Audiobus: Use your music apps together.

What is Audiobus?Audiobus is an award-winning music app for iPhone and iPad which lets you use your other music apps together. Chain effects on your favourite synth, run the output of apps or Audio Units into an app like GarageBand or Loopy, or select a different audio interface output for each app. Route MIDI between apps — drive a synth from a MIDI sequencer, or add an arpeggiator to your MIDI keyboard — or sync with your external MIDI gear. And control your entire setup from a MIDI controller.

Download on the App Store

Audiobus is the app that makes the rest of your setup better.

The Audiobus Forum University of Music Production or Audio/Sound Engineering Discussion

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Comments

  • I thought about building an emitter follower using the amp module in Drambo so I could use it as a limiter..... I need a zener diode module.

  • @ehehehe said:
    This kind of internal shit should be moved to a subreddit

    I'm really glad @ehehehe has opted out of this thread. We will never see another post and the chill factor it sends for a productive "Discussion". We don't need to waste time begging someone to join a Discussion and actually contribute.

    The functioning of the forum is driven by participation. Ignore a thread and it's gone into the bowels of the search box. If a thread bothers anyone... just stay out of it and never comment. That will speed it's death. But never go into a thread and tell people they are wasting YOUR time. That front page real estate can also be managed by only viewing favorites. Then periodically just checking for new content and favoriting anything worth your attention. Simple.

    The Drambo thread dominated the front page for months and drove tight assed jerks crazy
    because it never had anything they valued until the App dropped. Then it rolled away into the archive with scatter posts of highly useful information wrapped in snide remarks and chit chat. As it should be on a forum people use to connect around topics of shared interest.

    This thread will have value when someone asks an interesting question. That has yet to happen.

    Can someone explain "Gain Staging" technique with a practical example?

    A Guitar
    into an Audio Interface
    into an iPad
    into AUM
    into an Amp Sim
    into some AU FX
    into a File Recorder in AUM

    What are some tips for getting a great recording? Do you always want each stage to
    hit dB 0 on the meters to maximize the signal to noise ratio?
    Pointers to helpful YouTube Videos appreciated since text is often the least useful media
    for education on some of these topics. Musical examples are needed to illustrate the tips.

    If we get great answers we can move the text into the Wiki or create a list of links that point back to the relevant pages here. Like a Wiki, this works when someone contributes more than snide remarks. For anyone that feels my comments are useless crap, I will make it clear
    my strongest contribution is and will be comic relief. It amuses me as a creative exercise. If you don't practice you'll never improve.

  • edited September 17

    id like to have a grey noise source
    (white noise adjusted according to fletcher/munson curve)

    @horsetrainer

  • @McD said:

    I'm really glad @ehehehe has opted out of this thread. We will never see another post and the chill factor it sends for a productive "Discussion". We don't need to waste time begging someone to join a Discussion and actually contribute.

    Agreed.

    The functioning of the forum is driven by participation. Ignore a thread and it's gone into the bowels of the search box. If a thread bothers anyone... just stay out of it and never comment. That will speed it's death. But never go into a thread and tell people they are wasting YOUR time. That front page real estate can also be managed by only viewing favorites. Then periodically just checking for new content and favoriting anything worth your attention. Simple.

    The Drambo thread dominated the front page for months and drove tight assed jerks crazy
    because it never had anything they valued until the App dropped. Then it rolled away into the archive with scatter posts of highly useful information wrapped in snide remarks and chit chat. As it should be on a forum people use to connect around topics of shared interest.

    One of my most favourite bookmarks is the dRambo thread.

    This thread will have value when someone asks an interesting question. That has yet to happen.

    You seem to be asking all of the most interesting questions.
    We'll wait for you.

    Can someone explain "Gain Staging" technique with a practical example?

    A Guitar

    For a guitarist? Turn it up.
    For everyone else? Turn it down.

    into an Audio Interface

    set the input gain to -12db and
    if you're feeling hot -6db when using line level signals.
    When using microphones and guitars etc?
    Keep the loudest part of the signal out of the red.

    into an iPad

    Get an audio interface.

    into AUM
    into an Amp Sim
    into some AU FX
    into a File Recorder in AUM

    See above.
    Get an audio interface unless you only use pre recorded samples,
    software synths and you like the graininess of the internal mic.
    The internal mic is quite useful by the way.

    What are some tips for getting a great recording? Do you always want each stage to
    hit dB 0 on the meters to maximize the signal to noise ratio?

