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.

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Samples rates on iPad Air 3 and similar devices - stuck or not

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Comments

  • @CapnWillie said:
    Can someone translate in Laidman terms for one who doesn’t speak geekbench?

    I have a new iPad Pro 2021 which I presume to be locked at whatever samplerate (48khz?).

    How does it matter? Will it noticeably effect the sound of my projects at some point?

    Hopefully it won't matter at all. On the newer devices leaving the sample rate at 48k is the best option. You might encounter problems with plugins or samples that are expecting 44.1 but that should be pretty rare by now.

  • @CapnWillie said:
    Can someone translate in Laidman terms for one who doesn’t speak geekbench?

    I have a new iPad Pro 2021 which I presume to be locked at whatever samplerate (48khz?).

    How does it matter? Will it noticeably effect the sound of my projects at some point?

    Depends on your projects and what you want. If you really need the control, say in a professional mastering situation, then it could matter.

    For me, I treat my iPad more like a synth or guitar processing chain. So, as long as it sounds good to me everything is good. If I can think a freaking noisy ass class A guitar amp sounds amazing or the 12-bit output of an early DX7 is great, then my iPad is just fine. But, I did say above that if I were making recordings for archival purposes or for later mixing that I would want to use as high a bit rate and sample depth as I possibly could and because of that I wouldn't use my iPad for that job.

  • Seconded. 96khz is a waste of resources, unless the material was explicitely captured or synthesized for this rate.
    Which means both adequate input for digitizing sources and playback devices.
    Such stuff does exist (specialized publishers content), but it‘s pointless for all iTunes and streaming.

  • This all makes me wonder what happened to the oversampling AD/DA converters that used to be on early devices?
    ...I have an old Zoom 1202 FX unit, it's 16-bit 44.1K but it also uses 64x oversampling at the AD stage to minimize any kind of aliasing on the way in to the device and I've never heard any audible aliasing from it when using high-frequency saw waves etc.

  • @NeonSilicon said:

    @tja said:
    [...]

    I am still a bit confused by what you wrote about iOS without interface - you could not manage to set a higher sample rate in this case? How do MultiTrack DAW and Auria Pro achive this?

    The sample rate I'm referring to is the sample rate of the audio session for the whole iOS device. This is the sample rate of the singleton instance of the AVAudioSession. This instance controls the audio for the live output on the device.

    There are a couple of ways that you could run a host or other sort of app using a different sample rate from that being run by the AVAudioSession. One way would be to use the enableManualRenderingMode method on the AVAudioEngine. I'd think that this would be useful for rendering mixes to files at whatever sample rate the user wanted. Another would be to set up a more complicated audio graph that included nodes that could handle the sample rate conversion from the running audio on the graph to the output nodes that go to the audio output.

    The audio subsystem on iOS is pretty flexible and can do all sorts of stuff that you don't normally see. I was only looking at the ability to set the format of the live session output.

    The discussion has been totally focused on the iOS audio subsystem so far... but nothing keeps a DAW developer from simply implementing their own audio rendering altogether (and in fact, that's what I would recommend given the pretty lame documentation (at least 2 years ago) of AVAudioEngine and related stuff and the large amount of ambiguities). I'm pretty sure that's what NS2 does, for example. I don't think it uses the iOS audio libraries for anything other than hardware input and output (and of course (reluctantly in Matt's perspective) for hosting AUv3s).

    I must admit that I'm not a huge fan of libraries/frameworks in general and tend to avoid them whenever I can, even if that means more implementation work... I'm just too burnt by bad documentation and bugs. But maybe I'm also just too demanding 😆

    (in my defence though -- I have added experimental AUv3 hosting in Xequence just for the heck of it and the amount of "strangeness" I've encountered during that experiment -- both in terms of AVAudioEngine's documentation / behaviour and especially unstable / non-"standard"-conforming plugins -- has been so demotivating that I've quickly given up on it 😒 I have no idea how @j_liljedahl managed to make AUM that stable and compatible and forgiving 😁)

    Anyway... that was a huge diversion. Just what you can expect when I enter a thread!

  • @Samu said:
    This all makes me wonder what happened to the oversampling AD/DA converters that used to be on early devices?
    ...I have an old Zoom 1202 FX unit, it's 16-bit 44.1K but it also uses 64x oversampling at the AD stage to minimize any kind of aliasing on the way in to the device and I've never heard any audible aliasing from it when using high-frequency saw waves etc.

    Funny, Zoom 1202 😄 my first FX unit, bought in 1997! I kept it all those years for decorative purposes!

  • tjatja
    edited January 2022

    @richardyot said:
    I've done a few more tests this morning:

    In Loopy Pro I can set any sample rate from 44.1k --> 48k --> 96k no matter which device is connected.

    Yes, but it does not work - in my tests, it spits out 48k files (without interface)!

    In AUM I can only set 48k with no interface connected, and with an interface connected I can set 44.1k and 48k but not 96k. This seems to apply to all my interfaces, even the class-compliant ones that should support 96k.

    Same here, with and without interface.

    In Auria Pro I can set any sample rate from 44.1k --> 48k --> 96k no matter which device is connected.

    And this works even with NO interface attached!
    Same as with MultiTrack DAW.

    The Dragonfly Red (and the Black as well) have an LED that changes colour depending on the sample rate.

    In Loopy Pro the colour of that LED correctly reflects the selected sample rate.

    In Auria Pro this only applies to 44.1k and 48k, when the project is set to 96k the LED glows green to indicate the output from Auria is at 44.1k

    In AUM it's really weird: if I set the sample rate to 96k there is an error message to say the sample rate is not supported, but the LED on the Dragonfly glows magenta to indicate that the sample rate is set to 96k. I have no idea what's going on to be honest.

