https://theultimatetone.com/ Mon, 11 May 2026 12:47:40 +0000 MyBB https://theultimatetone.com/Thread-PZM-Experiments Fri, 27 Dec 2024 09:42:11 +0000 physics]]> https://theultimatetone.com/Thread-PZM-Experiments https://theultimatetone.com/Thread-Warming-up-the-sound Sun, 21 Jan 2024 16:19:28 +0000 K O'Connor]]> https://theultimatetone.com/Thread-Warming-up-the-sound
These days, the default for everything seems to be "digital". There may be no real statistics regarding the type of equipment home studios and commercial studios use, but cost-wise the recording and storage of the music is often in digital memory. Apple made things easy for home recording and mix-downs with the Garage Band software, which seems to be an entry-level version of Protools, which as the name suggests, is used by professional sound engineers and studios.

The problem with digital is that the sound played back can be a bit "lifeless". The opinion of most people is formed partly by what equipment marketers tell them, so it is common to think that digital is distortionless. The fact is that digital processing adds its own unique kinds of distortion that all fall into the range of IM, intermodulation distortion, inasmuch as IM and digital distortion and other digital noise are all non-music-related. Our brains get fatigued trying to sort out how this fits in with the music.The conversion of analog to digital (ADC) and back again (DAC) adds some of these artifacts, such as jitter, dither, and anti-aliasing noise, as a continuous wave is sliced and diced with each sample turned into a digital word of a specific bit length. The situation improves as more bits are added and the sampling rate increases.

When digital audio first appeared, 14-bit ADCs and DACs were considered "good enough" for portable players, like the Sony Discman, but 16-bit was better for home hifi. Modern systems use 24-bits which improves the resolution by 32 times. The novelty of it shrouded opinion and there were and are still many compromises.

The ADC compares the signal to a voltage divider with many small equally-spaced increments. The signal level matching a given point on that reference divider will cause the related output comparator to produce an output. This is a bit simplified but the basic description applies. A problem with the lowest signals is that they may be buried in noise, such as hiss. Our ear can still hear signals that low provided the noise does not cancel too many frequencies present in the signal. However, this makes the low-level bit range unusable and both resolution and dynamic range are lost. This conversion has to be extremely fast and synchronised to an accurate and clean clock. The conversion back to analog has to be similarly synchronised and requires a lot less circuitry.

For a recording system, there may be many signals being converted and stored at the same time and this can tax the computer, so it is best if this is the only thing the computer has to do at that time. I one is trying to process the audio digitally while capturing it, the computer time is divided between tasks and neither task may be done as well as possible. This is why there are dedicated boxes for streaming audio for high fidelity playback; the specialised box has only one task and can be otpimised to provide the best performance possible.

Analog recording systems are imperfect, too, Vintage equipment used vacuum tubes, then solid-state devices, evolving and improving in quantitative measure with reductions of total harmonic distortion (THD) and IM. recording devices generally used magnetic tape of various widths running at low or high speeds. Reaching the limits of each piece of equipment in the chain set limits of clean performance, but some of these limits were utilised for effect by players and recorders alike. For example, a given track width of magnetic particles on tape can only handle a specific signal size, after which the tape "saturates" - the signal is effectively clipped as it can go no larger. There is a "fattening" of the signal when this happens, similar to when a tube amp approaches clipping.

Apart from the extreme nonlinear effects offered by some analog equipment, the basic character for nominally clean sounds is present. Tubes handle small signals differently than they handle large signals, due to the curved transfer function they have. This means their harmonic additions to the sound changes with signal level and they add a veil of character many people find pleasing. There are others who find this distortion disturbing. Solid-state circuits generally have lower noise and distortion than tubes do, but their distortion profile can be more disturbing. In any case, the long history of analog recording and playback systems informs our ear and creative mind when it comes to using a variety of equipment and techniques for recording music.

Some people try to source analog equipment and steadfastly work only in that domain. The digital guys see that some of the techniques used in the past added often pleasant aspects to the musical result and aimed at mimicking these effects in the computer. Once the signal is digital, it is just a matter of performing mathematical functions to achieve a different sonic output. Delays. Echoes. Reverb. These are effects that are easily recognised as "added" and that do not have to be perfect in their execution to be effective. Tape saturation. Hiss reduction. Pitch adjustment. These things are more critical, particularly the latter, as they are often applied to Human voices where we have genetic programming to assess the result.

Pitch adjustment is of particular concern as it is already extremely over-used. This makes the voice sound unnatural and artificial - maybe that is the point? but it loses some of the organic quality we expect.

With all of these different algorithms overlayed to achieve the individual effects, the computer will tend to add more noise and nonmusical processing artifacts. Again, it is up to the skill of the operator to know how to minimise these artifacts with the system he is using.

One solution to the problem of over-taxing the computer and/or to cover some of the unwanted hash is to warm the signal prior to being captured by the computer. Many studios still have analog mic preamps and some seek out even older equipment and refurbish it. Old tube-based designs for microphone preamps and line amps that were good in their day but far from perfect. However, they do have character and add in warmth that is lacking from the best solid-state gear and from digital processors. They may warm the signal after the DSP, as well or instead of preceding the DSP. Yes, their are apps one can install in the computer to add these effects specifically, but this is a bandaid not a cure. Adding the external warming device is also a bandaid, but it is something much more relatable and that we can fine-tune to our own taste.