    No, trying to get it to hit 0db all the time isn't the way.

    The loudest peak should hit 0db that
    way you get dynamics and headroom.
    You can normalise a recorded track afterwards
    to increase overall levels but a ruined digital track?
    Is very difficult to rescue.
    Been there.

    Pointers to helpful YouTube Videos appreciated since text is often the least useful media
    for education on some of these topics. Musical examples are needed to illustrate the tips.

    I'll see what I can put together.
    My rig is a gain staging nightmare.
    It's a complete mixture of analog and digital.
    I check the calibration every two weeks.

    If we get great answers we can move the text into the Wiki or create a list of links that point back to the relevant pages here. Like a Wiki, this works when someone contributes more than snide remarks. For anyone that feels my comments are useless crap, I will make it clear
    my strongest contribution is and will be comic relief. It amuses me as a creative exercise. If you don't practice you'll never improve.

    Well said.
    We need more humour these days.

    My humour tends to be a little bit, out there
    so if stand up is your thang?
    Please, fire away... 🤩

  • @Gravitas said:
    I check the calibration every two weeks.

    Please explain the steps to calibrate your rig. Is the a useful signal source to hit known
    levels at each stage? We're trying to teach audio engineering for free so all donations of experience appreciated. If this experiment fails it's just noise to signal at work. I can't gate
    out the noise but I will just STFU. Seriously.

  • For anyone seeking solid advice on Gain Staging this thread has solid engineering advice:

    https://forum.audiob.us/discussion/41045/lowering-gain-staging-across-track-without-losing-relative-levels-preserving-volume-relationships#latest

    the comments from @drewfx1 are helpful. For example he writes:

    -5dB equals a ratio of about .56x*. So you're just multiplying every sample in each track by .56 and if you look at it that way you should see that the relative levels are the same.

    It's tricky because dB's are logarithmic and that changes the math. For instance, adding dB's is actually the same as multiplying the corresponding ratios - that's one of the reasons they're convenient to use. I think it's actually easier to understand what's happening terms of ratios, but in practice dB's are very convenient to use.

    *For anyone interested, these are the conversion formulas for the type of dB's used in audio:

    dB's = 20 * log10(ratio)

    ratio = 10 ^(dB/20)

    That should generate a few questions depending on your understanding of math. If the math is a barrier is there a useful translation of concepts for musicians? I suspect most of us
    dial in a sound without thinking in these concepts of dB monitoring and engineering.

    But a graduate of this thread should seek the knowledge. If you don't care. Don't ask. If we can pull @drewfx1 in to explain we'll all get a little closer to basic professional competency
    and stop being clueless.

  • Why is Bluetooth audio so useless for realtime audio making wireless headphones useless?

    @Wim points to a great explanation for engineering students:
    https://electronics.stackexchange.com/questions/354458/why-bluetooth-audio-works-at-so-high-latency

    Questions? How much latency can real time audio tolerate and not feel real?

  • wimwim
    edited September 17

    @McD said:
    Why is Bluetooth audio so useless for realtime audio making wireless headphones useless?
    Questions? How much latency can real time audio tolerate and not feel real?

    Answer: it depends completely on the individual. Some people can adapt to surprising amounts of latency.

    Just try to explain to someone sensitive to it that 5ms difference is approximately the amount of time that sound takes to travel 5 feet and see how well it's received. :D

    Still, 30 - 40ms in addition to the latency already coming from the OS seems like it would be pretty hard for anyone to adjust to. But it is possible, in theory. In fact there was one poster on the forum who claimed he had no latency with his Air Pods pro, even when playing live. Since that's physically impossible, I can only assume he had that type of tolerance.

  • @Max23 said:
    hm, its a very technical value that has little to do with what your ear is hearing, as this is not linear

    https://en.wikipedia.org/wiki/Equal-loudness_contour

    Interestingly there's a unit called a "phon" used in psychoacoustics that's calibrated to the equal loudness curves pictured for dB SPL at 1kHz. IOW, 40 phons = the equal loudness contour shown that equals 40 dB SPL at 1kHz.

    Anyway,

    1. A dB as used in audio is a logarithmic expression of a ratio between 2 signals.
    2. If dB is used with a suffix (dB SPL, dBFS, dBV, dBu, etc.) then you are comparing one value to a predefined reference level. For example -20dBFS means either "20dB below Full Scale" (fixed point) or "20dB below a value of 1" (in floating point, the convention is 0dBFS = 1).
    3. If dB is used with no suffix, then you are comparing 2 values "track B is 20dB below track A".