    Strange things.

    You would need to test with the interface attached and mixdown / export files to check the real rates that were used.

    I will send back the SoundBlaster interface and most probably will never buy another.

  • @tja said:
    I tried to use auGEN X within an Auria Pro project with 96k to create sound at 32k, but it refused and restricted the sound to 20.48k.... even as the session was shown as 96k:

    Shouldn't that be possible, @auDSPr?

    Any other plugin or App that can produce sound higher than 22.5kHz?
    I seek out those mentioned above

    Hi @tja and everyone,

    auGEN X is an audio generator (hence the name, hah hah) - so I hadn't envisioned it to be a supersonic audio generator. This is why the frequency setting range maxes out at 20,480 Hz. That being said, you still can use auGEN X to generate harmonics above 20 kHz. Have it generate a saw at 6 kHz and you'll have harmonics at 12k, 18k, 24k, on up. At 48 kHz sample rate, the 4th harmonic (24 kHz) and up will fold back (i.e., alias). Switch to 96 kHz sample rate, and now the foldback won't start until the 8th harmonic (48 kHz). You can hear this easily by using auGEN X's Frequency Sweep feature. I've attached an auGEN X Preset (zipped) which demonstrates this. Import it into auGEN X and hear it for yourself. When the nominal 6 kHz saw slowly increases in frequency (pitches up), you'll hear the upper harmonics pitching down - this your aliasing. When you increase sample frequency from 48 kHz to 96 kHz, you'll hear the aliasing noticeably diminish. You don't need a fancy audio system to hear this phenomenon, even using my iPad's built-in speakers my middle-aged ears can hear it clear as a bell. (Note that I had to use my iTrack Dock for the 96 kHz test.)

    I hope this helps. BTW, auGEN X is alias-suppressed not alias-free so this is why we can do this test. At musical fundamental frequencies (below around 4 kHz), you shouldn't have much trouble with aliasing.

    • Dave Simpao, auDSPr
  • Oh, many many thanks, @auDSPr !!!

    Now, as auGEN X is used to create test tones, I would have expected it to work with higher frequencies :smile:
    What if I need a sine wave at 1280kHz? i cannot expect your regular synth to create such audio ...

    It would be great to use auGEN X for such cases ... hint ... hint :smile:
    Just saying ...

    And special thanks for the preset - will check this out!

  • @tja said:

    @AudioGus said:

    @tja said:

    @AudioGus said:

    @tja said:

    @AudioGus said:

    @Telefunky said:

    @tja said:

    @Telefunky said:
    And it is really hard to tell the difference in a final mix. o:)

    This has absolutely nothing to do with the creation of music.
    This is just a topic around sound processing on iDevices.

    Sorry, I should have quoted...

    @tja said:
    But a DAW that could run at 44.1k, 48k, 88.2k, 96k, 192k or even higher, accepting, handling and exporting such content, without needing an attached interface, would simply be fantastic for all people creating music on iDevices that are hardware locked to 48k ... hint... hint 😅

    which was the reason for slightly extending the perspective, but just ignore it in this context.

    On the other hand it is related to the „locking to samplerate“ by Apple, as they have to consider a general audience.
    Afaik their audio codec chips are custom designed with no general specs available.
    These chips vary with hardware and there maybe whatever consequences affecting some and some not.

    Frankly said: Apple sacrificed a feature that only 1 out of 1000 customers even knows about, and only 1 out of 100 from this group considers it relevant.

    I just wish 1 out of 1 devs would fix their plugin to work at both 44 and 48 (or maybe it is the hosts faults?) ... le sigh

    Which plugin shows problems? And in which host?

    Egoist in NS2 has problems. I should give it a more clinical try again in BM3 and Cubasis.

    The VST has problems too with 44/48 so I just imagined it is a Sugar Bytes thing.

    I have DrumComputer, Factory, Aparillo, Unique, Cyclop, Thesys and Turnado, but not Egoist :-D
    May try later with one of them.

    I do have export problems with Aparillo from NS2 regardless of 44/48. The resulting export sometimes sounds all messed up like the parameters got randomly jumbled. If I play the project file after export it is messed up in the same way. Reloading fixes it. So I have been having to use BM3 to realtime record flakey NS2 tracks. I am just doing my mass exodus from iOS bouncing out all my stems.

    Did you mentioned that in the nanostudio forum or by support email?

    Yes, way back in the day, along with a couple other plugins.

    Mass exodus?

    Are you leaving i*OS?!? 😳 😳 😳

    Since I am not commuting anymore and have Maschine I dont really get the itch to make new stuff on iOS so much but a lot of the old stuff can find new life on desktop.

    I have a ton of tracks that I need to bounce out as audio before plugins/ios changes etc and they sound all messed up. I have a few now that are screwed up because of new AU issues.

  • @Samu said:
    This all makes me wonder what happened to the oversampling AD/DA converters that used to be on early devices?
    ...I have an old Zoom 1202 FX unit, it's 16-bit 44.1K but it also uses 64x oversampling at the AD stage to minimize any kind of aliasing on the way in to the device and I've never heard any audible aliasing from it when using high-frequency saw waves etc.

    Lots of delta-sigma type converters now where they do very high sample rate conversion at low bit depth and then convert that digitally to a lower sample rate but high bit depth signal. It allows for much easier to construct analog pre-filters and then use very good digital filters in the internal conversion as well as things like digital noise shaping to reduce quantization errors. Pretty slick stuff, https://en.wikipedia.org/wiki/Delta-sigma_modulation

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