There are modern tube mic preamps as well, so one does not have to deal with dead capacitors and carbon resistor noise and poor layouts. Some would say those things are part of the charm, but you can sometimes have too much of a good thing.

It would be interesting to know if people using and looking for that vintage gear generally prefer full mic preamps or just line amps?

Have fun]]>
These days, the default for everything seems to be "digital". There may be no real statistics regarding the type of equipment home studios and commercial studios use, but cost-wise the recording and storage of the music is often in digital memory. Apple made things easy for home recording and mix-downs with the Garage Band software, which seems to be an entry-level version of Protools, which as the name suggests, is used by professional sound engineers and studios.

The problem with digital is that the sound played back can be a bit "lifeless". The opinion of most people is formed partly by what equipment marketers tell them, so it is common to think that digital is distortionless. The fact is that digital processing adds its own unique kinds of distortion that all fall into the range of IM, intermodulation distortion, inasmuch as IM and digital distortion and other digital noise are all non-music-related. Our brains get fatigued trying to sort out how this fits in with the music.The conversion of analog to digital (ADC) and back again (DAC) adds some of these artifacts, such as jitter, dither, and anti-aliasing noise, as a continuous wave is sliced and diced with each sample turned into a digital word of a specific bit length. The situation improves as more bits are added and the sampling rate increases.

When digital audio first appeared, 14-bit ADCs and DACs were considered "good enough" for portable players, like the Sony Discman, but 16-bit was better for home hifi. Modern systems use 24-bits which improves the resolution by 32 times. The novelty of it shrouded opinion and there were and are still many compromises.

The ADC compares the signal to a voltage divider with many small equally-spaced increments. The signal level matching a given point on that reference divider will cause the related output comparator to produce an output. This is a bit simplified but the basic description applies. A problem with the lowest signals is that they may be buried in noise, such as hiss. Our ear can still hear signals that low provided the noise does not cancel too many frequencies present in the signal. However, this makes the low-level bit range unusable and both resolution and dynamic range are lost. This conversion has to be extremely fast and synchronised to an accurate and clean clock. The conversion back to analog has to be similarly synchronised and requires a lot less circuitry.

For a recording system, there may be many signals being converted and stored at the same time and this can tax the computer, so it is best if this is the only thing the computer has to do at that time. I one is trying to process the audio digitally while capturing it, the computer time is divided between tasks and neither task may be done as well as possible. This is why there are dedicated boxes for streaming audio for high fidelity playback; the specialised box has only one task and can be otpimised to provide the best performance possible.

Analog recording systems are imperfect, too, Vintage equipment used vacuum tubes, then solid-state devices, evolving and improving in quantitative measure with reductions of total harmonic distortion (THD) and IM. recording devices generally used magnetic tape of various widths running at low or high speeds. Reaching the limits of each piece of equipment in the chain set limits of clean performance, but some of these limits were utilised for effect by players and recorders alike. For example, a given track width of magnetic particles on tape can only handle a specific signal size, after which the tape "saturates" - the signal is effectively clipped as it can go no larger. There is a "fattening" of the signal when this happens, similar to when a tube amp approaches clipping.

Apart from the extreme nonlinear effects offered by some analog equipment, the basic character for nominally clean sounds is present. Tubes handle small signals differently than they handle large signals, due to the curved transfer function they have. This means their harmonic additions to the sound changes with signal level and they add a veil of character many people find pleasing. There are others who find this distortion disturbing. Solid-state circuits generally have lower noise and distortion than tubes do, but their distortion profile can be more disturbing. In any case, the long history of analog recording and playback systems informs our ear and creative mind when it comes to using a variety of equipment and techniques for recording music.

Some people try to source analog equipment and steadfastly work only in that domain. The digital guys see that some of the techniques used in the past added often pleasant aspects to the musical result and aimed at mimicking these effects in the computer. Once the signal is digital, it is just a matter of performing mathematical functions to achieve a different sonic output. Delays. Echoes. Reverb. These are effects that are easily recognised as "added" and that do not have to be perfect in their execution to be effective. Tape saturation. Hiss reduction. Pitch adjustment. These things are more critical, particularly the latter, as they are often applied to Human voices where we have genetic programming to assess the result.

Pitch adjustment is of particular concern as it is already extremely over-used. This makes the voice sound unnatural and artificial - maybe that is the point? but it loses some of the organic quality we expect.

With all of these different algorithms overlayed to achieve the individual effects, the computer will tend to add more noise and nonmusical processing artifacts. Again, it is up to the skill of the operator to know how to minimise these artifacts with the system he is using.

