretty good. There is a not-so-standard circuit described in this thread: https://www.diyaudio.com/community/threads/thatmicpre-an-open-source-mic-preamp.356317/ . It needs a clean 48 V supply and takes care of phantom supplying the microphone.
Attached is a schematic of a microphone preamplifier I once built for a local radio station here; it did not need to phantom supply the microphone, but had to survive the phantom supply voltage applied by another circuit. I built two of them and they were used continuously for several years. The input transistors are no longer available, but there are alternatives. I can clean up the schematic a bit if you are interested 😉
Do you have any specific constraints regarding supply voltages, used components, costs or anything else?
Attached is a schematic of a microphone preamplifier I once built for a local radio station here; it did not need to phantom supply the microphone, but had to survive the phantom supply voltage applied by another circuit. I built two of them and they were used continuously for several years. The input transistors are no longer available, but there are alternatives. I can clean up the schematic a bit if you are interested 😉
Do you have any specific constraints regarding supply voltages, used components, costs or anything else?
Attachments
🙂 I'll give this a try first and see how it operates. Seems like something that fits my level of electronics 😉 Thnxs for the help!
Wow, thnxs for resources. As for now the schematic seems a bit overwhelming for my level but if the first solution doesn't satisfy I'll might have to take the challenge.
As for constraints actually everything is possible but its not going to be used professionally so decent would be good enough. Also since I'm already having the PP2B phantom power supply just an amplifier circuit would do I suppose. Currently the main power supply is 12v which is converted to 48v in the PP2B. Maybe the biggest constrain is to keep it simple for a beginner in electronics. Maybe I could also check if there's something I can get from one of the custom pcb manufacturer's library to keep it a bit comprehensible.
But I'll first try to see how solution 1 will do.
Hi,
There seems to be commercial solutions available similar price than empty enclosure for custom build would cost and doesn't seem worth the trouble to built similar quality stuff DIY https://www.thomann.de/fi/the_t.mix_mix_802.htm
I understand if its for fun / learning then please go ahead!🙂
ps. on XLR cabling pin 1 is shield and contains error current only, do not connect that to anywhere else than to chassis, to earth. Outputs might have lift to pin 1 if there is ground loop. There is no reason to bring error current to any PCB. If in doubt, borrow Henry W Ott book about electro magnetic compatibility, or skim through Rane notes https://www.ranecommercial.com/legacy/note151.html
There seems to be commercial solutions available similar price than empty enclosure for custom build would cost and doesn't seem worth the trouble to built similar quality stuff DIY https://www.thomann.de/fi/the_t.mix_mix_802.htm
I understand if its for fun / learning then please go ahead!🙂
ps. on XLR cabling pin 1 is shield and contains error current only, do not connect that to anywhere else than to chassis, to earth. Outputs might have lift to pin 1 if there is ground loop. There is no reason to bring error current to any PCB. If in doubt, borrow Henry W Ott book about electro magnetic compatibility, or skim through Rane notes https://www.ranecommercial.com/legacy/note151.html
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Thnxs for your reply. Probably cheaper/ easier to buy something, but my idea is to built something like in this image 😉 And indeed enjoy the learning process along the way.
I'm waiting for the parts to arrive and see if the modified kit will do a decent enough job. I'll share an update asap
It's a pity that grounding pin 3 didn't help. I propose you disconnect pin 3 again then; if it doesn't increase the level, it can only cause some extra distortion.
Whether I agree with tmuikku's remark about pin 1 depends on whether you end up with a microphone preamplifier with an unbalanced or a balanced input. In the balanced case, you indeed have to connect all pins 1 straight to the chassis or enclosure and connect that to the ground of the electronics at one and only one point (per the AES-48 standard). In the unbalanced case, I would connect pin 1 of each microphone input straight to the corresponding microphone preamplifier's input ground terminal, with as little loop area as possible between the wires connecting pins 2 and 1 to the amplifier PCB.
Whether I agree with tmuikku's remark about pin 1 depends on whether you end up with a microphone preamplifier with an unbalanced or a balanced input. In the balanced case, you indeed have to connect all pins 1 straight to the chassis or enclosure and connect that to the ground of the electronics at one and only one point (per the AES-48 standard). In the unbalanced case, I would connect pin 1 of each microphone input straight to the corresponding microphone preamplifier's input ground terminal, with as little loop area as possible between the wires connecting pins 2 and 1 to the amplifier PCB.
