Hi everyone,
I don’t know if the title is correct as I don’t have much idea about audio technically speaking. Anyway, I will try to explain myself.
Recently I have been recording my bass guitar using the DI output from the amp and a microphone in the cabinet. I recorded each signal to one channel and when I was listening to both at the same time it sounded much wicker compared to the individual tracks. After some research I learned that this is due to a phase/time misalignment between both signals and this is creating cancelation issues that are especially noticeable in the low frequencies.
I have seen two methods to solve this problem:
The first one is to time/phase align both signals. This is relatively easy to do in the DAW as you can move the tracks to time align them; but it’s not so easy in a live situation. I’ve seen that there are some hardware devices to time/phase align the signals but they are quite expensive.
The second one is to apply a high pass filter to one signal and a low pass filter to the other signal, both in the same frequency, so you use one signal for the lows and the other one for the highs. I am interested in this approach since it makes sense to use the lows from the DI signal, which are clear and well defined, and use the mids and highs from the microphone, which is able to get the character of the cabinet and adds interesting flavors to the overall sound. This is also quite easy to do in a DAW, but not so easy in a live situation.
I have been figuring out how to do it live, but the only option seems to be to use a stereo parametric equalizer. And even with these I have several problems. The selectable high pass filter and low pass filter frequencies don’t normally overlap so I can’t achieve what I am looking for. Also, they normally work in line level and I would prefer to be able to do it in mic level since it’s what a sound guy is expecting from me. I want to be able to blend the signals myself and give a single mic level signal to the sound guy.
So, summarizing, what I want is a device to connect two balanced mic level signals, add a high pass filter to one, add a low pass filter to the other one, both with the same cutoff frequency, and then blend them (a crossover the other way araund?). It would be great to have individual level controls. The output should be a balanced mic level signal.
About the filters, in the DAW I had the best results using steep slope filters (36 dB/octave). If I am not wrong for this an active filter is needed. It would be great to have a selectable cutoff frequency from something like 80Hz up to 1kHz for instance. Again, I don’t know if this is something easy to do or It doesn’t make sense at all.
I imagine the device with two xlr inputs, one xlr output, and three potentiometers (DI level, MIC Level and cutoff frequency).
I attach a block diagram. Let me know if this makes sense and if it is easy to do. I would like to end up with a schematic of the device to be able to build it myself.
It would be a really handy device for me. I appreciate your help.
Have a nice day.
Unai
I don’t know if the title is correct as I don’t have much idea about audio technically speaking. Anyway, I will try to explain myself.
Recently I have been recording my bass guitar using the DI output from the amp and a microphone in the cabinet. I recorded each signal to one channel and when I was listening to both at the same time it sounded much wicker compared to the individual tracks. After some research I learned that this is due to a phase/time misalignment between both signals and this is creating cancelation issues that are especially noticeable in the low frequencies.
I have seen two methods to solve this problem:
The first one is to time/phase align both signals. This is relatively easy to do in the DAW as you can move the tracks to time align them; but it’s not so easy in a live situation. I’ve seen that there are some hardware devices to time/phase align the signals but they are quite expensive.
The second one is to apply a high pass filter to one signal and a low pass filter to the other signal, both in the same frequency, so you use one signal for the lows and the other one for the highs. I am interested in this approach since it makes sense to use the lows from the DI signal, which are clear and well defined, and use the mids and highs from the microphone, which is able to get the character of the cabinet and adds interesting flavors to the overall sound. This is also quite easy to do in a DAW, but not so easy in a live situation.
I have been figuring out how to do it live, but the only option seems to be to use a stereo parametric equalizer. And even with these I have several problems. The selectable high pass filter and low pass filter frequencies don’t normally overlap so I can’t achieve what I am looking for. Also, they normally work in line level and I would prefer to be able to do it in mic level since it’s what a sound guy is expecting from me. I want to be able to blend the signals myself and give a single mic level signal to the sound guy.