One solution to the problem of over-taxing the computer and/or to cover some of the unwanted hash is to warm the signal prior to being captured by the computer. Many studios still have analog mic preamps and some seek out even older equipment and refurbish it. Old tube-based designs for microphone preamps and line amps that were good in their day but far from perfect. However, they do have character and add in warmth that is lacking from the best solid-state gear and from digital processors. They may warm the signal after the DSP, as well or instead of preceding the DSP. Yes, their are apps one can install in the computer to add these effects specifically, but this is a bandaid not a cure. Adding the external warming device is also a bandaid, but it is something much more relatable and that we can fine-tune to our own taste.

There are modern tube mic preamps as well, so one does not have to deal with dead capacitors and carbon resistor noise and poor layouts. Some would say those things are part of the charm, but you can sometimes have too much of a good thing.

It would be interesting to know if people using and looking for that vintage gear generally prefer full mic preamps or just line amps?

Have fun]]> https://theultimatetone.com/Thread-Stereo-in-stereo-out Tue, 29 Aug 2023 19:45:18 +0000 K O'Connor]]> https://theultimatetone.com/Thread-Stereo-in-stereo-out
Anyone who has read my books knows that my preference for music playback is proper stereo. There are several aspects involved regarding how to achieve the same experience in the listening room as existed on stage during a performance. That stage might also have been a recording studio, where the recording process can alter the performance quite radically depending on how each instrument sound is captured.

The simplest approach to recording with an eye (ear?) to playing back in stereo, is to use a two-microphone  setup to capture the room mix. This mimics how a listener would experience the performance directly, and hopefully this transfers correctly to the listening room experience. There are thousands of recordings made this way from the earliest days of electronics up to the digital age. Some are great, some are poor, and most are good.

In our stereo recording approach, we can use two microphones on their own stands separated however the recording engineer determines, or two mics can be "crossed" at close proximity to each other, or a binaural mic-head can be used.

The separate microphones method is the easiest to implement as it requires no special equipment and actually the minimum of skill to achieve a pretty good result. The microphones should be in positions where listeners would be, far enough away that each microphone picks up all instruments. Their left-right positioning should be modest but distinctly not coincident; rather, tending to extreme separation. This provides a natural blending of the sounds with room ambience for a normal feel. If the microphones are widely spaced, the distinction of instrument placement across the stage may be enhanced, with caveats: Each instrument must have its own single speaker cabinet or "zone" where its sound is dominant; the PA sound ideally carries only vocals or has a proper mix corresponding to the stage positions of the instruments; or the PA sound is minimised as much as possible in the stereo microphone placement.

There are a lot of details in the above descriptions to be expended upon as we go along.

The binaural head-mic is designed to mimic the placement of Human ears within our head, and thus, the mount is shaped like a Human head, with ear canals that have microphones at the place where our ear drum would be. The exterior contouring is either highly simplified or shaped to represent an average shape of a Human ear. Ears are unique and no two people hear exactly the same thing. besides, our brains make an interpretation of the information coming from our ears as to what it is we are hearing.

The stereo signals captured by the microphones only need two tracks on tape or whatever form the recording system takes. Playing it back on a distortionless home listening system is the ideal case, especially if that system is properly positioned in a"Human conversation rated" room.]]>
Anyone who has read my books knows that my preference for music playback is proper stereo. There are several aspects involved regarding how to achieve the same experience in the listening room as existed on stage during a performance. That stage might also have been a recording studio, where the recording process can alter the performance quite radically depending on how each instrument sound is captured.

The simplest approach to recording with an eye (ear?) to playing back in stereo, is to use a two-microphone  setup to capture the room mix. This mimics how a listener would experience the performance directly, and hopefully this transfers correctly to the listening room experience. There are thousands of recordings made this way from the earliest days of electronics up to the digital age. Some are great, some are poor, and most are good.

In our stereo recording approach, we can use two microphones on their own stands separated however the recording engineer determines, or two mics can be "crossed" at close proximity to each other, or a binaural mic-head can be used.

The separate microphones method is the easiest to implement as it requires no special equipment and actually the minimum of skill to achieve a pretty good result. The microphones should be in positions where listeners would be, far enough away that each microphone picks up all instruments. Their left-right positioning should be modest but distinctly not coincident; rather, tending to extreme separation. This provides a natural blending of the sounds with room ambience for a normal feel. If the microphones are widely spaced, the distinction of instrument placement across the stage may be enhanced, with caveats: Each instrument must have its own single speaker cabinet or "zone" where its sound is dominant; the PA sound ideally carries only vocals or has a proper mix corresponding to the stage positions of the instruments; or the PA sound is minimised as much as possible in the stereo microphone placement.

There are a lot of details in the above descriptions to be expended upon as we go along.

The binaural head-mic is designed to mimic the placement of Human ears within our head, and thus, the mount is shaped like a Human head, with ear canals that have microphones at the place where our ear drum would be. The exterior contouring is either highly simplified or shaped to represent an average shape of a Human ear. Ears are unique and no two people hear exactly the same thing. besides, our brains make an interpretation of the information coming from our ears as to what it is we are hearing.

The stereo signals captured by the microphones only need two tracks on tape or whatever form the recording system takes. Playing it back on a distortionless home listening system is the ideal case, especially if that system is properly positioned in a"Human conversation rated" room.]]>