The preamp I use now has unbalanced mono input. I have removed the socket from the PP2B output. How about connecting pin 1 and 3 at the pp2b and then have one wire going to the pre amp ground? I guess this would give the smallest wire length.
Considering that you didn't notice any difference whether pin 3 was connected to ground or left open, it looks like your microphone + PP2B behaves as two signal sources, one between pin 2 and pin 1 and one in antiphase between pin 3 and pin 1. A typical dynamic microphone, on the other hand, would behave as a signal source between pins 2 and 3, with pin 1 used only for shielding. The only advantage I can think of now when you connect pin 3 to pin 1, is compatibility with such microphones.
Regarding the connection to the preamplifier, what matters is the enclosed loop area: the smaller the area, the less magnetic pick-up of mains hum and other varying magnetic fields. When you use normal shielded cable, with pin 2 connected to the centre conductor and pin 1 to the shield, the cable contributes almost nothing to the effective enclosed area because the centre conductor and shield are concentric.
You just have to look out what you do at the ends of the cable. What you preferably shouldn't do is connect the centre wire or centre pin of the connector (if you use one) straight to the unbalanced microphone amplifier PCB input and make a big detour from the shield of the connector to the microphone amplifier's input ground pin.
If you have a metal enclosure, it also has to be connected to ground to act as a shield to electric fields. The usual rule of thumb is to do that only at the most sensitive input, which would normally be the microphone input connector, but it is better for RF immunity (but sometimes worse for hum) to connect the enclosure to the shields of all input and output connectors.
Twisted cable also keeps the effective loop area small, because the wires are close together and the odd and even half turns sort of compensate each other. It doesn't have the electric shielding that you get from shielded cable, though.
The professional solution is balanced inputs and outputs and cables that are both shielded and twisted. The shield can then be connected straight to the enclosure, so the enclosures and shields form one big Faraday's cage, and the enclosed area that has to be kept small is the area between the twisted wires that connect pins 2 and 3. Anyway, a balanced input is the next step if the unbalanced amplifier doesn't work satisfactory.
Regarding the connection to the preamplifier, what matters is the enclosed loop area: the smaller the area, the less magnetic pick-up of mains hum and other varying magnetic fields. When you use normal shielded cable, with pin 2 connected to the centre conductor and pin 1 to the shield, the cable contributes almost nothing to the effective enclosed area because the centre conductor and shield are concentric.
You just have to look out what you do at the ends of the cable. What you preferably shouldn't do is connect the centre wire or centre pin of the connector (if you use one) straight to the unbalanced microphone amplifier PCB input and make a big detour from the shield of the connector to the microphone amplifier's input ground pin.
If you have a metal enclosure, it also has to be connected to ground to act as a shield to electric fields. The usual rule of thumb is to do that only at the most sensitive input, which would normally be the microphone input connector, but it is better for RF immunity (but sometimes worse for hum) to connect the enclosure to the shields of all input and output connectors.
Twisted cable also keeps the effective loop area small, because the wires are close together and the odd and even half turns sort of compensate each other. It doesn't have the electric shielding that you get from shielded cable, though.
The professional solution is balanced inputs and outputs and cables that are both shielded and twisted. The shield can then be connected straight to the enclosure, so the enclosures and shields form one big Faraday's cage, and the enclosed area that has to be kept small is the area between the twisted wires that connect pins 2 and 3. Anyway, a balanced input is the next step if the unbalanced amplifier doesn't work satisfactory.
Thanks again. I'll go with unbalanced for now since the preamp kit has mono input only. With regard to connecting the mic output to the preamp, and all other signals to the mixer, I soldered regular 24 awg insulated wires. But no I'm thinking I might better use something that is shielded like this? Or is the noise neglectable on these short wires within a wooden enclosure?
Also, I researched a bit about balanced signals, which makes me better understand the concept now. However I'm wondering what the component or circuit is called that would do the conversion to get the signal with the noise subtracted. From my understanding now there is some conversion to do at the end of the line before the signal is fed into the amp right?
Try and you will see, or hear actually. Try to keep the loop areas small by twisting, keep the wires that connect the microphone to the preamplifier far away from mains cables and mains transformers and keep your fingers crossed.