So, summarizing, what I want is a device to connect two balanced mic level signals, add a high pass filter to one, add a low pass filter to the other one, both with the same cutoff frequency, and then blend them (a crossover the other way araund?). It would be great to have individual level controls. The output should be a balanced mic level signal.
About the filters, in the DAW I had the best results using steep slope filters (36 dB/octave). If I am not wrong for this an active filter is needed. It would be great to have a selectable cutoff frequency from something like 80Hz up to 1kHz for instance. Again, I don’t know if this is something easy to do or It doesn’t make sense at all.
I imagine the device with two xlr inputs, one xlr output, and three potentiometers (DI level, MIC Level and cutoff frequency).
I attach a block diagram. Let me know if this makes sense and if it is easy to do. I would like to end up with a schematic of the device to be able to build it myself.
It would be a really handy device for me. I appreciate your help.
Have a nice day.
Unai
Attachments
The device you are after is not easy to design or build, though most live digital consoles have all the equalization, HP and LP filters and delay needed on every channel, use one for the direct signal and one for the mic.
To blend the signals and give a single mic level signal to the sound guy you could use a stereo pre-amp, send the outputs to a DSP (Digital Signal Processor) like the Mini DSP:
https://www.minidsp.com/products/minidsp-in-a-box/minidsp-2x4-hd
to do all the processing, combine the L/R signal within the DSP, and adjust the mono output to an acceptable level, balance the output with a DI or an adapter.
To blend the signals and give a single mic level signal to the sound guy you could use a stereo pre-amp, send the outputs to a DSP (Digital Signal Processor) like the Mini DSP:
https://www.minidsp.com/products/minidsp-in-a-box/minidsp-2x4-hd
to do all the processing, combine the L/R signal within the DSP, and adjust the mono output to an acceptable level, balance the output with a DI or an adapter.
Hello,
Thanks for your reply, I really apreciate it. I see, I thought it should be pretty easy but my knowledge about audio circuits it's very limited.
The digital world opened my mind and after some research I bought a used Behringer Shark DSP110 which costed 40€. This device has a lot of features including a delay line, and you can choose to work with mic level or line level. So my idea is to delay the DI signal to time align it with the microphone signal. Then I am planning to use a Radial 2:1 Mix to merge both signals.
The Behringer Shark DSP110 is a cheap device so probably it will degrade the quality of the DI signal. I wouldn't use it to record an album but I guess it will work fine for live situations.
I will post here the results as soon as I can make the tests.
Unai
Thanks for your reply, I really apreciate it. I see, I thought it should be pretty easy but my knowledge about audio circuits it's very limited.
The digital world opened my mind and after some research I bought a used Behringer Shark DSP110 which costed 40€. This device has a lot of features including a delay line, and you can choose to work with mic level or line level. So my idea is to delay the DI signal to time align it with the microphone signal. Then I am planning to use a Radial 2:1 Mix to merge both signals.
The Behringer Shark DSP110 is a cheap device so probably it will degrade the quality of the DI signal. I wouldn't use it to record an album but I guess it will work fine for live situations.
I will post here the results as soon as I can make the tests.
Unai
I made some tests and it was a disaster. I recorded two tracks in the daw. The DI from the amp and a mic in the cabinet. The mic track was about 5 ms later. I duplicated the track and I aligned them in the DAW. There was a noticable difference. When time aligned the tracks sounded fuller.
The I tried to do this but intead of moving the tracks in the DAW I added the Shark DSP110 to the DI signal to apply some delay. I recorded the tracks again and with the delay value set in 0 ms the signal was already like 10 ms later than the mic signal. I guess this is the latency of the DSP. So the device is not useful for what I was trying to do.
The I tried to do this but intead of moving the tracks in the DAW I added the Shark DSP110 to the DI signal to apply some delay. I recorded the tracks again and with the delay value set in 0 ms the signal was already like 10 ms later than the mic signal. I guess this is the latency of the DSP. So the device is not useful for what I was trying to do.