    The formulas for converting between dB's (as used in audio) and ratios are:

    dB's = 20 * log10(ratio)

    ratio = 10^(dB/20)

  • @McD said:

    @Gravitas said:
    I check the calibration every two weeks.

    Please explain the steps to calibrate your rig. Is the a useful signal source to hit known
    levels at each stage? We're trying to teach audio engineering for free so all donations of experience appreciated.

    I hear you.

    So let's begin.

    First things first load up the Oscillator app as it's the simplest sine wave generator.
    Then get the Levels app, though it's IAA it still works.
    Both of these apps are free.

    Set the Oscillator app to 1Kz and 0db output.
    Turn up the volume on the idevice to max.
    I'm using my iPhone but this can also be done with an iPad.

    You can do the same thing with only one device
    in that case then you need to loop back the signal.

    I'll explain that method a little later.

    Mute or turn down inputs on the audio interface.

    Plug in said device and turn up the input levels.

    The end result in regards to input level should
    look something like this in the Levels app.

    This is the input signal for my Behringer Q802usb into my 9.7" iPad Pro 1st gen.
    It would be the same going into my iPad Air 3 using the Zoom U-44.

    If I need to be a little bit hotter I'll push that to -7db or -6db on input.

    Using this method, l get a really good signal to noise ratio without it clipping.
    Once I've got the potentiometers or channel faders set then
    I can plug in a keyboard or compressor and I've got a good starting point.

    With analogue instruments such guitars and similar I tend to use my ears.

    Also knowing your equipment is important
    as sometimes you may want it to distort.

    My compressors and preamps have certain characteristics when overdriven.

    This is the signal chain that I tend to calibrate.

    Zoom U-44 output via 3+4 (Outputs 1+2 go to the studio monitors)
    iCon Pro Audio NeoPreamp
    Joemeek C2 stereo compressor
    Behringer q802usb
    Behringer SRC 2496 ultramatch pro which is
    being used as an Analogue to digital converter.

    I set the output of the Behringer mixer
    to -7db going into the SRC2496.
    SRC2496 converters that to an SPDIF signal
    which I then route back into the Zoom U-44.

    Here's a screenshot from the iPad Air 3.

    I set the inputs for analogue preamps and
    compressors by ear and eye for the input signal.

    The Joemeek inputs can be pushed for instance.

    Certain pieces of gear have huge amounts of self noise
    which is why you need to know your equipment.

  • @Gravitas said:
    So let's begin.

    Now that's what I was hoping for... Wiki quality instruction. Kudos to @wim and @drewfx1 as well for their valuable input.

  • edited September 17

    @drewfx1 said:

    @Max23 said:
    hm, its a very technical value that has little to do with what your ear is hearing, as this is not linear

    https://en.wikipedia.org/wiki/Equal-loudness_contour

    Interestingly there's a unit called a "phon" used in psychoacoustics that's calibrated to the equal loudness curves pictured for dB SPL at 1kHz. IOW, 40 phons = the equal loudness contour shown that equals 40 dB SPL at 1kHz.

    Anyway,

    1. A dB as used in audio is a logarithmic expression of a ratio between 2 signals.
    2. If dB is used with a suffix (dB SPL, dBFS, dBV, dBu, etc.) then you are comparing one value to a predefined reference level. For example -20dBFS means either "20dB below Full Scale" (fixed point) or "20dB below a value of 1" (in floating point, the convention is 0dBFS = 1).
    3. If dB is used with no suffix, then you are comparing 2 values "track B is 20dB below track A".

    The formulas for converting between dB's (as used in audio) and ratios are:

    dB's = 20 * log10(ratio)

    ratio = 10^(dB/20)

    db spl is a strange measurement to talk about as u have a signal at -40db,
    you add +6db
    you perceive it as now its double as loud ...

  • @McD said:

    Now that's what I was hoping for... Wiki quality instruction.

    Thank you.

    Kudos to @wim and @drewfx1 as well for their valuable input.

    +1

  • @Max23 said:
    db spl is a strange measurement to talk about as u have a signal at -40db,
    you add +6db
    you perceive it as now its double as loud ...

    It actually makes sense when you do the conversion from dB to ratio:

    ratio = 10^(6/20) = ~1.995

    You'll also see 3dB shows up in a lot of places. It corresponds to a ratio of the square root of two.

  • edited September 17

    @drewfx1 said:
    ratio = 10^(6/20) = ~1.995

    this is totally new to me, never heard of ratio in this context :)
    I understand how perception works,
    but this math formula is alien language to me

    I see u inserted the 6db and it makes 2
    meaning double the perceived loudness? :)

    odd, I have never seen this formula before, but I skipped most of my math classes and went smoking in the gents room. I should have payed more attention. ^^

  • McDMcD
    edited September 17

    @Max23 said:

    @drewfx1 said:
    ratio = 10^(6/20) = ~1.995

    but this math formula is alien language to me

    Just for anyone that is interested in the Math notation:

    1. 6 / 20 = 0.3
    2. 10 to the 0.3 power (this takes a scientific calculator) = ~1.995
      • on a scientific calculator look for the 10 with a little X in the superscript:
        ~ means approximately or close to.

    Hit it for the result with x = 0.3 and you get the result:

    If you can do this in your head it's probably a fairly common calculation you have seen before. Knowing how to apply a fraction as a power of 10 is not something they drill in school but engineers get to know how to use these calculators when required. They used to have slide rules to assist and before that they published books of Logs so you could look it up in a table and someone calculated the value in the book by hand and that's another lesson.

    NOTE: Students will be expected to provide their own scientific calculator for use in class.
    See the ABF Store for details and student loans.

  • edited September 17

    hihi
    I cant remember a thing about math except Cross-multiplication & %
    ^^
    the rest of the stuff I never used in real life ;)

  • @Gravitas said:

    @McD said:

    @Gravitas said:
    I check the calibration every two weeks.

    Please explain the steps to calibrate your rig. Is the a useful signal source to hit known
    levels at each stage? We're trying to teach audio engineering for free so all donations of experience appreciated.

    I hear you.

    So let's begin.

    First things first load up the Oscillator app as it's the simplest sine wave generator.
    Then get the Levels app, though it's IAA it still works.
    Both of these apps are free.

    Set the Oscillator app to 1Kz and 0db output.
    Turn up the volume on the idevice to max.
    I'm using my iPhone but this can also be done with an iPad.

    You can do the same thing with only one device
    in that case then you need to loop back the signal.

    I'll explain that method a little later.

    Mute or turn down inputs on the audio interface.

    Plug in said device and turn up the input levels.

    The end result in regards to input level should
    look something like this in the Levels app.

    This is the input signal for my Behringer Q802usb into my 9.7" iPad Pro 1st gen.
    It would be the same going into my iPad Air 3 using the Zoom U-44.

    If I need to be a little bit hotter I'll push that to -7db or -6db on input.

    Using this method, l get a really good signal to noise ratio without it clipping.
    Once I've got the potentiometers or channel faders set then
    I can plug in a keyboard or compressor and I've got a good starting point.

    With analogue instruments such guitars and similar I tend to use my ears.

    Also knowing your equipment is important
    as sometimes you may want it to distort.

    My compressors and preamps have certain characteristics when overdriven.

    This is the signal chain that I tend to calibrate.

    Zoom U-44 output via 3+4 (Outputs 1+2 go to the studio monitors)
    iCon Pro Audio NeoPreamp
    Joemeek C2 stereo compressor
    Behringer q802usb
    Behringer SRC 2496 ultramatch pro which is
    being used as an Analogue to digital converter.

    I set the output of the Behringer mixer
    to -7db going into the SRC2496.
    SRC2496 converters that to an SPDIF signal
    which I then route back into the Zoom U-44.

    Here's a screenshot from the iPad Air 3.

    I set the inputs for analogue preamps and
    compressors by ear and eye for the input signal.

    The Joemeek inputs can be pushed for instance.

    Certain pieces of gear have huge amounts of self noise
    which is why you need to know your equipment.

    its pretty simple,
    drive analog stuff as hot as you can to get a nice s/n ratio,
    with digital stuff there is no need to drive it hot so make sure you never hit the red

  • @Max23 said:
    hihi
    I cant remember a thing about math except Cross-multiplication & %
    ^^
    the rest of the stuff I never used in real life ;)

    I learned a bunch of stuff back in the day that I largely forgot because I never had a use for it.

    Then I started playing around with DSP and wished I had remembered more (or paid better attention). :|

  • edited September 18

    @drewfx1 said: Then I started playing around with DSP and wished I had remembered more (or paid better attention). :|

    yeah, me too ^^
    but luckily I have ppl I can feed my ideas to, so I dont have to deal with the complicated math ^^

    I guess that's the way memory goes, u dont use it u forget it, I was able to talk French fluently, these days I can just tell you the way to the supermarket. ^^
    maybe it comes back if I had it more around my ears.

  • @Max23 said:

    its pretty simple,
    drive analog stuff as hot as you can to get a nice s/n ratio,

    with digital stuff there is no need to drive it hot so make sure you never hit the red

    True.

    As long as it's not distorting or it's distorting in the way one
    needs it for the sound then make sure it doesn't over load.

    I'm working on several different projects and configurations.

    I try to keep levels as constant as possible.

  • edited September 18

    The Bel was established before it was used to measure electrical signals... but still around the time of early telephone development.

    Someone decided they wanted to measure "Sound Level" as perceived by humans. As we know sound is an air pressure wave, but measuring the absolute amplitude of the air pressure wave that makes a 'sound' wasn't useful. They wanted to measure what people_ hear. _
    So first they set off to find the "zero point" of hearing. The Threshold of Human Hearing. By generating low level sounds and testing what were described as a group of youths, they discovered that the smallest air pressure a human could perceive was 0.98 pW/m2 at 1 atmosphere and 25 °C. Thats a very small amount of air-pressure. Only a few percent above the noise created by molecular motion.

    (OK, OK ... it's obviously different for different people... etc etc ... but science ).

    So, this 'sound' became defined as 0 Bels. Less pressure than this than this produces "no sound'. Great. Then they gradually increased the pressure level of the test sound until the test group reported that it had become "twice as loud". That became 1 Bel. "Bel" literally meant "twice as loud". 2Bels.. twice as loud again. Yes, very subjective but don't blame me.

    It pretty quickly became clear that the increase in air pressure required to produce increases in units of 1 Bel wasn't linear. It was Logarithmic. That is if they doubled the number of Bel's, the air pressure went up much more than twice . It also became clear that the "Bel" was not granular enough to be useful ... so it was divided into 10 subunits. 10 decibels = 1 Bel = 10dB. Done.

    Since telephony was a purely electrical analog of sound pressure/waves then it followed that the dB would be a great measurement tool for it as well. Variations in the electrical levels of a signal followed the same fundamental rules as actual sound waves.

    Cool. So then we got
    For measuring sound in the air - ** dB spl** = "sound pressure level" 0dB = 0.98 pW/m2
    For measuring an electrical signal amplitude - dBu = using .775 volts as the 0 point

    and yes that ".775" makes the math more difficult so eventually someone suggested
    For measuring an electrical signal amplitude - dBv = "electrical signal amplitude" using 1 volt as the 0 point

    Because u and v never get mixed up.

    There are other dB scales ... dBm for instance which measures the "area under the curve of the sound/electrical wave"... otherwise known as it's "Power". This most closely correlates to human hearing. Double the power ..it sounds twice as sound. Kind of.

    Oh and one last thing. When measuring electrical signals how come they decided .775Volds ( or 1 Volt ) was "zero" dB?? Pretty weird... 0dB is clearly not "zero signal". Well it was because they originally used dB to measure how much signal LOSS was present between two ends of a telephone connection. If the signal left the sender at .775 Volts and arrived at .775 Volts it dropped by 0dB. The meter therefore showed Zero (loss) ... or full strength. If it arrived at 38.Volts the meter would show -6dB ... a loss of half its original amplitude.

    From here you can dig deeper if you want.

  • McDMcD
    edited September 18

    @ltf3 Great history and science lesson for the curriculum. Free Cookie for you at:

    https://magazineline.com

    Just hit "Accept"

  • @ltf3 said:
    and yes that ".775" makes the math more difficult so eventually someone suggested
    For measuring an electrical signal amplitude - dBv = "electrical signal amplitude" using 1 volt as the 0 point

    Because u and v never get mixed up.

    Actually it's more mixed up than that. :#

    0 dBv (small v) is the same .775V as dBu.

    It's dBV (capital V) that's referenced to 1V.

    Apparently they thought it was less confusing to make it more confusing.

    https://en.wikipedia.org/wiki/Decibel#Voltage

    Also, all of this explains why the -10dBV and +4dBu standards are NOT 14dB apart. They are referenced to 2 different voltages and you have to convert them to the same voltage reference to see how far apart they are.

  • @McD said:

    @Gravitas said:
    So let's begin.

    Now that's what I was hoping for... Wiki quality instruction. Kudos to @wim and @drewfx1 as well for their valuable input.

    Big +1. Took a while to bring up to proper temperature, but now we have a fine thread going...
    Dang it now I need to get that Levels app

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